VoIP.ms Wiki - User contributions [en]

File:StartProcedure.png

2020-11-15T07:10:15Z

Rod:

Voicemail

2020-07-08T10:18:47Z

Rod:

VoIP.ms has an Advanced Voicemail feature that is free to use, and you also have the option to forward your messages to your email address as an attachment.

In order to use the Voicemail feature with VoIP.ms you will have to create a Voicemail entry and then assign your entry to one of your DID or Accounts.

*The total allowed voicemail messages on a single mailbox is 100.
*Please white list the mail domain names *@voip.ms and *@voipinterface.net.
*The Maximum Time for a Voicemail message is 3 minutes.
*5 Seconds of Silence will end a Voicemail Recording.

== Tutorial Video ==
[http://youtu.be/v2rqE1yOFMQ Voicemail Video Tutorial]

== Voicemail Accounts ==

From the Customer Portal refer to DID Numbers -> Voicemail, then you will see the following screen. In there you will be able to see a list of any existing Voicemail accounts, or create a new Voicemail account (Mailbox) by clicking on the "Create New Voicemail Account" Button.

[[File:VoicemailAccts.png|900px]]

== Create a Voicemail mailbox ==

Pressing the "Create New Voicemail Account" button will display a new window.

[[File:NewVoicemailAcc.png|370px]]

You will be prompted with the following information:

'''Voicemail Number:''' This will be used as a unique identifier for your mailbox. The minimum is adding one digit to up to ten digits, for example you can set 1 or 5432897.

'''NOTE:''' The number cannot be set to 0. If you have your mailbox to 0 it's suggested to change to a different value to prevent issues with the voicemail service.

'''Name:''' This can be used as a note or description to easily identify your mailboxes.

'''Password:''' The password is used to enter your mailbox options such as listen to your messages or record your greeting.

'''Skip Password Prompt:''' If set to Yes, when dialing *97 from an account associated to this mailbox will skip the password prompt and login directly.

'''Notification Email:''' If an email address is entered here, the mailbox system will send an Email notification every time you receive a new message. For the moment you can only set 1 email address. However you can optionally configure an email forward between your email accounts as a work around.

'''Voicemail Language:''' This sets the language you and the caller will hear when instructions or menus are played by the voicemail.

Pressing the "View Advanced Mode" button will display further options to modify your mailbox.

[[File:Create Voicemail.png|900px]]

'''Unavailable Message Recording:''' This is the greeting that the system will play to the callers that reach your voicemail. There are three options available for this recording:

*System Default - The system will use the generic greeting message recording when the callers reach the Voicemail.

*Set by Phone - The system will use the recording set by the phone by the user via dialing *97 / *98 and choosing option 0 to record it.

*Recordings - The system will use the associated [[Recordings]] previously uploaded via the Customer Portal tool at DID Numbers > Recordings.

'''Voicemail Transcription:''' If enabled, a voicemail to text transcription will be sent to your email. You should select the language that the callers are likely to use when they leave voicemail messages. This feature has a cost of '''$0.005 per minute of transcribed message.'''

'''Attach message to email:''' If set to YES, the Mailbox will attach a .WAV file containing the new message every time it sends an email notification.

'''Delete Voicemail Message''' If set to YES, the Mailbox will delete the new message automatically after sending the Email notification with attachment.

'''Attachment Format:''' You can select between WAV49, recommended because of its smaller size and easiness to handle; WAV which is uncompressed, causing a bigger file size; and mp3 which offers great compatibility and is very small.

'''Say Instructions:''' If set to YES, the caller will hear instructions on how to leave a message to your mailbox before the beep sound.

'''Say Time / Date:''' If set to YES, when checking your messages you will hear the date and time when the message was received.

'''Time Zone:''' The time envelope will use this time zone to provide the correct date and time of the message's reception.

'''Say Caller ID:''' If set to YES, when checking your messages you will hear the Caller ID of the message sender.

== Assigning your Voicemail to your DID==

After you have created your Voice Mail entry, you can assign it to any of your DID from your main portal. Please refer to DID Numbers -> [[Manage DID]] -> Edit DID -> Voicemail. Also under the same screen you can set the Ring Time (The maximum amount of time a call to your DID can stay in "Ringing State" before we cancel the call to no answer). Please note that 30s equals to 6 rings.

[[File:selectVoicemail.png|750px|thumb|left|Select Voicemail - click to enlarge]]
<br clear="all" />

== Assigning your Voicemail to your Account==

If you would like to assign a Voicemail entry to your Main Account, please from your main portal refer to Main Menu -> [[Account Settings]] -> General -> Voicemail Associated to the Main Account.

[[File:Mainvoicemail.jpg]]

If you need to assign a Voicemail to a specific subaccount, you need to go to the [[Sub Accounts]] Edit page, following the route, Subaccounts >> Manage Subaccounts >> Edit, from the menu tabs.
You will see at the bottom of the page the "Internal Extension Voicemail" option. Here you can set it.

[[File:Subvoicemail.jpg]]

== Manage Voicemail and Messages ==

[[File:VoicemailAccts.png|800px]]

Your mailbox will appear under '''Voicemail Accounts''' after it is created through your Customer Portal>>DID Numbers>>Voicemail page. In there, you can click the '''Delete''' button to delete the mailbox completely or '''Delete Msgs''' to delete all of the messages in the given mailbox.

If you click '''Edit''', a new window will be displayed where you will be able to change the settings in your voicemail. However, please note that you cannot change the mailbox ID once it has been created.

You can also manage your voicemail messages by clicking on the '''List''' icon next to your mailbox. Within this new window, you will be able to listen to a message, mark it as urgent, forward this message to an email address, download the audio file, mark it as "read", store it in a folder or delete it.

[[File:Voicemail List.png|800px]]

You also have the possibility to select multiple entries and delete them in one click by selecting them on the left row and then pressing *Delete Selected Records*
[[File:VmDel.png|800px]]

== Manually Transcribing a Voicemail ==

You can manually request a transcription of the messages left on your Mailbox through the Voicemail interface. Using this feature has a cost of '''$0.005 per minute of transcribed message.'''

To begin the transcription process, you need to click on the dialogue box icon next to the message that you wish to be transcribed. The color of this icon represents the message transcription status:

* Blue: Ready to transcribe
* Yellow: Transcription in rogress
* Red: Transcription failed, you may retry the transcription
* Green: Transcription Successful and Available

[[File:Transcription Interface.png|800px]]

The transcription process can take a couple of minutes, depending on the duration of the message. You need to refresh the webpage (By pressing F5 on your keyboard or the refresh button of your browser) to see the updated status of this process.

Once the transcription is finished, the dialogue box icon will turn green and clicking on it will show the transcribed message.

[[File:Transcription Successful.png|350px]]

== Voicemail greeting customization ==

If you want to change the default voicemail greeting on your mailbox you will have 2 options.

* The '''First''' option is to do it directly from your phone, once you have created the mailbox and you have assigned it to your DID number and to your account, dial *97 from your phone and at the Voicemail menu select the option "0" (Voicemail options) and then the option "1" (Unavailable message) there you will be able to record your greeting, and to save it. Any greeting set this way will take precedence over any greeting options, including uploading a Recording or using the System Default greeting.

* The '''Second''' option is in case that you already have the recording that you want to use as your voicemail greeting. You can upload this recording to DID Numbers > [[Recordings]] in the Customer Portal. Your uploaded recording will then appear as a greeting option at DID Numbers > Voicemail > Edit Mailbox > View Advanced Mode (Upper Right) > Unavailable Message Recording.

*Please note that you must change the '''Unavailable Message Recording''' option of your Voicemail account to the desired greeting for the changes to be reflected.

*Recording your greeting following the First option, using your phone and the audible Voicemail system, will take precedence over any Recording uploaded or the System Default.
If you wish to use an uploaded Recording or the System Default greeting, please delete your recorded greeting first.

== Voicemail Access Codes: ==

You can access your voicemail with any device/system connected directly with your account or subaccount to VoIP.ms using the codes below:

*<nowiki>*97</nowiki> (Asterisk 97) is used to access the mailbox directly associated to the account you are dialing from. If you would like to check which mailbox is associated to your account refer to [[Voicemail#Assigning_your_Voicemail_to_your_Account|Assign Voicemail]]

*<nowiki>*98</nowiki> (Asterisk 98) is used to access your Voicemail and choose one of your mailbox accounts. (Will prompt for mailbox ID and Password)

If for any reason you do not have access to our VoIP network, you can check your Voicemail by just dialing your DID. Once the Voicemail system answers your call, press the asterisk key (*). You will be prompted for the mailbox ID (or hear the mailbox automatically selected, depending on the server you are using) and Password, once logged in to your Voicemail, press 0 (zero) for options. You can also record your greeting from there by selecting option 4 (four). Please be aware this is considered a regular incoming call and will be charged according to your monthly DID plan.

== Navigate the Voicemail Menu ==

Once you access your voicemail you're going to be prompted with the number of new and/or old messages you have in the mailbox. Here's the list of options you have with the voicemail system of VoIP.ms

* 1 - Play the first new/old message available in your mailbox.

* 2 - Change folders. This option allows you to change to another folder in order to hear the messages you have stored in that folder. At the moment it's not possible to change the name of the folders.
0 - New Messages
1 - Old Messages
2 - Work
3 - Family
4 - Friends
# - Cancel

* 0 - Voicemail options. In here you can change your greetings and record your name, also you can change the password for your voicemail.

1 - Unavailable message
2 - Busy message
3 - Name.
4 - Temporary message
5 - Change Password
* - Return to the Main Menu

'''Notes:'''
* The Temporary Message will override the Voicemail message option set in the Customer Portal until it is deleted through your phone by dialing *97, 0 for options, 4 for Temporary Message and then 2 to Delete it.
* To Delete a Recording you can just record again, a moment of silence, or create a new recording overwriting the old.
* Please note that if you change the password over the phone ('''Option 5'''), it will not change at your customer portal>>Voicemail options,
and you will have to manually change the password there. If you don't change it at the customer portal and you modify any other value
later, when you apply the changes the old password will be set again to your mailbox, since the modify option apply all the values set at
the voicemail options.

'''The following options are available when you're listening to your messages.'''

* 3 - Advanced Options
1 - Send a reply. Currently not available.
2 - Message Envelope. Speak the date and time at which the message was received.
* - Return to the Main menu.

* 4 - Play the previous message.

* 5 - Repeat the message.

* 6 - Play the next message.

* 7 - Delete the current message, without confirmation.

* 8 - Forward message to another user. When prompted for an extension, you must enter the full mailbox ID of the intended destination (e.g., 52739100).
1 - Prepend the message with a recording.
2 - Send the message without a prepending message.
* - Return to the Main Menu

== Known Voicemail issues or frequent questions ==

'''I can no longer listen to my Voicemail on my Android phone.'''

VoIP.ms uses a compressed WAV format for the voicemail messages (WAV49).

This helps server stability and lessen the bandwidth used by VoIP.ms customers to download the file.
Please download another FREE Audio Player like Remote Wave and this will resolve this issue instantly.
You will be given a choice of what application to use to listen to the file or you can make it your Default Audio Application.
If this doesn't work for you, you can always go back to uncompressed WAV format, on your Manage Mailbox options.

'''My Voicemail messages keep stopping at 3 minutes.'''

The Current Maximum Length, a voicemail message can be, is 3 minutes.

'''I keep receiving cut off Voicemails from people'''
While recording a message, the system listens for 5 seconds of silence. If the system detects no sounds for 5 seconds, it will
stop the VM Recording and send a person to VM options.

'''How can I delete all Voicemail messages in my mailbox.'''

You can only delete all messages from your Customer Portal>> DID Numbers>> Voicemail page. Please refer to [[Voicemail#Manage_Voicemail|Manage Voicemail]].

'''I have Voicemail messages but when I login the system tells me I have no Voicemails'''

It is possible that you have selected the option to Delete Voicemail Message once it has been sent to email.
Please see the [[Voicemail#Creating_a_Voicemail_mailbox|Creating a Voicemail mailbox Section]] to change this option.

'''I dial *97 but the system still asks for mailbox number'''
In order for *97 to work you have to make sure the device or softphone you are using is connecting to the servers with a SIP account associated with that mailbox.
So if you are connecting with your main account or a sub account, please see [[Voicemail#Assigning_your_Voicemail_to_your_Account|Assigning Your Voicemail to Your Account]].
If you are connecting with a phone not connected to VoIP.ms service then for security reasons you will have to enter the mailbox number and password.

[[category:guides]]

File:Create Voicemail.png

2020-07-08T10:07:42Z

Rod:

File:Transcription Complete.png

2020-07-08T09:39:51Z

Rod:

File:Transcription Successful.png

2020-07-08T09:37:17Z

Rod:

File:Transcription Interface.png

2020-07-08T09:36:42Z

Rod:

Elegir servidor

2017-05-03T13:05:55Z

Rod:

[[File:Serverlocation3.png]]

= Elegir un Servidor =

VoIP.ms le ofrece la posibilidad de conectarse a diferentes servidores, pero ¿cuál es el que debe elegir? Es común pensar que se debe elegir el servidor más cercano a su localidad, sin embargo esto no es necesario en la mayoría de las veces. Si usted está en Estados Unidos, cualquier servidor en EEUU le debe dar un buen resultado en latencia y calidad. También es de utilidad saber que hay una herramienta que le puede ayudar a seleccionar el mejor servidor a usar, este normalmente se llama "ping", el cual le provee la latencia entre usted y el servidor, así usted opcionalmente puede elegir específicamente el servidor que le provea una menor latencia.

Los servidores marcados como (NUEVO) están usando la versión más reciente de software, por lo tanto también le recomendamos usar cualquiera de ellos.

*Amsterdam, NL ('''amsterdam.voip.ms''') 37.58.88.242
*Atlanta 1, GA ('''atlanta.voip.ms''') 75.127.65.130
*Atlanta 2, GA ('''atlanta2.voip.ms''') 72.9.246.170
*Chicago 1, IL ('''chicago.voip.ms''') 208.100.39.52
*Chicago 2, IL ('''chicago2.voip.ms''') 208.100.39.53
*Chicago 3, IL ('''chicago3.voip.ms''') 208.100.39.54
*Chicago 4, IL ('''chicago4.voip.ms''') 208.100.39.55
*Dallas, TX ('''dallas.voip.ms''') 158.85.149.162
*Dallas 2, TX ('''dallas2.voip.ms''') 158.85.149.163
*Denver 1, CO ('''denver.voip.ms''') 173.248.161.90
*Denver 2, CO ('''denver2.voip.ms''') 173.248.159.210
*Houston, TX ('''houston.voip.ms''') 173.193.85.18
*Houston 2, TX ('''houston2.voip.ms''') 173.193.85.19
*London, UK ('''london.voip.ms''') 159.8.157.212
*Los Angeles 1, CA ('''losangeles.voip.ms''') 96.44.149.186
*Los Angeles 2, CA ('''losangeles2.voip.ms''') 96.44.149.202
*Melbourne, AU ('''melbourne.voip.ms''') 168.1.73.84
*Montreal 1, QC ('''montreal.voip.ms''') 67.205.74.184
*Montreal 2, QC ('''montreal2.voip.ms''') 67.205.74.187
*Montreal 3, QC ('''montreal3.voip.ms''') 192.175.96.68
*Montreal 4, QC ('''montreal4.voip.ms''') 67.205.74.179
*Montreal 5, QC ('''montreal5.voip.ms''') 192.175.96.69
*Montreal 6, QC ('''montreal6.voip.ms''') 192.175.96.70
*Montreal 7, QC ('''montreal7.voip.ms''') 192.175.96.71
*Montreal 8, QC ('''montreal8.voip.ms''') 192.175.96.72
*New York 1, NY ('''newyork.voip.ms''') 74.63.41.218
*New York 2, NY ('''newyork2.voip.ms''') 107.6.67.236
*New York 3, NY ('''newyork3.voip.ms''') 107.6.67.237
*New York 4, NY ('''newyork4.voip.ms''') 107.6.67.238
*New York 5, NY ('''newyork5.voip.ms''') 23.29.136.28
*New York 6, NY ('''newyork6.voip.ms''') 23.29.136.29
*New York 7, NY ('''newyork7.voip.ms''') 23.29.136.38
*New York 8, NY ('''newyork8.voip.ms''') 23.29.136.40
*Paris, FR ('''paris.voip.ms''') 159.8.85.180
*San Jose, CA ('''sanjose.voip.ms''') 23.246.247.146
*San Jose 2, CA ('''sanjose2.voip.ms''') 23.246.247.147
*Seattle 1, WA ('''seattle.voip.ms''') 50.23.160.53
*Seattle 2, WA ('''seattle2.voip.ms''') 50.23.149.166
*Seattle 3, WA ('''seattle3.voip.ms''') 50.23.160.54
*Tampa, FL ('''tampa.voip.ms''') 162.254.144.173
*Tampa 2, FL ('''tampa2.voip.ms''') 162.254.144.176
*Toronto 1, ON ('''toronto.voip.ms''') 158.85.70.148
*Toronto 2, ON ('''toronto2.voip.ms''') 158.85.70.149
*Toronto 3, ON ('''toronto3.voip.ms''') 158.85.70.150
*Toronto 4, ON ('''toronto4.voip.ms''') 158.85.70.151
*Toronto 5, ON ('''toronto5.voip.ms''') 184.75.215.106
*Toronto 6, ON ('''toronto6.voip.ms''') 184.75.215.114
*Toronto 7, ON ('''toronto7.voip.ms''') 184.75.215.146
*Toronto 8, ON ('''toronto8.voip.ms''') 184.75.213.210
*Vancouver 1, BC ('''vancouver.voip.ms''') 162.213.157.82
*Vancouver 2, BC ('''vancouver2.voip.ms''') 162.213.157.117
*Washington 1, DC ('''washington.voip.ms''') 208.43.234.226
*Washington 2, DC ('''washington2.voip.ms''') 208.43.234.227

= ¿Qué es un Ping? =

Ping es la herramienta estándar usada para probar la latencia (demora) en conexiones en red. Normalmente es usada para determinar si un servidor o dispositivo puede ser alcanzado a través de la red y su tiempo de latencia (tiempo que toma en enviarse un paquete a su destino y regresar a su origen) para responder.

La herramienta Ping es parte de Windows, MAC OS y Linux, también se puede encontrar en algunos ruteadores.

== ¿Cómo funciona la herramienta Ping? ==

Esta envía mensajes de petición a un una dirección de red determinada o nombres DNS en ciertos intervalos de tiempo y contabiliza el tiempo que tarda en recibirse un mensaje de respuesta. (mejor conocido como tiempo de latencia)

== ¿Cómo enviar un ping? ==

*Windows: Abrir una ventana de comandos DOS e introducir el comando ping xxxx.voip.ms (reemplace xxxx por el nombre del servidor que quiere probar. Por ejempo, montreal.voip.ms )

*Mac: Abrir una terminal y escribir ping xxxx.voip.ms(reemplace xxxx por el nombre del servidor que quiere probar. Por ejempo, montreal.voip.ms )

Si el ping resulta no ser consistente, puede tener un problema de ''jitter'' (variación del tiempo que tardan en llegar los paquetes debido a una congestión en la red o a diferentes rutas tomadas). Para tratar este problema, puede ajustar el nivel de ''jitter'' en su configuración del dispositivo VoIP (si cuenta con esta opción). Normalmente es recomendado un ping por debajo de 150ms para tener una buena calidad de audio durante la llamada. El tiempo de latencia del servidor es importante, sin embargo hay otros factores importantes a tomar en cuenta para la calidad de las llamadas como son la pérdida de paquetes (las comunicaciones VoIP son muy sensibles a esto) y el nivel de ''jitter'' de su conexión a Internet.

A continuación se presenta un ejemplo de ping con destino al servidor losangeles.voip.ms

#ping losangeles.voip.ms
Ping to losangeles.voip.ms [67.215.241.250] with 32 bytes de datos:
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
Response from 67.215.241.250: bytes=32 time=69ms TTL=52
Response from 67.215.241.250: bytes=32 time=68ms TTL=52
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
ping statistics from 67.215.241.250:
4 packets transmitted, 4 received, 0% packet lost. rtt min/avg/max/mdev = 67ms, 69ms, 67ms

= La Latencia y su importancia =

La latencia es muy importancia para VoIP, esto determina el tiempo que tarda en transmitirse un paquete de información a su destino. Una latencia alta le podría causar eco o retraso en la voz de sus llamadas.

La latencia es medida en milisegundos (ms). Por ejempo, una latencia de 150ms es apenas apreciable, por lo tanto es aceptable. Si es mas alta, la calidad de audio comienza a bajar. Cuando llega a 300 ms ya es inaceptable.

[[category:Guías]]

Choosing Server

2017-05-03T13:04:09Z

Rod:

[[File:Serverlocation3.png]]

= Choosing a Server =

[http://www.voip.ms VoIP.ms] offers many different servers, but which one should you choose? One misconception is that you should pick the closest to your location, however this is not needed most of the time. For example, if you are in the USA, any of the US servers will provide a really good latency and service quality. The newest server within a city is indicated with the highest number attached to the name, as they are classified in ascending order. Also worth noting is that there is a network tool that will help you when deciding which server you want to use, generally named a "ping", which will provide you the latency between you and the server. Therefore the server which provides you less latency should be used.

=== IPs ===

*Amsterdam, NL ('''amsterdam.voip.ms''') 37.58.88.242
*Atlanta 1, GA ('''atlanta.voip.ms''') 75.127.65.130
*Atlanta 2, GA ('''atlanta2.voip.ms''') 72.9.246.170
*Chicago 1, IL ('''chicago.voip.ms''') 208.100.39.52
*Chicago 2, IL ('''chicago2.voip.ms''') 208.100.39.53
*Chicago 3, IL ('''chicago3.voip.ms''') 208.100.39.54
*Chicago 4, IL ('''chicago4.voip.ms''') 208.100.39.55
*Dallas, TX ('''dallas.voip.ms''') 158.85.149.162
*Dallas 2, TX ('''dallas2.voip.ms''') 158.85.149.163
*Denver 1, CO ('''denver.voip.ms''') 173.248.161.90
*Denver 2, CO ('''denver2.voip.ms''') 173.248.159.210
*Houston, TX ('''houston.voip.ms''') 173.193.85.18
*Houston 2, TX ('''houston2.voip.ms''') 173.193.85.19
*London, UK ('''london.voip.ms''') 159.8.157.212
*Los Angeles 1, CA ('''losangeles.voip.ms''') 96.44.149.186
*Los Angeles 2, CA ('''losangeles2.voip.ms''') 96.44.149.202
*Melbourne, AU ('''melbourne.voip.ms''') 168.1.73.84
*Montreal 1, QC ('''montreal.voip.ms''') 67.205.74.184
*Montreal 2, QC ('''montreal2.voip.ms''') 67.205.74.187
*Montreal 3, QC ('''montreal3.voip.ms''') 192.175.96.68
*Montreal 4, QC ('''montreal4.voip.ms''') 67.205.74.179
*Montreal 5, QC ('''montreal5.voip.ms''') 192.175.96.69
*Montreal 6, QC ('''montreal6.voip.ms''') 192.175.96.70
*Montreal 7, QC ('''montreal7.voip.ms''') 192.175.96.71
*Montreal 8, QC ('''montreal8.voip.ms''') 192.175.96.72
*New York 1, NY ('''newyork.voip.ms''') 74.63.41.218
*New York 2, NY ('''newyork2.voip.ms''') 107.6.67.236
*New York 3, NY ('''newyork3.voip.ms''') 107.6.67.237
*New York 4, NY ('''newyork4.voip.ms''') 107.6.67.238
*New York 5, NY ('''newyork5.voip.ms''') 23.29.136.28
*New York 6, NY ('''newyork6.voip.ms''') 23.29.136.29
*New York 7, NY ('''newyork7.voip.ms''') 23.29.136.38
*New York 8, NY ('''newyork8.voip.ms''') 23.29.136.40
*Paris, FR ('''paris.voip.ms''') 159.8.85.180
*San Jose, CA ('''sanjose.voip.ms''') 23.246.247.146
*San Jose 2, CA ('''sanjose2.voip.ms''') 23.246.247.147
*Seattle 1, WA ('''seattle.voip.ms''') 50.23.160.53
*Seattle 2, WA ('''seattle2.voip.ms''') 50.23.149.166
*Seattle 3, WA ('''seattle3.voip.ms''') 50.23.160.54
*Tampa, FL ('''tampa.voip.ms''') 162.254.144.173
*Tampa 2, FL ('''tampa2.voip.ms''') 162.254.144.176
*Toronto 1, ON ('''toronto.voip.ms''') 158.85.70.148
*Toronto 2, ON ('''toronto2.voip.ms''') 158.85.70.149
*Toronto 3, ON ('''toronto3.voip.ms''') 158.85.70.150
*Toronto 4, ON ('''toronto4.voip.ms''') 158.85.70.151
*Toronto 5, ON ('''toronto5.voip.ms''') 184.75.215.106
*Toronto 6, ON ('''toronto6.voip.ms''') 184.75.215.114
*Toronto 7, ON ('''toronto7.voip.ms''') 184.75.215.146
*Toronto 8, ON ('''toronto8.voip.ms''') 184.75.213.210
*Vancouver 1, BC ('''vancouver.voip.ms''') 162.213.157.82
*Vancouver 2, BC ('''vancouver2.voip.ms''') 162.213.157.117
*Washington 1, DC ('''washington.voip.ms''') 208.43.234.226
*Washington 2, DC ('''washington2.voip.ms''') 208.43.234.227

===Server Realms===

For IOS, Please click here [http://wiki.voip.ms/article/Server_Realms Server Realms] to get the Realm Name for the server you plan on using, this can differ from the Domain Name being used.

Please note that the following servers will not be available to select as a DID Point of Presence for newer accounts:
Atlanta 1, Chicago 1, New York 1, New York 4, Seattle 1, Montreal 1, Montreal 2, Montreal 3, Montreal 4, Toronto, Toronto 2, Toronto 3, Toronto 4.

= What is a Ping? =

Ping is a standard tool used to test network connections. It is mostly used to determine if a server or device can be reached across the network and the latency of the response(the time it takes to send a packet to the destination and for it to return to your computer).

Ping tools are part of Windows, Mac OS X and Linux as well as some routers.

== How does the ping work? ==

It sends request messages to a target network address or DNS names at periodic intervals and measures the time it takes for a response message to arrive and return(better known as latency).

==How to ping on a PC==

Pinging is a command which tells you if the connection between your computer and a particular domain is working correctly.

In Windows, select Start > Programs > Accessories > Command Prompt. This will give you a window like the one below.

Enter the word ping, followed by a space, then the domain name.(montreal.voip.ms) in this case domain is our server name.

If the results show a series of replies, the connection is working. The time shows you how fast the connection is. If you see a "timed out" error instead of a reply, there is a breakdown somewhere between your computer and the domain.

[[File:Ping.gif]]

==How to ping on a Mac Computer==

1- Click on Finder in the dock.

2- Click on Applications.

3- Click on Utilities.

4- Double-click on Network Utility. *

* In OS X Mavericks (10.9.x) this utility app changed location. Launch it from spotlight instead, either press "command"+"space bar" or click on spotlight directly (magnifying glass icon at top right of screen), type "network utility" and hit "return"

5- In the Network Utility window, click on the Ping tab

6- In the field under "Please enter the network address to ping," like montreal.voip.ms

''If pings results are not consistent, you may have an issue with Jitter. You can work on this issue by adjusting the "Network Jitter Level" setting on your VoIP device. Usually a ping of under 150 ms is recommended in order to have good quality. The latency time to the server is important, however there are also other factors that could affect the quality of the calls such as packet loss (VoIP communications are very sensitive to this), and the Jitter level of your Internet connection.''

The following is the output of running ping with the target losangeles.voip.ms.

#ping losangeles.voip.ms
Ping to losangeles.voip.ms [67.215.241.250] with 32 bytes de datos:
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
Response from 67.215.241.250: bytes=32 time=69ms TTL=52
Response from 67.215.241.250: bytes=32 time=68ms TTL=52
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
ping statistics from 67.215.241.250:
4 packets transmitted, 4 received, 0% packet lost. rtt min/avg/max/mdev = 67ms, 69ms, 67ms

Sample ping output in windows:
C:\Windows\system32>ping montreal.voip.ms

Pinging montreal.voip.ms [67.205.74.184] with 32 bytes of data:
Reply from 67.205.74.184: bytes=32 time=85ms TTL=49
Reply from 67.205.74.184: bytes=32 time=86ms TTL=49
Reply from 67.205.74.184: bytes=32 time=86ms TTL=49
Reply from 67.205.74.184: bytes=32 time=85ms TTL=49

Ping statistics for 67.205.74.184:
Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 85ms, Maximum = 86ms, Average = 85ms

= Latency Testing Scripts (User Submitted) =
<p>All the following scripts were produced by voip.ms users who felt others might also benefit from the output of their efforts. They were written over a span of Years and probably need adjusting before you use them, to cater for changes in servers over time and changes in policies (like not testing heavily subscribed servers which are not open to new registrations)</p>
<p>If you aren't satisfied that the scripts are safe or simply don’t like the way they syntactically appear, you can still manually ping a selection of servers and choose a server based on the best latency. The following scripts are essentially just wrappers around the ping command which support lists of servers to feed to ping and present the output in a readable format.</p>
<p>If you feel you have a simpler cleaner script that works for another platform or language, please do add it to this wiki via a support ticket.
</p>
=== Bash Script To Handle The Mac Ping Output Format ===

<p>To make use of this script (1) save as a plain text file (2) set permissions on the file to executable (3) invoke script
e.g. Save script below using your favourite editor as pingVoipMS.sh (2) chmod u+x pingVoipMS.sh (3) ./pingVoipMS.sh
This is a bash 3.x script, so it also works in Linux, just change the ping packet loss field from 7 to 6 in the final loop below (or wherever the loss field is in your ping output format). Depending upon your distro curl might need to change to wget.
</p>
<pre>
#!/bin/bash
# Ping several servers and display Latency, Jitter and Packet Loss
# Usage: [-c <count>][-i <wait time>][-r test restricted servers][<server list file>]
#
# The optional text file should be formatted with one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
# If no args are supplied, this script will scrape a ping server list from voip.ms
#
USER_FILE=""
COUNT=3; INTERVAL=5; RESTRICTED=0
restrictedList=(atlanta.voip.ms chicago.voip.ms
montreal.voip.ms montreal2.voip.ms montreal3.voip.ms montreal4.voip.ms
newyork.voip.ms newyork4.voip.ms seattle.voip.ms
toronto.voip.ms toronto2.voip.ms toronto3.voip.ms toronto4.voip.ms)

# Handle any passed in script arguments
while getopts c:i:r parm
do
case $parm in
c)count_opt=$OPTARG;;
i)interval_opt=$OPTARG;;
r)RESTRICTED=1;;
*)echo -e "Invalid arg\nUsage:\t[ -c <count of ECHO_REQUESTs to Tx, default 3> ] \
\n\t[ -i <wait time (s) between datagrams, default 5> ] \
\n\t[ -r ] Include restricted servers in latency test \
\n\t[FILE <ping server list> ]";exit 1;;
esac
done

# Test if an option was specified and whether it's a +ve non-zero integer
[[ -n $count_opt && ($count_opt =~ ^[[:digit:]]+$) && $count_opt -gt 0 ]] && COUNT=$count_opt
[[ -n $interval_opt && ($interval_opt =~ ^[[:digit:]]+$) && $interval_opt -gt 0 ]] && INTERVAL=$interval_opt

shift $(($OPTIND -1))

# Validate supplied file (server list)
[[ -n $1 && !(-f $1 && -r $1) ]] && { echo "\"$1\" file does not exist or is not readable"; exit 1; }
[[ -n $1 && -f $1 && -r $1 ]] && USER_FILE="$1"

if [[ -n $USER_FILE ]]
then
# Bash 3.x in macOS does not support readarray, need to do cumbersome array loops instead
while IFS= read -r line; do
serverList+=( "$line" )
done < <(grep -e '[[:alpha:]]*[[:alnum:]]\.voip\.ms' $USER_FILE | grep -v '^\s*#' | awk NF)
else
# N.B. The script looks for the html boldface tags <b> </b> inside a bracket
# If the website alters and the parse fails, manually create the list and
# supply as a script arg (or perhaps update the parsing to work again :)
while IFS= read -r line; do
serverList+=( "$line" )
done < <(curl --silent http://wiki.voip.ms/article/Choosing_Server | \
grep '(<b>[[:alpha:]]*[[:alnum:]]\.voip\.ms</b>)' | \
tr "<>" " " | awk '{print $(NF-3)}' )
fi

# Newer voip.ms clients can't register onto these over-subscribed servers
# Don't test the restricted list unless explicitly asked (with the -r cmd line option)
if [[ $RESTRICTED -eq 0 ]]
then
for server in ${restrictedList[@]}
do
ix=$(printf "%s\n" "${serverList[@]}" | grep -n "^${server}" | cut -d ":" -f1)

while IFS= read -r idx; do
keys+=($idx)
done < <([[ $ix -gt 0 ]] && echo $((ix-1)))
done
for ((i=${#keys[@]} - 1; i >= 0; i--)); do unset serverList[keys[i]]; done
fi

if [[ ${#serverList[@]} -eq 0 ]]
then
echo "No unrestricted Voip.ms servers could be found, please supply a server list"
exit 1
fi

runTime=$(( $COUNT * $INTERVAL * ${#serverList[@]} ))

echo "PING will send $COUNT packet(s) with a wait of $INTERVAL sec(s) between each packet"
echo "Change the PING options by invoking this script with -c and/or -i, default \"-c 3 -i 5\""
echo "Over $((${#serverList[@]})) server(s) the estimated script Run Time will be $runTime seconds"
echo "================================================================"
printf "%-20s %-18s %7s %8s %6s %s\n" "VoIP Server" "IP Address" "Latency" "Jitter" "Loss" "Countdown"
echo "================================================================ (seconds)"

for myLn in ${serverList[@]}
do
while IFS=$'\n' read -r line; do
pingList+=( "$line" )
printf "%-64s %5d %2d/%-2d\n" "$line" "$((runTime - $COUNT * $INTERVAL * ${#pingList[@]}))" "${#pingList[@]}" "${#serverList[@]}"
done < <( ping -c $COUNT -i $INTERVAL -q $myLn | awk '\
/^PING / {myH=$2}
/^PING / {
IP = substr($3,2,15)
split(IP,myIP,")") }
/packet loss/ {myPL=$7}
/min\/avg\/max/ {
split($4,myS,"/")
printf("%-20s %-18s %7.3f %8.3f %6s\n",
myH, myIP[1], myS[2], myS[4], myPL ) }
' )
done

echo "================================================================"
echo -e "\nMost appropriate server listed in order of best latency\n"
echo "================================================================"
printf "%-20s %-18s %7s %8s %6s\n" "VoIP Server" "IP Address" "Latency" "Jitter" "Loss"
echo "================================================================"
printf "%s\n" "${pingList[@]}" | LC_ALL=C sort -n -k 3,3 -k 5,5 -k 4,4 | awk '{printf("%s $%2d$\n",$0, NR)}'
echo "================================================================"
</pre>

=== Perl Script ===
Pings list of voip.ms servers round robin with optional output csv file.

# usage ping_voip.ms.pl <number of times> <seconds in between> <output.csv>
use Net::Ping;
use Time::HiRes;
use strict;

# input list
my @hosts = qw(
atlanta.voip.ms
atlanta2.voip.ms
chicago.voip.ms
chicago2.voip.ms
chicago3.voip.ms
chicago4.voip.ms
dallas.voip.ms
denver.voip.ms
denver2.voip.ms
houston.voip.ms
losangeles.voip.ms
losangeles2.voip.ms
newyork.voip.ms
newyork2.voip.ms
newyork3.voip.ms
newyork4.voip.ms
seattle.voip.ms
seattle2.voip.ms
seattle3.voip.ms
tampa.voip.ms
washington.voip.ms
washington2.voip.ms
montreal.voip.ms
montreal2.voip.ms
montreal3.voip.ms
montreal4.voip.ms
toronto2.voip.ms
toronto3.voip.ms
toronto4.voip.ms
toronto.voip.ms
london.voip.ms
);

$| = 1; #autoflush
# High precision syntax (requires Time::HiRes)
my $p = Net::Ping->new("icmp",1);
$p->hires();
my $max_name_length = (reverse sort { $a <=> $b } map { length($_) } @hosts)[0];
my $count = 4; # number of times to ping
my $interval = 5; # seconds between ping rounds
my $output_file = "";
my @data;

# check for arguments
my $num_args = @ARGV;
if ($num_args >= 1) {$count = $ARGV[0];}
if ($num_args >= 2) {$interval = $ARGV[1];}
if ($num_args >= 3) {$output_file = $ARGV[2];}

# check argument validity
$0 =~ /^.*\\(.*)$/;
my $script = $1;
if ($count !~ /^\d+$/ or $interval !~ /^\d+$/) {die "Usage: $script <number of rounds> <seconds between rounds> <output.csv>\n";}
if (length($output_file) > 0 and $output_file !~ /\.csv$/) {$output_file .= ".csv";}

# main loop
for my $i (1..$count)
{
sleep $interval unless $i == 1;
print "Round $i\n";
my $host_num=0;
foreach my $host (@hosts)
{
(my $ret, my $duration, my $ip) = $p->ping($host);
$ip =~ /(\d+)\.(\d+)\.(\d+)\.(\d+)/;
if ($ret)
{
printf("%*s [ip: %3s.%3s.%3s.%3s] is alive (%6.2f ms)\n", $max_name_length, $host, $1, $2, $3, $4, $duration*1000);
$data[$host_num][$i]=$duration*1000;
}
else
{
printf("%*s [ip: %3s.%3s.%3s.%3s] is dead\n", $max_name_length, $host, $1, $2, $3, $4);
}
$host_num++;
}
print "\n";
}
$p->close();

# if output file name given
if (length($output_file)>0)
{
# print output to file
open FILE, ">$output_file" or die "$!\n";

# print column headers
print FILE "Server\\Round";
for my $i (1..$count)
{
print FILE ", $i";
}
print FILE ", Average\n";

# print data
my $i = 0;
foreach my $host (@hosts)
{
print FILE "$host";
my $sum = 0;
for my $j (1..$count)
{
$sum += $data[$i][$j];
printf FILE ", %8.4f",$data[$i][$j];
}
printf FILE ", %8.4f\n",$sum/$count;
$i++;
}

close FILE;
print "Data written to $output_file\n";
}

# print summary to screen
my $i = 0;
printf("%-*s Average (ms)\n", $max_name_length, "Server");
foreach my $host (@hosts)
{
my $sum = 0;
for my $j (1..$count)
{
$sum += $data[$i][$j];
}
printf("%-*s %8.4f\n", $max_name_length+1, $host, $sum/$count);
$i++;
}

Output:
Round 1
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.97 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.99 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.70 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.76 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.53 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.73 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 94.99 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.05 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.13 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (102.87 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 64.92 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 63.41 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (131.75 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (120.64 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (120.49 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (111.43 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.25 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.86 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 90.85 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (123.29 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.71 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.19 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 81.82 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 86.13 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 77.09 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.18 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.70 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (131.27 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (125.13 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.26 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (152.77 ms)

Round 2
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.14 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.86 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 50.03 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.44 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.33 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.22 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 95.58 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.94 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.29 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (102.73 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.59 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 64.27 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (112.74 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (121.22 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (121.34 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.75 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.06 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.33 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 91.58 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (122.94 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.28 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.40 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 81.91 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 85.64 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 75.15 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.79 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.10 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (150.85 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.40 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.45 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (170.79 ms)

Round 3
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.76 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.86 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.65 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.01 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.05 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.53 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 95.82 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.02 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.60 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (103.35 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.79 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 64.05 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (113.01 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (121.41 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (122.23 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.62 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 93.65 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.19 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 90.75 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (125.12 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.19 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.98 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 80.16 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 87.16 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 76.54 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 97.51 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.18 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (142.81 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.95 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.78 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (153.14 ms)

Round 4
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 89.19 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.98 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.21 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.50 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.68 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.18 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 93.93 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.22 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.10 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (103.67 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.58 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 63.60 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (114.76 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (120.44 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (121.05 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.51 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.04 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 96.92 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 91.23 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (123.28 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.45 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (100.94 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 82.33 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 85.02 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 76.85 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.32 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.22 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (148.33 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (141.61 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (105.91 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (152.85 ms)

Server Average (ms)
atlanta.voip.ms 88.7630
atlanta2.voip.ms 92.9233
chicago.voip.ms 49.6477
chicago2.voip.ms 59.9305
chicago3.voip.ms 59.3972
chicago4.voip.ms 49.9152
dallas.voip.ms 95.0790
denver.voip.ms 94.8077
denver2.voip.ms 85.2797
houston.voip.ms 103.1562
losangeles.voip.ms 65.4693
losangeles2.voip.ms 63.8347
newyork.voip.ms 118.0643
newyork2.voip.ms 120.9265
newyork3.voip.ms 121.2778
newyork4.voip.ms 110.8275
seattle.voip.ms 93.9993
seattle2.voip.ms 95.8267
seattle3.voip.ms 91.1035
tampa.voip.ms 123.6570
washington.voip.ms 98.4065
washington2.voip.ms 101.3774
montreal.voip.ms 81.5525
montreal2.voip.ms 85.9863
montreal3.voip.ms 76.4058
montreal4.voip.ms 96.7013
toronto2.voip.ms 103.7986
toronto3.voip.ms 143.3156
toronto4.voip.ms 136.0254
toronto.voip.ms 104.1012
london.voip.ms 157.3885

=== Powershell ===

This code was optimized by a Customer (Jean Lem), server's list was also updated.

<pre>
# Usage: Copy and paste the following code into a powershell window
# To run it from a command prompt, save this file with extension ps1. Then run Powershell.exe -file "pathtothisscript.ps1"
#Get the list of servers into an array
$Servers =
@("amsterdam.voip.ms","atlanta.voip.ms","atlanta2.voip.ms","chicago.voip.ms","chicago2.voip.ms","chicago3.voip.ms","chicago4.voip.ms","dallas.voip.ms","dallas2.voip.ms","denver.voip.ms","denver2.voip.ms","houston.voip.ms","houston2.voip.ms","london.voip.ms","losangeles.voip.ms","losangeles2.voip.ms","melbourne.voip.ms","montreal.voip.ms","montreal2.voip.ms","montreal3.voip.ms","montreal4.voip.ms","montreal5.voip.ms","montreal6.voip.ms","montreal7.voip.ms","montreal8.voip.ms","newyork.voip.ms","newyork2.voip.ms","newyork3.voip.ms","newyork4.voip.ms","newyork5.voip.ms","newyork6.voip.ms","newyork7.voip.ms","newyork8.voip.ms","paris.voip.ms","sanjose.voip.ms","sanjose2.voip.ms","seattle.voip.ms","seattle2.voip.ms","seattle3.voip.ms","tampa.voip.ms","tampa2.voip.ms","toronto.voip.ms","toronto2.voip.ms","toronto3.voip.ms","toronto4.voip.ms","toronto5.voip.ms","toronto6.voip.ms","toronto7.voip.ms","toronto8.voip.ms","vancouver.voip.ms","vancouver2.voip.ms","washington.voip.ms","washington2.voip.ms")
$k = 0 #Counting variable so we know what server number we are testing
#num of servers to test
$servercount = $servers.length
#Do the following code for each server in our array
ForEach($server in $servers){
$k++ #Add one to the counting variable....we are on server #1...then server 2, then server 3 etc...
Write-Progress -Activity "Testing Server: ${server}" -status "Testing Server $k out of $servercount" -percentComplete ($k /
$servercount*100) #Update the progress bar
$i = 0 #Counting variable for number of times we tried to ping a given server
Do{
$pingsuccess = $false #assume a failure
$i++ #Add one to the counting variable.....1st try....2nd try....3rd try etc...
Try{
$currentping = (test-connection $server -count 1 -ErrorAction Stop).responsetime #Try to ping
$pingsuccess = $true #If success full, set success variable
}Catch { $pingsuccess = $false } #Catch the failure and set the success variable to false
}While($pingsuccess -eq $false -and $i -le 5) #Try everything between Do and While up to 5 times, or while $pingsuccess is not true
If($pingsuccess -and ($currentping -lt $bestping -or (!($bestping)))){ #Compare the last ping test with the best known ping test....if there is no known best ping test, assume this one is the best $bestping = $currentping #If this is the best ping...save it
$bestserver = $server #Save the best server
}
write-host "tested: $server at $currentping ms after $i attempts" #write the results of the test for this server
}
write-host "The server with the best ping is: $bestserver at $bestping ms" #write the end result
Pause
</pre>

=== Linux Shell Script ===
Pings several voip.ms servers

#!/bin/sh
# Ping several servers and display Latency, Jitter and Packet Loss
#
# First, create a text file with all servers you want to ping - one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
myHF="voip_ping_hosts.txt"
# Sample file:
# toronto.voip.ms
# montreal.voip.ms
# seattle.voip.ms
# chicago.voip.ms
# newyork.voip.ms
#
echo "============================================"
printf "%-20s %7s %8s %6s\n" "VoIP Server" "Latency" "Jitter" "Loss"
echo "============================================"
cat ${myHF} |\
while read myLn
do
ping -c 3 -i 5 -q $myLn |\
awk '/^PING / {myH=$2}
/packet loss/ {myPL=$6}
/min\/avg\/max/ {
split($4,myS,"/")
printf( "%-20s %3.1f %1.3f %4s\n", myH, myS[2], myS[4], myPL)
}'
done
echo "============================================"

Output:

============================================
VoIP Server Latency Jitter Loss
============================================
toronto.voip.ms 68.3 0.439 0%
montreal.voip.ms 89.6 0.197 0%
seattle.voip.ms 71.2 0.387 0%
chicago.voip.ms 71.6 0.084 0%
newyork.voip.ms 79.1 0.411 0%
============================================

= Latency and its importance =

Latency is very important for Voip, this will determine the time that will take for the data package transmission to reach the destination. A high latency will lead to a delay and echoes in the communication.

Latency is measured in milliseconds (ms) For example: a latency of 150ms is barely noticeable, thus acceptable. Higher than that, quality starts to suffer. When it gets higher than 300 ms, it becomes unacceptable.

[[category:guides]]

SIP URI

2017-04-25T18:04:22Z

Rod:

A SIP URI is the SIP addressing schema to call another person via SIP. In other words, a SIP URI is a user's SIP phone number. The SIP URI resembles an e-mail address and is written in the following format: x@y:port (x=Username, y=host|domain|IP)

: Example: johnsmith@my-uri.com

A general description of SIP addressing is at [[wikipedia:SIP address]]. The addresses, which use the same user@domain... format as e-mail addresses, allow an individual Internet telephony user to be reached directly online without passing via the public switched telephone network or incurring the corresponding tolls.

A SIP address may be used as a destination to which to forward a voip.ms DID number, as a target for an individual speed dial entry (*75xx) in a voip.ms user address book or as a means to transfer incoming calls into your voip.ms extensions or numbers from outside Internet servers.

# One is to send calls to an external SIP URI, via your DID number,
# A second option is to receive calls via SIP URI, we can achieve this using our DID number or an internal extension from a [[Sub Accounts|sub account]].
# The third option is to use a Virtual number.

Please note SIP URI is only functional through an external source, it is not intended to be used internally with VoIP.ms.

__TOC__

== Send calls to an external SIP URI address ==

You can route incoming calls to your DID numbers using a SIP URI address from other companies. You will need to create a string like youraccount@yourip to which you can route the DID(s).

: '''Make sure the other company or provider supports the use of SIP URI'''

=== Creating a new SIP URI ===
To make a SIP URI that can be used with multiple DIDs, use the {DID} tag. The {DID} expression will be replaced automatically by the DID number as it is listed in the "Manage DID section".

[[Image:Forward.jpg]]

; Examples:
: 1{DID}@128.144.122.12
: 12143221234@128.144.122.12
: some_extension_name@128.144.122.12:5080
: other_extension_name@voip.example.com

''Dialing out a SIP URI or using a SIP URI as forwarding does not generate cost (Outgoing call).''

=== Creating a phone book entry ===
A SIP URI may be associated with a [[phone book]] or speed dial entry in the same manner as any other telephone number.

See [[Phone book#Create a Phone Book Entry with a SIP URI]].

[[File:Pb entry sipuri.jpg|800px]]

'''SIP URI''': Here you can either select '''Use Existing''' or '''Create New''' to assign the [[SIP URI]] to your Phone Book entry.

This replaces alphanumeric addresses (such as sip:user@provider.example.org) with numeric abbreviations (such as *7501) which can be easily dialled from [[devices|IP phones]] that only offer a numeric keypad.

=== Codec Negotation ===

By default when you route your incoming calls to an external SIP URI address, the system sends the INVITE allowing all VoIP.ms supported codecs (ulaw, g729a and GSM).
In that case if you want to use a specific codec (from the supported ones) you need to restrict that in your end. For instance, if you are using an Asterisk/PBX System and only wish to use ulaw codec, you will need to make sure to have the following settings in the trunk:

: disallow=all
: allow=ulaw

== Receiving incoming calls from a SIP URI ==

=== Using your DID number ===
You can receive SIP URI calls using the following format {Number}@sip.voip.ms, this can be used with your local US and Canada numbers, so they can be reached from outside.

[[Image:Did.jpg]]

This format of SIP address is used by services such as [[wikipedia:SIP Broker]] as a means to reach voip.ms subscribers.

''' We have been notified that SIP Broker service is not working correctly. This is a service we have no control over.'''

Another variant, also valid, is to specify the specific voip.ms server on which your DID is registered, ie:
:sip:4166471234@toronto.voip.ms

'''Please note that the option to dial sip.voip.ms is more reliable than using the server as you won't have to specify the Point of Presence. Using the server instead, you will have to dial the correct server that the number is using.'''

=== Using your sub account internal extension ===
When you assign an internal extension for a [[Sub Accounts|sub account]], it can also be used as an external SIP URI. For example, if your extension is 2, you could be reached directly via SIP from another network with a URI like this: 1000002@houston.voip.ms

:(Replace houston.voip.ms by the server you are registered to, 100000 by your account ID and the 2 by your internal extension).

Important: no call flow or filtering can be applied to calls make to the external SIP URI. Calls will immediately ring the device registered to this sub-account.

[[Image:Extension.jpg]]

=== Using iNum ===
Any iNum number (from any provider) is a SIP URI; just append @sip.inum.net

For example, iNum 883510009999999 becomes:
:883510009999999@sip.inum.net

=== Using a Virtual number ===

Virtual SIP numbers are similar to standard DID numbers. The major difference is that virtual SIP numbers are not accessible via "PSTN". They can only be reached via "SIP URI" over internet. For example, if you have a DID number with another provider and they support SIP URI Forwarding, you could forward your number to a virtual number at voip.ms just like if it was one of our numbers.

All virtual numbers consist of the following digits: 11 + Accountcode + 3 digits of your choice for a total of 11 digits. The final uri will be that number followed by the @ sign at one of our server. If you intend to send the calls to a phone or adapter, you'll need to point it to the proper server.

: Example SIP URI: 11100000123@houston.voip.ms

[[File:Virtualsip.jpg]]

[[category:guides]]

SIP URI

2017-04-25T17:50:16Z

Rod:

A SIP URI is the SIP addressing schema to call another person via SIP. In other words, a SIP URI is a user's SIP phone number. The SIP URI resembles an e-mail address and is written in the following format: x@y:port (x=Username, y=host|domain|IP)

: Example: johnsmith@my-uri.com

A general description of SIP addressing is at [[wikipedia:SIP address]]. The addresses, which use the same user@domain... format as e-mail addresses, allow an individual Internet telephony user to be reached directly online without passing via the public switched telephone network or incurring the corresponding tolls.

A SIP address may be used as a destination to which to forward a voip.ms DID number, as a target for an individual speed dial entry (*75xx) in a voip.ms user address book or as a means to transfer incoming calls into your voip.ms extensions or numbers from outside Internet servers.

# One is to send calls to an external SIP URI, via your DID number,
# A second option is to receive calls via SIP URI, we can achieve this using our DID number or an internal extension from a [[Sub Accounts|sub account]].
# The third option is to use a Virtual number.

Please note SIP URI is only functional through an external source, it is not intended to be used internally with VoIP.ms.

__TOC__

== Send calls to an external SIP URI address ==

You can route incoming calls to your DID numbers using a SIP URI address from other companies. You will need to create a string like youraccount@yourip to which you can route the DID(s).

: '''Make sure the other company or provider supports the use of SIP URI'''

=== Creating a new SIP URI ===
To make a SIP URI that can be used with multiple DIDs, use the {DID} tag. The {DID} expression will be replaced automatically by the DID number as it is listed in the "Manage DID section".

[[Image:Forward.jpg]]

; Examples:
: 1{DID}@128.144.122.12
: 12143221234@128.144.122.12
: some_extension_name@128.144.122.12:5080
: other_extension_name@voip.example.com

''Dialing out a SIP URI or using a SIP URI as forwarding does not generate cost (Outgoing call).''

=== Creating a phone book entry ===
A SIP URI may be associated with a [[phone book]] or speed dial entry in the same manner as any other telephone number.

See [[Phone book#Create a Phone Book Entry with a SIP URI]].

[[File:Pb entry sipuri.jpg|800px]]

'''SIP URI''': Here you can either select '''Use Existing''' or '''Create New''' to assign the [[SIP URI]] to your Phone Book entry.

This replaces alphanumeric addresses (such as sip:user@provider.example.org) with numeric abbreviations (such as *7501) which can be easily dialled from [[devices|IP phones]] that only offer a numeric keypad.

=== Codec Negotation ===

By default when you route your incoming calls to an external SIP URI address, the system sends the INVITE allowing all VoIP.ms supported codecs (ulaw, g729a and GSM).
In that case if you want to use a specific codec (from the supported ones) you need to restrict that in your end. For instance, if you are using an Asterisk/PBX System and only wish to use ulaw codec, you will need to make sure to have the following settings in the trunk:

: disallow=all
: allow=ulaw

== Receiving incoming calls from a SIP URI ==

=== Using your DID number ===
You can receive SIP URI calls using the following format {Number}@sip.voip.ms, this can be used with your local US and Canada numbers, so they can be reached from outside.

[[Image:Did.jpg]]

This format of SIP address is used by services such as [[wikipedia:SIP Broker]] as a means to reach voip.ms subscribers.

''' We have been notified that SIP Broker service is not working correctly. This is a service we have no control over.'''

Another variant, also valid, is to specify the specific voip.ms server on which your DID is registered, ie:
:sip:4166471234@toronto.voip.ms

=== Using your sub account internal extension ===
When you assign an internal extension for a [[Sub Accounts|sub account]], it can also be used as an external SIP URI. For example, if your extension is 2, you could be reached directly via SIP from another network with a URI like this: 1000002@houston.voip.ms

:(Replace houston.voip.ms by the server you are registered to, 100000 by your account ID and the 2 by your internal extension).

Important: no call flow or filtering can be applied to calls make to the external SIP URI. Calls will immediately ring the device registered to this sub-account.

[[Image:Extension.jpg]]

=== Using iNum ===
Any iNum number (from any provider) is a SIP URI; just append @sip.inum.net

For example, iNum 883510009999999 becomes:
:883510009999999@sip.inum.net

=== Using a Virtual number ===

Virtual SIP numbers are similar to standard DID numbers. The major difference is that virtual SIP numbers are not accessible via "PSTN". They can only be reached via "SIP URI" over internet. For example, if you have a DID number with another provider and they support SIP URI Forwarding, you could forward your number to a virtual number at voip.ms just like if it was one of our numbers.

All virtual numbers consist of the following digits: 11 + Accountcode + 3 digits of your choice for a total of 11 digits. The final uri will be that number followed by the @ sign at one of our server. If you intend to send the calls to a phone or adapter, you'll need to point it to the proper server.

: Example SIP URI: 11100000123@houston.voip.ms

[[File:Virtualsip.jpg]]

[[category:guides]]

Softphones

2017-04-25T13:02:30Z

Rod:

A softphone is a software program for making telephone calls over the Internet using a general purpose computer, tablet or smartphone, rather than using dedicated hardware. Often a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a phone with a display panel and buttons with which the user can interact. A softphone is usually used with a headset connected to the sound card of the PC, or as an installed app on your phone. See also: [[Smartphone]]

==Acrobits==

[[File:AcrobitsLogo.png|left|Acrobits]]

'''Product:''' Acrobits

'''Company:''' Acrobits

'''OS:''' IOS, Android

'''Overview:'''
Acrobits is a mobile software development company with a focus on SIP Clients for mobile devices; their SIP client, Acrobits Softphone, is the leading app of it's type on the iTunes App Store.

[[Acrobits|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

== Android Sip Client ==

[[File:android_sip_client.jpg|left]]

'''Product:''' Android

'''Company:''' Google Android

'''OS:''' Android

'''Overview:''' Android SIP Client that is included with Android 4.1.1. We cannot guarantee 100% compatibility with all future Android releases since settings may change. We recommend the use of a Wi-Fi connection with these features, instead of a 3G connection. You also could be billed for your data usage if used over 3G/EDGE.

[[Android_Sip_Client|See Configuration Details]]

==Bria==

[[File:Bria.jpg|left|Bria]]

'''Product:''' Bria

'''Company:''' CounterPath

'''OS:''' Windows, MacOS, Linux (Ubuntu), IOS, Android

'''App Storage Consumption:''' 6.11mb (intl. mem only- SD card unusable. Data uses an additional ~2mb approx.)

'''Overview:'''
Bria 3 is a carrier-grade next generation softphone application that enables you to manage your communications easily and efficiently – all from your computer desktop. Replacing or complementing your hard phone, the Bria softphone allows you to make VoIP and Video calls over IP, see when your contacts are available, send Instant Messages and transfer files with ease and efficiency.

Bria Android Edition is a highly secure, standards-based mobile VoIP softphone that works over both 3G and Wi-Fi networks.

Using the device’s existing contact list, Bria Android Edition facilitates easy and effective communication management with an intuitive interface. Deskphone-class calling functionality includes the ability to swap between two calls, merge calls and perform attended and unattended transfers.

[[Bria|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==CSipSimple==

[[File:Csip-0.png|200px|thumb|left|CSipSimple]]

'''Product:''' CSipSimple

'''Company:''' CSipSimple

'''OS:''' Android

'''Overview:'''
CSipSimple is a Voice over Internet Protocol (VoIP) application for Google Android operating system using the SIP.

CSipSimple is an open source and free software released under the GNU General Public License.

It relies on the pjsip SIP stack and get features provided by this sip stack.

The key features of this software are :

Various codecs support : Speex (narrow band and wideband), G.711 (u-law, a-law), GSM, iLBC, SILK(narrow band, wideband and ultra wideband), G.722

*Can use native audio driver
*STUN and ICE NAT traversal
*Integration with Android operating system with filters and rewriting rules
*A distribution of the software available on the website allow SRTP, SIP over TLS and ZRTP
*SIP-SIMPLE messaging

[[CSipSimple|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Dialer+==

[[File:Dialerplus.png|200px|thumb|left|Dialer+]]

'''Product:''' cd26

'''Company:''' cd26

'''OS:''' IOS

'''Overview:'''
Dialer+ is a SIP-based softphone for iPhone , iPod touch and iPad

Standard Phone Features
* Fast Contact Search – Use phone pad to find contacts by names, initials or any part of the numbers.
• iOS 4.0+ Background Support
• Supports GSM calls when the network is unavailable or slowly
• Multiple Account Support
• Auto switch to best service provider when use multiple account
• Pre-configured VoIP providers list
• Call display
• Speakerphone, Mute and Hold
• Call history
• Contact List and Contact Favorites - leveraging the iPhone Contacts
• Ringtones and contact avatars
• Multiple incoming Call Support - swap between two active calls; merge and split calls
• Audio codecs include G.711,G.722, iLBC ,Speex and GSM, Make an in app purchase to add G.729 for great performance over 3G networks
• Support for RFC 2833 DTMF
• VPN Support
• uPNP Support
• iPhone 4 Retina Display Support

[[Dialer+|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Diax==

[[File:Diax1.png|200px|thumb|left|Diax Softphone]]

'''Product:''' Diax

'''Company:''' Diax

'''OS:''' Windows

'''Overview:'''This is a full featured and very small IAX based software phone for Microsoft Windows platforms (only). The application is distributed as a freeware for personal use.

This is a full featured and very small (just around 300KB for all the files) IAX2 based software phone for Microsoft. It is based on the iaxclient library.

[[Diax_Softphone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Ekiga==

[[File:Ekigaw0.png|200px|thumb|left|Ekiga Softphone]]

'''Product:''' Ekiga

'''Company:''' Ekiga

'''OS:''' Windows, Linux

'''Overview:'''Ekiga is a software phone and video conferencing application. It allows you to hear and see your friends for free using your computer and Internet. You can also chat with them and see if they are online or not. It works on GNU/Linux and Windows.

More specifically, Ekiga is a VoIP, IP Telephony, and Video Conferencing application that allows you to make audio and video calls to remote users with SIP or H.323 compatible hardware and software.

It supports many audio and video codecs and all modern VoIP features for both SIP and H.323. Ekiga is the first Open Source application to support both H.323 and SIP, as well as audio and video. Ekiga was formerly known as GnomeMeeting.

[[Ekiga_Softphone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Express Talk==

[[File:Express0.png|200px|thumb|left|Express Talk]]

'''Product:''' Express Talk

'''Company:''' NCH Software

'''OS:''' Windows, MacOS

'''Overview:''' Express Talk works like a telephone to let you make calls through your computer. Call anyone via the internet who also has a softphone installed and if you sign up with a VoIP gateway service company you call regular telephone numbers as well.

With Express Talk you could potentially call someone using their internet IP (eg. "bob@1.2.3.4") but this is not usually practical. It is much easier to use a friendly address like "bob@myphoneco.com". To have this you need to sign up with a SIP proxy service. This is almost always free to signup for and usually requires nothing more than an email address. See our Recommended SIP Service Providers List for a list of options.

Some of the companies also provide gateway services. This means connecting phone calls to ordinary (non internet or analog PSTN) phone lines. These calls tend to be much less expensive particularly for international calls compared to regular long distance charges since they can offer local call rates.

[[Express_Talk|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==FgVoIP==

[[File:Fgvoip0.jpeg|200px|thumb|left|FgVoIP]]

'''Product:''' FgVoIP

'''Company:''' FG Microtec

'''OS:''' BlackberryOS

'''Overview:''' fg's BlackBerry SIP VoIP Client is a feature rich soft phone application for BlackBerry devices. It can register to any SIP VoIP provider or IP/SIP capable PBX using a WiFi connection in office, campus, home, hotel rooms, public hot-spots etc.

Its much like having a BlackBerry desktop phone in your pocket, providing significantly higher overall productivity while at the same time lowering telephony costs.

Experience the freedom of choosing your own VoIP provider whenever you want, wherever you want. All you need is a WiFi network and a VoIP account.

[[FgVoIP|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Fring==

[[File:Fring00.jpg|left|Fring]]

'''Product:''' Fring

'''Company:''' Fring

'''OS:''' Windows, MacOS, IOS, Linux, Android, Symbian

'''Overview:'''
Fring mobile communication service that gives users internet-rich communication from their mobile phones.

Fring members make FREE video calls, voice calls, live chat to other fringsters and to friends on other social networks.

You can now use your SIP account even from non-SIP enabled phones.

Choose whichever SIP provider you have and in just few clicks, start using this account to make calls to landlines, GSM phones, or to other SIP providers.

Supports Symbian S60, iPhone/ iPod touch, Android, Windows Mobile, MeeGo, Linux, J2MEand proprietary OS.

Fring allows you to easily switch between them, without having to reconfigure your handset through a nightmarish process.

[[Fring|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Grandstream Wave==

[[File:Grandstream Wave.png|200px|Thumb|left|Grandstream Wave]]

'''Product:''' Grandstream Wave

'''Company:''' Grandstream

'''OS:''' Android

'''Overview:'''
Grandstream Wave is a FREE softphone application that allows users to make and receive voice calls through their business or residential SIP accounts on any Android™ device (version 4.0+) from anywhere in the world.

This application supports integration of up to 6 SIP accounts, 6-way voice conferencing, and allows users to monitor their PBX (such as Grandstream's UCM6100 series IP PBX & UCM6510 IP PBX) while utilizing speed dial with up to 24 virtual BLF keys.

Grandstream Wave also supports advanced SIP telephony features including call transfer, LDAP phonebook integration and more.

[[Grandstream Wave|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Idoubs==

[[File:Idoubs2.png|left|Idoubs]]

'''Product:''' Idoubs

'''Company:''' Idoubs

'''OS:''' iOS (iPhone, iPad and iPod Touch) and MAC OS X

'''Overview:'''
iDoubs v2.x is the first fully featured open source 3GPP IMS Client for iOS devices (iPhone, iPod Touch and iPad).

As the SIP follows RFC 3261 and 3GPP TS 24.229 Rel-9 specifications, this will allow you to connect to any compliant SIP registrar. <br>

Crystal clear audio quality (adaptive jitter buffer, echo cancellation, noise suppression, automatic resampling, gain control, ...)<br>

Better video quality (low latency, low cpu usage, ...)<br>

Audio codecs: G.722, G.729AB, AMR-NB, iLBC, GSM, PCMA, PCMU, Speex-NB, Speex-WB, Speex-UWB<br>

Video codecs: . VP8, H264, MP4V-ES, Theora, H.263, H.263-1998<br>

Chat (SMS-like) & Favorites<br>

[[Idoubs Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Jitsi==

[[File:Logo2.png|200px|left|Jitsi]]

'''Product:''' Jitsi

'''Company:''' Jitsi

'''OS:''' Windows, MacOS, Linux

'''Overview:''' Jitsi (formerly SIP Communicator) is an audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, AIM/ICQ, Windows Live, Yahoo! and many other useful features.

Jitsi is an Open Source / Free Software, and is available under the terms of the LGPL.

[[Jitsi|See Configuration Details]]
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==Kiax==

[[File:Kiax3.png|left|200px|Kiax]]

'''Product:''' Kiax

'''Company:''' Kiax

'''OS:''' Windows, MacOS, Linux

'''Overview:'''
Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.

[[Kiax|See Configuration Details]]
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==Linphone==

[[File:Linphone.jpg|left|200px|Linphone]]

'''Product''': Linphone

'''Company''': Linphone

'''OS:''' Windows, Linux, MacOSX, IPhone, Android.

'''App Storage Consumption:''' 8.09mb (ability to move app to SD card. Data uses an additional ~0.89mb approx.)

'''Overview:''' Linphone is an internet phone or Voice Over IP phone (VoIP). With linphone you can communicate freely with people over the internet, with voice, video, and text instant messaging. Linphone makes use of the SIP protocol, an open standard for internet telephony. You can use Linphone with any SIP VoIP operator.

[[Linphone|See Configuration Details]]
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==NetDial Sip Phone==

[[File:NetDial.jpg|left|200px|Linphone]]

'''Product''': NetDial Sip Phone

'''Company''': NeoMecca

'''OS:''' iOS, Android.

[[NetDial_Sip_Phone|See Configuration Details]]
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==Media5-fone==

[[File:Media5Android.jpg|left|Media5Android]]

'''Product:''' Media5-fone

'''Company:''' Media5 Corporation

'''OS:''' Android, iOS

'''Overview:'''
The Media5-fone is a softphone application that runs on the Android and the iOS Operating System. It is a SIP Client (softphone) that enables users to make and receive VoIP calls, enabling them to use their devices as an IP-PBX phone extension in their office or anywhere else in the world.

[[Media5-fone|See Configuration Details]]
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==MizuPhone==

[[File:Mizuphone3.png|left|200px|MizuPhone]]

'''Product:''' MizuPhone

'''Company:''' Mizutech

'''OS:''' Windows

'''Overview:'''
Mizu SoftPhone is our award winner professional VoIP client application based on the open standard SIP protocol with an easy to use modern interface. With Mizu SoftPhone you can connect to any SIP server on the public internet or on your local area network.

Multiple accounts and multiple SIP server registrations combined with a powerful dial plan can minimize your telecom bills while using the greatest features from the VoIP industry.

Features include built-in encryption, IM, presence, HD Video, history with voice records, skype like voice quality using ultra-wideband codecs and much more.

A free edition is also available based on the same engine as our corporate sip softphone but it has some features disabled (like G.729 and wideband codecs, multiple accounts and file transfer over SIP).

[[MizuPhone|See Configuration Details]]
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==NinjaLite==

[[File:Ninja6.png|left|200px|NinjaLite]]

'''Product:''' NinjaLite

'''Company:''' Global IP Telecommunications

'''OS:''' Windows

'''Overview:'''
Ninja Lite offers Audio- and video telephony for everybody, Ninja features individual skins and ringtones!, Ninja is natively Vista 32bit /64 bit and Windows7 compatible.

[[NinjaLite|See Configuration Details]]
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==Nokia E71==

[[File:NokiaE71.jpg|left|150px|PhonerLite]]

'''Product:''' Nokia E71 SIP Client

'''Company:''' Nokia

'''OS:''' Symbian

[[Nokia E71|See Configuration Details]]
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==Nokia N82==

[[File:Nokia_n82_Black.jpg|left|222px|NokiaN82]]

'''Product:''' Nokia N82 SIP Client

'''Company:''' Nokia

'''OS:''' Symbian OS 9.2, S60 3rd Edition, Feature Pack 1

[[Nokia_N82|Configuration Details]]
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==PhonerLite==

[[File:PhonerLite.png|left|300px|PhonerLite]]

'''Product:''' PhonerLite

'''Company:''' PhonerLite

'''OS:''' Windows

'''Overview:'''
PhonerLite is a clearly arranged application for Windows. PhonerLite enables your PC to use it for internet telephony (VoIP , Voice over IP ). Pre-conditions are a full-duplex sound card , a microphone and speakers (alternativelya headset), an internet connection and a registration at a provider supporting the protocol SIP .PhonerLite supports several SIP profiles, each configurable independently. In thesame way the integrated phone book and call log are easy to use.

[[PhonerLite|See Configuration Details]]
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==PortGo==

[[File:Portgo3.png|left|200px|PortGo]]

'''Product:''' PortGo

'''Company:''' PortSIP Solutions, Inc

'''OS:''' Windows (desktop and mobile)

'''Overview:'''
PortGo is the newest SIP softphone from PortSIP, it's built base on PortSIP VoIP SDK, allowing users to enjoy multimedia communications in a dynamic way.

Featuring an intuitive interface, PortGo is expanding the softphone experience by making it even easier to make VoIP and Video over IP calls, see when your contacts are available and send Instant Messages. PortGo features an IM interface which focuses on your contacts and friends.

[[PortGo|See Configuration Details]]
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==QuteCom==

[[File:Qutecom3.png|left|200px|QuteCom]]

'''Product:''' QuteCom

'''Company:''' QuteCom

'''OS:''' Windows,MacOS,Linux

'''Overview:'''
QuteCom is a community of enthusiasts and developers, creating free software products related to communication over IP. The flagship product of the QuteCom project is a softphone which allows you to make free PC to PC video and voice calls, and to integrate all your IM contacts in one place.

[[QuteCom|See Configuration Details]]
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==SIP Droid==

[[File:Sipdroid_00.png|200px|thumb|left|SIP Droid]]

'''Product:''' SIP Droid

'''Company:''' SIP Droid

'''OS:''' Android

'''App Storage Consumption:''' 1.85mb (ability to move app to SD card. Data uses an additional ~4kb approx.)

'''Overview:''' SIPDroid is a java based, open source SIP client that has recently been developed for use with mobile devices based on Google’s Android platform.

[[SIP_Droid|See Configuration Details]]
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==SJPhone==

[[File:Sjphone0.png|200px|thumb|left|SJPhone]]

'''Product:''' SJPhone

'''Company:''' SJ Labs, Inc.

'''OS:''' Windows,Windows Mobile

'''Overview:'''
SJphone is a VOIP softphone that allows you to speak with any other softphone, any stand-alone IP-phone, or using ITSP with any traditional wired or mobile phone.

It supports both SIP and H.323 standards and is fully inter-operable with most major VOIP vendors and ITSP.

[[SJPhone|See Configuration Details]]
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==Taki==

[[File:blackberry_z30_taki_oncall.png|200px|thumb|left|Taki]]

'''Product:''' Taki

'''Company:''' Taki

'''OS:''' BlackBerry 10, BlackBerry PlayBook

'''Project site:''' http://taki.sourceforge.net/

'''App in the AppWorld:''' http://appworld.blackberry.com/webstore/content/130295

'''Overview:''' Taki is a native SIP softphone for BlackBerry® PlayBook™ and
BlackBerry® 10 platforms.

'''Main Features:'''
- Multiple SIP accounts support
- Multiple simultaneous calls
- Call recording
- Conference calls
- Advanced call control: Transfer, Hold, Mute, Reject, Redial, switch
between multiple active calls
- Speaker phone support
- Address book. Integration with local contacts on BlackBerry 10 phones
- Comprehensive SIP settings (port binding, outbound proxy, registrar,
codecs priority, NAT traversal, STUN, Presence, and more)
- Transports: UDP/TCP
- Audio codecs: G.711a/u, GSM, iLBC, G.722 (HD), Speex (NB, WB, UWB)
- Built for BlackBerry

[[Taki|See Configuration Details]]

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==Telephone==

[[File:telephone-user-interface.png|200px|thumb|left|Telephone]]

'''Product:''' Telephone

'''Company:''' 64 Characters

'''OS:''' OS X 10.9 or later

'''Project site:''' http://www.64characters.com/telephone/

'''Mac App Store:''' https://itunes.apple.com/us/app/telephone/id406825478

'''Overview:''' Telephone is a fully-featured SIP client with a minimalist user interface.

<br>

'''Main Features:'''

* Multiple SIP accounts
* Multiple simultaneous calls w/conferencing
* Call control: Transfer, Hold, Mute, Reject, Redial, Call Waiting, multiple active calls
* Speaker phone/headset support (via Mac sound device)
* Integration with Contacts app, dial by name, autocomplete, click to dial
* Send DTMF tones
* Languages: English, German, Russian

[[Telephone|See Configuration Details]]

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==Vimphone==

[[File:Vimphone0.png|200px|thumb|left|Vimphone]]

'''Product:''' Vimphone

'''Company:''' Vimtura

'''OS:''' Android

'''Overview:''' Vimphone is a free app (GPL) which allows Android users to make inexpensive calls using VoIP.ms (or any SIP-compliant provider) and a wi-fi connection.

[[Vimphone|See Configuration Details]]
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==X-Lite==

[[File:Xlite_00.png|200px|thumb|left|X-Lite]]

'''Product:''' X-Lite

'''Company:''' CounterPath

'''OS:''' Windows, MacOS, Linux

'''Overview:''' CounterPath's X-Lite helps you seamlessly transition from a traditional phone environment into the world of Voice over IP.

The latest release of X-Lite provides a completely redesigned interface that allows for a contact-centric or dialpad-centric user experience, or a combination of the two. It also provides you with some of the most popular features of our fully loaded Bria and eyeBeam softphones so you can take them for a test drive before you make your purchase.

Having a simple voice conversation, you’ll soon see why having a softphone on your desktop or laptop is the ultimate communications experience.

[[X-Lite|See Configuration Details]]
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==YakaPhone==

[[File:Yakaphone.png|200px|thumb|left|YakaPhone]]

'''Product:''' YakaPhone

'''Company:''' Bits Valley Ltd

'''OS:''' Windows, MacOS, Linux

'''Overview:''' Yaka Software focuses on providing solutions based on established open source technologies within the Voice-Over-IP sphere. Their flagship products are the customizable IAX-based softphone.It is Open Source and may be modified but any pre or post modification commercial usage requires their prior authorization.

[[YakaPhone|See Configuration Details]]
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==Zoiper Classic==

[[File:Zoiper0.png|300px|thumb|left|Zoiper Classic]]

'''Product:''' Zoiper Classic

'''Company:''' Zoiper

'''OS:''' Windows, Mac OS, Android, iOS

'''Overview:''' IAX & SIP multilanguage softphone is a VoIP soft client, meant to work with any IP-based communications systems and infrastructure.

Zoiper is available for Windows, Mac OS X and Linux and supports the following languages: English, German, Spanish, French, Dutch, Portuguese, Russian, Chinese, Japanese, Italian, Polish, Magyar.

*[[Zoiper_Classic|See Configuration Details for PC/MAC]]<br />
*[[ZoIPer_for_Android|See Configuration Details for ANDROID]]<br/>
*[[ZoIPer_for_iOS|See Configuration Details for iOS]]<br/>
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[[category:softphones]]

==Zoiper Communicator==

[[File:Zoiperco1.png|200px|thumb|left|Zoiper Communicator]]

'''Product:''' Zoiper Communicator

'''Company:''' Zoiper

'''OS:''' Windows, MacOS, Linux, Solaris

'''Overview:''' Zoiper Communicator IAX & SIP softphone is a converged Internet communication tool combining high-quality voice and video calls, fax, instant messaging and presence through a contact-centric intuitive interface.

[[Zoiper_Communicator|See Configuration Details]]
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Devices

2017-04-24T15:18:12Z

Rod:

To be able to place or receive calls using VoIP, you may need a hardware setup that will allow you to use your regular phone (ATA) or a special phone that connects directly to VoIP networks (IP Phone).

* '''ATA:''' An '''Analog Telephone Adapter''' is a device used to connect one or more standard analog telephones to a digital telephone system (such as Voice over IP) or a non-standard telephone system.
* '''IP Phone:''' An IP Phone uses voice over IP (VoIP) technologies allowing telephone calls to be made over an IP network such as the Internet instead of the ordinary PSTN system.

==ATAs==

===Atcom AG188N===

[[File:Atcom-ag188n.jpg|300px|thumb|left|Atcom AG188N]]

'''Product:''' Atcom AG188N

'''Company:''' Atcom

'''Overview:''' AG188N voice gateway is an Internet based one port voice gateway. AG188N ATA adapts multi voice control protocols and voice compression codec to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.

AG188N ATA one FXS voice over IP gateway supports SIP, and IAX2 protocol, offering one 10/100Mbps Ethernet interface, one RJ11 telephone interface, one built-in router and lifeline port. AG-188N ATA is small and can be widely used in SOHO, small office and enterprise branches.

[[Atcom AG188N|See Configuration Details]]
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===Auerswald COMpact 5010===

[[File:Auerswald_5010.jpg|300px|thumb|left|Auerswald COMpact 5010]]

'''Product:''' Auerswald COMpact 5010

'''Company:''' Auerswald

'''Overview:''' Communication center for small companies and ambitious private users with ample expansion possibilities

The COMpact 5010 VoIP makes an ideal communication center for agencies, law firms and small craft businesses as well as the technology-conscious. With the according modules, it gives you 8 trunk lines (analogue, ISDN, VoIP) and connections for 10 internal participants (analogue, ISDN, VoIP). And if you want, our Automatic Update will always keep the operating software of your COMpact system up to speed!

[[Auerswald COMpact 5010|See Configuration Details]]
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===Cisco===

====Cisco Linksys PAP2====

[[File:Pap2.jpg|300px|thumb|left|Cisco Linksys PAP2]]

'''Product:''' Linksys PAP2

'''Company:''' Cisco

'''Overview:''' The Linksys PAP2 is an analog telephony adapter (ATA), which allows for the connection of up to two analog telephones to a softswitch or VoIP carrier using Session Initiation Protocol (SIP). The physical connections on the device are two RJ11 jacks for the telephones and one RJ45 jack for ethernet. The ethernet connection is 10 Mbit/s, half-duplex. The device is configured through a web interface.
Each analog telephone is presented as a distinct SIP user.

[[Cisco Linksys PAP2|See Configuration Details]]
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====Cisco Linksys PAP2T====

[[File:Pap2t.jpg|300px|thumb|left|Cisco Linksys PAP2T]]

'''Product:''' Linksys PAP2T

'''Company:''' Cisco

'''Overview:''' The Cisco PAP2T Internet Phone Adapter with 2 VoIP Ports enables high-quality feature-rich VoIP service through your broadband Internet connection. Just plug it into your home router or gateway and use the two standard telephone ports to connect analog phones or fax machines. Each phone port operates independently, with separate phone service and phone numbers, like having two telephone lines. You'll get clear reception and reliable fax connections, even while using the Internet.

With Internet telephony, along with low domestic and international phone rates, impressive arrays of special telephone features are available. Choose your preferred local dialing number, regardless of where you live. Or add a virtual telephone number to have forwarded to your Internet phone. You can even add a toll-free number. The Cisco PAP2T Internet Phone Adapter is compatible with these and all of the other special telephone features that are available from your Internet telephone service provider, such as caller ID, call waiting, voicemail, call forwarding, distinctive ring, and much more.

[[Cisco Linksys PAP2T|See Configuration Details]]<br>
[[PAP2T With Static IP|See Configuration Details with Static IP]]
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====Cisco SPA112 and SPA122====

[[File:SPA112.jpg|300px|thumb|left|Cisco SPA112]]

'''Product:''' SPA112, SPA122

'''Company:''' Cisco

'''Overview:''' The Cisco SPA112 2 Port Adapter connects to VoIP service through a wired broadband Internet connection and provides two virtual telephone (FxS) lines to which standard touch-tone 'phones may be connected. The SPA122 is very similar to the SPA112 but includes a second network connection, allowing it to be installed as a bridge or router.

Both include two standard telephone ports to easily connect existing analog phones or fax machines. Each phone line can be configured independently.

Introduced in late 2011, this box represents an inexpensive means to continue using existing analogue hardware while migrating to voice over IP.

[[Cisco SPA112|See Configuration Details]]

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====Cisco WRP400 and WRP500====

[[File:WRP400.jpg|300px|thumb|left|Cisco WRP400]]

'''Product:''' WRP400, WRP500

'''Company:''' Cisco

'''Overview:''' The Cisco WRP400/WRP500 wi-fi routers connect to VoIP service through a wired broadband Internet connection and provide two virtual telephone (FxS) lines to which standard touch-tone 'phones may be connected. These devices effectively deliver a complete local area network and a two-line analogue telephone adapter in one box.

The WRP400 provided a wired LAN (four 100baseT downlink ports), plus full-featured wireless B/G networking. Its faster but more expensive successor, the WRP500, provides quicker wi-fi (802.11ac, dual band) and wired LAN (1000baseT) connections. If wired Internet is not available, uplink for these routers can fall back to a mobile Internet connection if appropriate hardware is connected to the USB port.

The telephony capabilities of the WRP400/WRP500 are very similar to the SPA122. Both include two standard telephone ports to easily connect existing analog phones or fax machines. Each phone line can be configured independently. T.38 faxing is supported if a provider provides this capability.

[[Cisco WRP400|See Configuration Details]]

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====Cisco SPA2100 Phone Adapter====

[[File:Sipura2100.jpg|300px|thumb|left|Cisco SPA2100 Phone Adapter]]

'''Product:''' Cisco SPA2100 Phone Adapter

'''Company:''' Cisco

'''Overview:''' Inexpensive, easy to install, and simple to use, the Cisco® SPA2100 Phone Adapter with Router (Figure 1) connects a standard telephone or fax machine to an IP-based data network. Voice over IP (VoIP) service providers can offer residential and business users traditional and enhanced communication services via the customer's broadband connection to the Internet.

The Cisco SPA2100 features two basic telephone ports to connect existing analog phones or fax machines to a private branch exchange (PBX) system. It also includes two RJ-45 Ethernet interfaces to connect to a home or office LAN, as well as an Ethernet connection to connect a broadband modem or router (WAN). Each phone line can be configured independently via software controlled by the service provider or the end user. With the SPA2100, users are able to protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines, as well as control their migration to IP with an extremely affordable, reliable solution.

[[Cisco SPA2100 Phone Adapter|See Configuration Details]]
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====Cisco SPA2102 Phone Adapter with Router====

[[File:Spa2102b.jpg|300px|thumb|left|Cisco SPA2102 Phone Adapter with Router]]

'''Product:''' Cisco SPA2102 Phone Adapter with Router

'''Company:''' Cisco

'''Overview:''' Inexpensive, easy to install, and simple to use, the Cisco SPA2102 Phone Adapter with Router connects a standard telephone or fax machine to an IP-based data network. Voice over IP (VoIP) service providers can offer residential and business users traditional and enhanced communication services via the customer's broadband connection to the Internet.

The Cisco SPA2102 features two basic telephone ports to connect existing analog phones or fax machines to a private branch exchange (PBX) system. It also includes two 100BASE-T RJ-45 Ethernet interfaces to connect to a home or office LAN, as well as an Ethernet connection to connect a broadband modem or router (WAN). Each phone line can be configured independently via software controlled by the service provider or the end user. With the SPA2102, users are able to protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines, as well as control their migration to IP with an extremely affordable, reliable solution.

[[Cisco_SPA2102_Phone_Adapter_with_Router|See Configuration Details]]
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===Grandstream===

====Grandstream HandyTone 486====

[[File:HT486.jpg|300px|thumb|left|Grandstream HandyTone 486]]

'''Product:''' Grandstream HandyTone 486

'''Company:''' Grandstream

'''Overview:''' Grandstream's award-wining HandyTone-486 is an all-in-one VoIP integrated access device that features superb audio quality, rich functionalities, high level of integration, compactness and ultraaffordability. The HandyTone-486 is fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market.

Grandstream HandyTone-486 has been awarded the Best of Show product in 2004 Internet Telephony Conference and Expo.

[[Grandstream_HandyTone_486|See Configuration Details]]
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====Grandstream HandyTone 502 - HT502====

[[File:Ht502.jpg|300px|thumb|left|Grandstream HandyTone 502 - HT502]]

'''Product:''' Grandstream HandyTone 502 - HT502

'''Company:''' Grandstream

'''Overview:''' The HT502 is a powerful VoIP router. The product's inclusion of an integrated high performance NAT router and 10Mbps Ethernet WAN and LAN ports enables a shared broadband connection between multiple ethernet devices.

In addition to being SIP 2.0 standard compliant, the product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles, and advanced telephony features. It provides two lines (FxS), supporting both tone and [[pulse dial]] for compatibility with any standard telephone.

Enhanced security

Automated provisioning using symmetric and asymmetric voice

Support for a broad range of popular voice codec

[[Grandstream_HandyTone_502_-_HT502|See Configuration Details]]
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====Grandstream HandyTone 702 - HT702====

[[File:Ht702.jpg|300px|thumb|left|Grandstream HandyTone 702 - HT702]]

'''Product:''' Grandstream HandyTone 702 - HT702

'''Company:''' Grandstream

'''Overview:''' The HT702 is Grandstream´s Newest powerful VoIP router. The product's inclusion of an integrated high performance NAT router and 10Mbps Ethernet WAN and LAN ports enables a shared broadband connection between multiple ethernet devices.

In addition to being SIP 2.0 standard compliant, the product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles/Phones and advanced telephony features.

Enhanced Security

Automated provisioning using symmetric and asymmetric voice

Support for a broad range of popular voice codec

[[Grandstream_HandyTone_702_-_HT702|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Mediatrix 4100 Series===

[[File:Mediatrix_4100_1_media5.jpg|300px|thumb|left|Mediatrix 4102]]

'''Product:''' : Mediatrix 4100 Series running DGW 2.0 firmware

'''Company:''' Media5 Corporation

'''Overview:''' The Mediatrix® 4100 Series features VoIP adaptors that interconnect up to 24 analog telephones, faxes, and modems ports into SIP-based systems. The Series allows operators to shorten deployment time to enable cloud telephony services into branch offices and SMBs. The Mediatrix 4100 also offers security features such as SIP over TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signaling, and media transmission aspects.

[[Media5 Mediatrix 4100|See Configuration Details]]
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===Netgear WGR615V===

[[File:NetgearWGR615V.jpg|300px|thumb|left|Netgear WGR615V]]

'''Product:''' WGR615V

'''Company:''' Netgear

'''Overview:''' The WGR615V was never sold to directly to the public when it was new but since the model is discontinued, lots of them are being sold.
It is a wireless (802.11 G) router that happens to have a built-in ATA.
It is possible to use just the ATA part.

[[Netgear WGR615V|See Configuration Details]]
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===OBi 100/110 & OBi 200===

[[File:OBi110-ATA.jpg|300px|thumb|left|OBi110]]

'''Product:''' OBi 100/110 & OBi 200

'''Company:''' OBIHAI Technology Inc

'''Overview:''' With OBi devices you are in control of your communications life. From the OBi's on-board connections as well as via the Internet, you have the power to bridge mobile, fixed line and Internet telephone services. The OBi100 & OBi110 are stand-alone, dedicated devices, built with a high-performance "system on a chip" platform to ensure high-quality voice conversations. Every OBi device has high availability and reliability. It is 'always-on' to make or receive a call. With an OBi device, a computer is not required and a computer does not need to be on to talk on the phone or use the OBi's powerful service bridging features. To get started, all you need is a phone, power and a connection to the Internet.

The OBi110 has one phone port and one line port. Use the OBi110 when you need an analog line connection to a traditional telephone network or service. Think of the OBi110 LINE port as a gateway to a traditional phone service. If you do not have a traditional phone service at home, then the OBi100 is probably the right product to get.

[[OBi 100/110 & OBi 200|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Telco AC-211===

[[File:Ac211n.jpg|300px|thumb|left|Telco AC-211]]

'''Product:''' Telco AC-211

'''Company:''' Telco

'''Overview:''' Telco AC-211 is a SIP ATA supporting two lines and numerous configuration options. This ATA is most commonly found as the AC-211-SR & AC-211N from the now defunct SunRocket service. This device works well with voip.ms once configured.

[[Telco_AC-211|See Configuration Details]]

<div style="width:100%;overflow:hidden;clear:both"></div>

===TP-Link TD-VG3631===

[[File:Tplink vg3631 front.png|300px|thumb|left|TPLink VG3631]]

'''Product:''' TP-Link TD-VG3631

'''Company:''' TP-Link

'''Overview:''' High speed DSL modem, NAT router and wireless access point in one device providing a one-stop networking solution Wireless N speed up to 300Mbps makes it ideal for heavy bandwidth consuming or interruption sensitive applications like online gaming, Internet calls and even the HD video streaming Supporting both traditional land lines and VoIP network offers you a multiple choice when making phone calls. Various call features such as caller ID, call waiting, call holding, call forwarding, 3-way conference calls and voicemail USB 2.0 port supports Storage Sharing (Samba or FTP), Media Server, Print Server and multi accounts Ethernet WAN (EWAN) offers another broadband connectivity option for connecting to Cable, VDSL or Fiber modems IP based Bandwidth Control makes it easier for you to manage the bandwidth of devices that connected to the modem router Wi-Fi On/Off Button allows users to simply turn their wireless radio on or off.

[[TP-Link TD-VG3631|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==IP Phones==

===3COM 3108 Wireless Phone===

[[File:3com_3108_1.jpg|300px|thumb|left|3COM 3108 Wireless Phone]]

'''Product:''' 3COM 3108 Wireless Phone

'''Company:''' 3COM

'''Overview:''' The 3Com 3108 Wireless Telephone is a Session Initiation Protocol (SIP)-based wireless Voice over Internet Protocol (VoIP) telephone. SIP is an internationally recognized standard <nowiki>(IETF RFC 3261)</nowiki> for implementing VoIP. You can make and receive VoIP calls as long as your Wireless Telephone is registered with a SIP proxy server and you are operating it within range of an IEEE 802.11b/g enabled wireless network (WLAN). The SIP proxy server can belong to a wireless Internet Telephony Service Provider (ITSP) or corporate VoIP PBX system, such as the 3Com NBX® System.

[[3COM_3108_Wireless_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Aastra 6730i/6731i VoIP Phone===

[[File:Aastra6730i.jpg|300px|frame|left|Aastra 6730i VoIP Phone]]

'''Product:''' Aastra 6730i/6731i VoIP Phone

'''Company:''' Aastra

'''Overview:'''The Aastra 6730i, a new member of the carrier-grade, open-standards based 67xi SIP portfolio, offers exceptional features and flexibility in an enterprise grade IP telephone. With a sleek, elegant design and a compact footprint, this multi-line SIP telephone delivers the advanced features and performance traditionally found only in higher priced products. Featuring a 3 line LCD display, the 6730i supports up to 6 lines with call appearances, offers advanced XML capability to access custom applications and is fully interoperable with leading IP-PBX platforms.

Supported by a host of Aastra configuration options, XML development tools and continuous product enhancements via software releases, the value offered by the 6730i makes it is ideally suited for light telephone use for the small and home-based business applications.

[[Aastra_6730i_VoIP_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Cisco===

====Cisco Linksys SPA942 NA====

[[File:SPA942-0.jpg|300px|thumb|left|Cisco Linksys SPA942 NA]]

'''Product:''' Linksys SPA942 NA

'''Company:''' Cisco

'''Overview:''' Four active lines and dual switched Ethernet ports offer incredible flexibility to the home or small-business Internet telephony user. Part of Cisco Small Business IP Phone Series, the SPA942 IP Phone can be configured to carry a unique phone number or extension for each of its lines, or can be configured to use a shared number over multiple phones.

Based on the SIP standard, the SPA942 is able to easily integrate new features and services offered by providers. A high-resolution display makes for easy menu- and web-based configuration.

[[Cisco_Linksys_SPA942_NA|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco SPA525G====

[[File:525g.jpg|300px|thumb|left|Cisco SPA525g Phone Adapter with Router]]

'''Product:''' Cisco SPA525G

'''Company:''' Cisco

'''Overview:''' The Cisco SPA525G 5-line IP Phone with Color Display is a full-featured VoIP (Voice over Internet Protocol) phone that provide voice communication over an IP network. It provides traditional features, such as call forwarding, redialing, speed dialing, transferring calls, conference calling, and accessing voice mail. Calls can be made or received with a handset, headset or speaker.
Your Cisco IP Phone provides a web interface for the phone user that allows you to configure some features of your phone by using a web browser.
This article will guide you through the steps for basic configuration to make it work with VoIP.ms

[[Cisco SPA525G|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco IP Phone 7940/7960====

[[File:Cisco_7960g.png|300px|thumb|left|Cisco IP Phone 7940/7960]]

'''Product:''' Cisco IP Phone 7940/7960

'''Company:''' Cisco

'''Overview:''' The Cisco IP Phone 7940/7960 is a full-feature telephone that provides voice communication over an IP network. This phone functions like a traditional analog phone, allowing you to place and receive telephone calls. This phone also supports features, such as network call forwarding, redialing, speed dialing, transferring calls, placing conference calls, and accessing voice mail. Phones require Power Over Ethernet (PoE) or [http://www.ciscopowercube.com Cisco CP-PWR-CUBE] 48V AC Adapter to power up.

[[Cisco_IP_Phone_7940/7960|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco SPA30x and SPA50x series IP phones====

[[File:Cisco_spa504g.jpg|300px|thumb|left|Cisco SPA504G Phone]]

'''Product:''' Cisco SPA504G Phone

'''Company:''' Cisco

'''Overview:'''
The Cisco SPA504G is an office-style desk telephone with built-in voice over Internet.

It is one in a series of similar models (SPA30x and SPA50x) which vary primarily in the number of lines (extensions) on the 'phone, power source (some models use power-over-Ethernet) and the availability of a second Ethernet connector. These devices are well-suited to office and IP PBX applications. These do not provide a virtual line for connecting analogue devices such as standard telephone handsets; they are instead self-contained to connect directly to VoIP.

[[Cisco_SPA504G_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Gigaset A510 IP===

[[File:Gigaset_a510_IP.jpg#file|300px|thumb|left]]

'''Product:'''Gigaset A510 IP

'''Company:'''Gigaset

'''Overview:'''The Gigaset A510 and C610 IP phones are fitting solutions if you are looking for the flexibility of VoIP and the convenience of using a cordless handset.

[[Gigaset_A510_IP| See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Grandstream===

====Grandstream DP715/DP710====

[[File:Grandstream 715-710.png|300px|thumb|left]]

'''Product:''' Grandstream DP715/DP710

'''Company:''' Grandstream

'''Overview:''' The DP715/710 is the next generation of powerful, affordable, high quality and simple to configure DECT Cordless IPPhone for
small business and residential users. Their compact size, superb voice quality, rich feature set, market
leading price-performance and wide range radio coverage enable consumers to maximize the power of IP
voice application and mobility for a minimum investment.

[[Grandstream_DP715/DP710| See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Grandstream GXP2200====

[[File:Grandstream_GXP2200.png|300px|thumb|left|Grandstream GXP2200]]

'''Product:''' Grandstream GXP2200

'''Company:''' Grandstream

'''Overview:''' The GXP2200 is one of the most advanced AndroidTM desktop IP phones available on the market today. The innovative phone includes the AndroidTM version 2.3 operating system with a 4.3 inch capacitive touchscreen LCD and the ability to host 6 SIP accounts. Web applications such as news, social media sites, and games can be downloaded directly via Google Play Store, and applications can be created to fit any need and downloaded directly to the phone for customized use.

[[Grandstream_GXP2200|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Grandstream GXP2120 IP Phone====

[[File:Gxp2110.png|300px|thumb|left|Grandstream GXP2120 IP Phone]]

'''Product:''' Grandstream GXP2120 IP Phone

'''Company:''' Grandstream

'''Overview:''' The Grandstream GXP2120 is a 6 line SIP Phone which features HD Voice hardware and software support and a large 320 x 160 backlit graphical LCD. The GXP2120 can handle 6 SIP accounts represented by 6 dual-color line keys and 4 XML programmable context-sensitive soft keys. In addition, the GXP2120 has 7 dual-color BLF extension keys for the most common calls and transfers making it an ideal phone for an office user with moderate to heavy interoffice calling needs.

[[Grandstream_GXP2120_IP_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Panasonic KX-TGP 550===

[[File:Pana550b.jpg|300px|thumb|left|Panasonic KX-TGP 550]]

'''Product:''' Panasonic KX-TGP 550

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-TGP550 responses to the needs of SIP IP-Centrix/Hosted PBX systems and Asterisk users. Conveniently, no need to set up a system telephone at every base. This system also enables you to use a range of convenient services provided by the carrier such as Call Forward, Voice Mail etc.

*Support for 3 simultaneous network conversations
*Up to 6 DECT cordless handsets
*Support for up to 8 SIP registrations (e.g. up to 8 DID lines or extensions)
*Hands-free speaker phone on cordless handset

[[Panasonic KX-TGP 550|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Pirelli DP-L10===

[[File:Pirelli1.jpg|300px|thumb|left|Pirelli DP-L10]]

'''Product:''' Pirelli DP-L10

'''Company:''' Pirelli

'''Overview:''' Pirelli's DualPhone DP-L10 phone combines GSM tri-band capabilities with SIP-based Wireless VoIP via an integrated WLAN 802.11b/g interface, combining cell phone technology with the added advantage of transporting phone calls over the Internet.

The Dual Phone enables mobile phone features such as Internet browsing, e-mail, SMS, and MMS, which can also be used through a fixed-broadband connection, providing the same mobile services with the added benefits of greater bandwidth and cost savings.

[[Pirelli DP-L10|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Polycom===

====Polycom SoundStation IP 4000 Conference Phone====

[[File:Polycom-4000-0.gif|300px|thumb|left|Polycom SoundStation IP 4000 Conference Phone]]

'''Product:''' Polycom SoundStation IP 4000 Conference Phone

'''Company:''' Polycom

'''Overview:''' the SoundStation IP 4000 SIP voice conferencing unit is the answer for organizations that are ready for the
benefits and versatility of a SIP enabled business. Designed for offices or small to medium sized conference rooms, the SoundStation IP 4000 SIP provides remarkable room coverage. You can speak naturally from up to 10-feet away from a microphone and still be heard clearly on the far end of the call. Need coverage for a larger room? The optional extension microphones offer an increased pickup for larger rooms. Plus, with gated microphone technology, echo and background noise is almost entirely eliminated.

The SoundStation IP 4000 SIP also provides familiar features that are easy to use. The menu driven user interface, viewable on a high-resolution backlit LCD, offers convenient access to business telephony functions you use most including transfer, hold, redial, and conference.

[[Polycom_SoundStation_IP_4000_Conference_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Polycom SoundPoint IP 501, 550, 650, etc.====

[[File:Polycom501.jpg|258px|thumb|left|Polycom SoundPoint IP 501]]

'''Product:''' Polycom SoundPoint IP 501, 550, 650, etc.

'''Company:''' Polycom

'''Overview:''' The SoundPoint IP series are multi-line (3-6 lines depending on model) Voice over IP telephones that seamlessly integrate with IP PBX and softswitch vendors’ IP solutions.

As protocols develop and standards evolve, it’s easy to update the phone in the field via a software download, thus enabling new features and functionality for the phone and protecting your investment.

Whether you deploy MGCP or SIP standards, Polycom offers a solution that fits all your business communication needs.

[[Polycom_SoundPoint_IP_501|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Polycom SoundPoint IP 601====

[[File:Voipms-polycom601.jpg|258px|thumb|left|Polycom SoundPoint IP 601]]

'''Product:''' Polycom SoundPoint IP 601

'''Company:''' Polycom

'''Overview:''' The 6-line Polycom® SoundPoint IP™ 601 offers industry-leading functionality and call handling unmatched voice quality an intuitive user interface & expandability to 12 lines!

[[Polycom_SoundPoint_IP_601|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Siemens Gigaset C450-Ip===

[[File:Siemens_gigaset_c450_IP.jpg|300px|thumb|left|Siemens Gigaset C450-Ip]]

'''Product:''' Siemens Gigaset C450-Ip

'''Company:''' Siemens

'''Overview:''' The Gigaset C450 is an entry phone in the Siemens Gigaset product line, the addition of IP connectivity via a standard Ethernet 10/100 BaseT interface and SIP protocol within a reasonable price bracket, opens large applications where wireless operation and SIP are required.
The Ethernet connectivity allows the unit to work without a PC.

The dual mode (SIP/PSTN) enables incoming call on legacy interface with existing directory number and safe emergency outgoing calls.

The presentation of the unit with an independent base and a dedicated phone charger/support will simplify cabling.

[[Siemens Gigaset C450-Ip|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Snom===

====Snom 320 VoIP Phone====

[[File:Snom320.png|300px|frame|left|Snom 320 VoIP Phone]]

'''Product:''' Snom 320 VoIP Phone

'''Company:''' Snom

'''Overview:'''Ideal for the office and everyone who spends a lot of time on the phone, the snom 320 is an affordable, yet powerful SIP business phone with built-in, full-duplex speakerphone and three-party conference bridging.

[[SNOM 320|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Snom m3 VoIP Phone====

[[File:Snom_m3.png|300px|thumb|left|Snom m3 VoIP Phone]]

'''Product:''' Snom m3 VoIP Phone

'''Company:''' Snom

'''Overview:''' The snom m9 is the next generation DECT handheld (CAT-iq) that empowers users with the convenience of wireless communication along with the widely accepted benefits and feature richness of Voice-over-IP telephony.

The snom m9 promises to deliver excellent speech quality through digital and wideband audio (klarVOICE).

By combining professional functions of versatile business communication with the intuitive features of the mobile-carrier world, the snom m9 is ideally suited for professional and private use alike. With features such as hands free mode, calling line identification (CLI) by displaying name, number and image of the caller as well as typical mobile-phone features such as Address Book, Calendar, Calculator and Alarm function, the snom m9 provides the perfect blend of mobility and accessibility.

[[Snom_m3_VoIP_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Yealink SIP-T28P (VSRF)===

[[File:Yealink-sip-t28p.jpg|300px|thumb|left|Yealink SIP-T28P]]

'''Product:''' Yealink SIP-T28P

'''Company:''' Yealink

'''Overview:''' Yealink SIP-T28P represents the next generation VoIP phone which designed for the business user who needs rich telephony features, friendly UI and super voice quality. It is equipped with the TI TITAN chipset, offers high definition voice quality through TI voice engine, HD handset, HD speaker and HD codec (G.722).

By the large, high-resolution graphical display, and together with all the 48 keys, SIP-T28P offers an excellent user experience to configure, make calls, express XML browser, etc. Moreover, to avoid problems with unwanted violations of your audio data, Yealink SIP-T28P supports the security standards TLS, SRTP, HTTPS, 802.1x, Open VPN and AES encryption which are necessary to protect against electronic eavesdropping and data theft.

[[Yealink_SIP-T28P|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>
[[Category:Devices]]

===Zycoo ZP502===

[[File:ZP502.jpg|left|Zycoo ZP502]]

'''Product:''' Zycoo ZP502

'''Company:''' Zycoo ZP502

'''Overview:''' ZP502 VoIP Phone is ZYCOO business IP phone terminal which adopts SIP and IAX2 protocols.

With Broadcom solution,compatible with various Platforms such as Asterisk, FreePBX, Broadsoft, Cisco call manager etc.

[[Zycoo ZP502|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Devices

2017-04-19T17:22:16Z

Rod:

To be able to place or receive calls using VoIP, you may need a hardware setup that will allow you to use your regular phone (ATA) or a special phone that connects directly to VoIP networks (IP Phone).

* '''ATA:''' An '''Analog Telephone Adapter''' is a device used to connect one or more standard analog telephones to a digital telephone system (such as Voice over IP) or a non-standard telephone system.
* '''IP Phone:''' An IP Phone uses voice over IP (VoIP) technologies allowing telephone calls to be made over an IP network such as the Internet instead of the ordinary PSTN system.

==3COM 3108 Wireless Phone==

[[File:3com_3108_1.jpg|300px|thumb|left|3COM 3108 Wireless Phone]]

'''Product:''' 3COM 3108 Wireless Phone

'''Company:''' 3COM

'''Overview:''' The 3Com 3108 Wireless Telephone is a Session Initiation Protocol (SIP)-based wireless Voice over Internet Protocol (VoIP) telephone. SIP is an internationally recognized standard <nowiki>(IETF RFC 3261)</nowiki> for implementing VoIP. You can make and receive VoIP calls as long as your Wireless Telephone is registered with a SIP proxy server and you are operating it within range of an IEEE 802.11b/g enabled wireless network (WLAN). The SIP proxy server can belong to a wireless Internet Telephony Service Provider (ITSP) or corporate VoIP PBX system, such as the 3Com NBX® System.

[[3COM_3108_Wireless_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Aastra 6730i/6731i VoIP Phone==

[[File:Aastra6730i.jpg|300px|frame|left|Aastra 6730i VoIP Phone]]

'''Product:''' Aastra 6730i/6731i VoIP Phone

'''Company:''' Aastra

'''Overview:'''The Aastra 6730i, a new member of the carrier-grade, open-standards based 67xi SIP portfolio, offers exceptional features and flexibility in an enterprise grade IP telephone. With a sleek, elegant design and a compact footprint, this multi-line SIP telephone delivers the advanced features and performance traditionally found only in higher priced products. Featuring a 3 line LCD display, the 6730i supports up to 6 lines with call appearances, offers advanced XML capability to access custom applications and is fully interoperable with leading IP-PBX platforms.

Supported by a host of Aastra configuration options, XML development tools and continuous product enhancements via software releases, the value offered by the 6730i makes it is ideally suited for light telephone use for the small and home-based business applications.

[[Aastra_6730i_VoIP_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Atcom AG188N==

[[File:Atcom-ag188n.jpg|300px|thumb|left|Atcom AG188N]]

'''Product:''' Atcom AG188N

'''Company:''' Atcom

'''Overview:''' AG188N voice gateway is an Internet based one port voice gateway. AG188N ATA adapts multi voice control protocols and voice compression codec to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.

AG188N ATA one FXS voice over IP gateway supports SIP, and IAX2 protocol, offering one 10/100Mbps Ethernet interface, one RJ11 telephone interface, one built-in router and lifeline port. AG-188N ATA is small and can be widely used in SOHO, small office and enterprise branches.

[[Atcom AG188N|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Auerswald COMpact 5010==

[[File:Auerswald_5010.jpg|300px|thumb|left|Auerswald COMpact 5010]]

'''Product:''' Auerswald COMpact 5010

'''Company:''' Auerswald

'''Overview:''' Communication center for small companies and ambitious private users with ample expansion possibilities

The COMpact 5010 VoIP makes an ideal communication center for agencies, law firms and small craft businesses as well as the technology-conscious. With the according modules, it gives you 8 trunk lines (analogue, ISDN, VoIP) and connections for 10 internal participants (analogue, ISDN, VoIP). And if you want, our Automatic Update will always keep the operating software of your COMpact system up to speed!

[[Auerswald COMpact 5010|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco Linksys PAP2==

[[File:Pap2.jpg|300px|thumb|left|Cisco Linksys PAP2]]

'''Product:''' Linksys PAP2

'''Company:''' Cisco

'''Overview:''' The Linksys PAP2 is an analog telephony adapter (ATA), which allows for the connection of up to two analog telephones to a softswitch or VoIP carrier using Session Initiation Protocol (SIP). The physical connections on the device are two RJ11 jacks for the telephones and one RJ45 jack for ethernet. The ethernet connection is 10 Mbit/s, half-duplex. The device is configured through a web interface.
Each analog telephone is presented as a distinct SIP user.

[[Cisco Linksys PAP2|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco Linksys PAP2T==

[[File:Pap2t.jpg|300px|thumb|left|Cisco Linksys PAP2T]]

'''Product:''' Linksys PAP2T

'''Company:''' Cisco

'''Overview:''' The Cisco PAP2T Internet Phone Adapter with 2 VoIP Ports enables high-quality feature-rich VoIP service through your broadband Internet connection. Just plug it into your home router or gateway and use the two standard telephone ports to connect analog phones or fax machines. Each phone port operates independently, with separate phone service and phone numbers, like having two telephone lines. You'll get clear reception and reliable fax connections, even while using the Internet.

With Internet telephony, along with low domestic and international phone rates, impressive arrays of special telephone features are available. Choose your preferred local dialing number, regardless of where you live. Or add a virtual telephone number to have forwarded to your Internet phone. You can even add a toll-free number. The Cisco PAP2T Internet Phone Adapter is compatible with these and all of the other special telephone features that are available from your Internet telephone service provider, such as caller ID, call waiting, voicemail, call forwarding, distinctive ring, and much more.

[[Cisco Linksys PAP2T|See Configuration Details]]<br>
[[PAP2T With Static IP|See Configuration Details with Static IP]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco SPA112 and SPA122==

[[File:SPA112.jpg|300px|thumb|left|Cisco SPA112]]

'''Product:''' SPA112, SPA122

'''Company:''' Cisco

'''Overview:''' The Cisco SPA112 2 Port Adapter connects to VoIP service through a wired broadband Internet connection and provides two virtual telephone (FxS) lines to which standard touch-tone 'phones may be connected. The SPA122 is very similar to the SPA112 but includes a second network connection, allowing it to be installed as a bridge or router.

Both include two standard telephone ports to easily connect existing analog phones or fax machines. Each phone line can be configured independently.

Introduced in late 2011, this box represents an inexpensive means to continue using existing analogue hardware while migrating to voice over IP.

[[Cisco SPA112|See Configuration Details]]

<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco WRP400 and WRP500==

[[File:WRP400.jpg|300px|thumb|left|Cisco WRP400]]

'''Product:''' WRP400, WRP500

'''Company:''' Cisco

'''Overview:''' The Cisco WRP400/WRP500 wi-fi routers connect to VoIP service through a wired broadband Internet connection and provide two virtual telephone (FxS) lines to which standard touch-tone 'phones may be connected. These devices effectively deliver a complete local area network and a two-line analogue telephone adapter in one box.

The WRP400 provided a wired LAN (four 100baseT downlink ports), plus full-featured wireless B/G networking. Its faster but more expensive successor, the WRP500, provides quicker wi-fi (802.11ac, dual band) and wired LAN (1000baseT) connections. If wired Internet is not available, uplink for these routers can fall back to a mobile Internet connection if appropriate hardware is connected to the USB port.

The telephony capabilities of the WRP400/WRP500 are very similar to the SPA122. Both include two standard telephone ports to easily connect existing analog phones or fax machines. Each phone line can be configured independently. T.38 faxing is supported if a provider provides this capability.

[[Cisco WRP400|See Configuration Details]]

<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco Linksys SPA942 NA==

[[File:SPA942-0.jpg|300px|thumb|left|Cisco Linksys SPA942 NA]]

'''Product:''' Linksys SPA942 NA

'''Company:''' Cisco

'''Overview:''' Four active lines and dual switched Ethernet ports offer incredible flexibility to the home or small-business Internet telephony user. Part of Cisco Small Business IP Phone Series, the SPA942 IP Phone can be configured to carry a unique phone number or extension for each of its lines, or can be configured to use a shared number over multiple phones.

Based on the SIP standard, the SPA942 is able to easily integrate new features and services offered by providers. A high-resolution display makes for easy menu- and web-based configuration.

[[Cisco_Linksys_SPA942_NA|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

== Cisco SPA525G ==

[[File:525g.jpg|300px|thumb|left|Cisco SPA525g Phone Adapter with Router]]

'''Product:''' Cisco SPA525G

'''Company:''' Cisco

'''Overview:''' The Cisco SPA525G 5-line IP Phone with Color Display is a full-featured VoIP (Voice over Internet Protocol) phone that provide voice communication over an IP network. It provides traditional features, such as call forwarding, redialing, speed dialing, transferring calls, conference calling, and accessing voice mail. Calls can be made or received with a handset, headset or speaker.
Your Cisco IP Phone provides a web interface for the phone user that allows you to configure some features of your phone by using a web browser.
This article will guide you through the steps for basic configuration to make it work with VoIP.ms

[[Cisco SPA525G|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco IP Phone 7940/7960==

[[File:Cisco_7960g.png|300px|thumb|left|Cisco IP Phone 7940/7960]]

'''Product:''' Cisco IP Phone 7940/7960

'''Company:''' Cisco

'''Overview:''' The Cisco IP Phone 7940/7960 is a full-feature telephone that provides voice communication over an IP network. This phone functions like a traditional analog phone, allowing you to place and receive telephone calls. This phone also supports features, such as network call forwarding, redialing, speed dialing, transferring calls, placing conference calls, and accessing voice mail. Phones require Power Over Ethernet (PoE) or [http://www.ciscopowercube.com Cisco CP-PWR-CUBE] 48V AC Adapter to power up.

[[Cisco_IP_Phone_7940/7960|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco SPA2100 Phone Adapter==

[[File:Sipura2100.jpg|300px|thumb|left|Cisco SPA2100 Phone Adapter]]

'''Product:''' Cisco SPA2100 Phone Adapter

'''Company:''' Cisco

'''Overview:''' Inexpensive, easy to install, and simple to use, the Cisco® SPA2100 Phone Adapter with Router (Figure 1) connects a standard telephone or fax machine to an IP-based data network. Voice over IP (VoIP) service providers can offer residential and business users traditional and enhanced communication services via the customer's broadband connection to the Internet.

The Cisco SPA2100 features two basic telephone ports to connect existing analog phones or fax machines to a private branch exchange (PBX) system. It also includes two RJ-45 Ethernet interfaces to connect to a home or office LAN, as well as an Ethernet connection to connect a broadband modem or router (WAN). Each phone line can be configured independently via software controlled by the service provider or the end user. With the SPA2100, users are able to protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines, as well as control their migration to IP with an extremely affordable, reliable solution.

[[Cisco SPA2100 Phone Adapter|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco SPA2102 Phone Adapter with Router==

[[File:Spa2102b.jpg|300px|thumb|left|Cisco SPA2102 Phone Adapter with Router]]

'''Product:''' Cisco SPA2102 Phone Adapter with Router

'''Company:''' Cisco

'''Overview:''' Inexpensive, easy to install, and simple to use, the Cisco SPA2102 Phone Adapter with Router connects a standard telephone or fax machine to an IP-based data network. Voice over IP (VoIP) service providers can offer residential and business users traditional and enhanced communication services via the customer's broadband connection to the Internet.

The Cisco SPA2102 features two basic telephone ports to connect existing analog phones or fax machines to a private branch exchange (PBX) system. It also includes two 100BASE-T RJ-45 Ethernet interfaces to connect to a home or office LAN, as well as an Ethernet connection to connect a broadband modem or router (WAN). Each phone line can be configured independently via software controlled by the service provider or the end user. With the SPA2102, users are able to protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines, as well as control their migration to IP with an extremely affordable, reliable solution.

[[Cisco_SPA2102_Phone_Adapter_with_Router|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco SPA30x and SPA50x series IP phones==

[[File:Cisco_spa504g.jpg|300px|thumb|left|Cisco SPA504G Phone]]

'''Product:''' Cisco SPA504G Phone

'''Company:''' Cisco

'''Overview:'''
The Cisco SPA504G is an office-style desk telephone with built-in voice over Internet.

It is one in a series of similar models (SPA30x and SPA50x) which vary primarily in the number of lines (extensions) on the 'phone, power source (some models use power-over-Ethernet) and the availability of a second Ethernet connector. These devices are well-suited to office and IP PBX applications. These do not provide a virtual line for connecting analogue devices such as standard telephone handsets; they are instead self-contained to connect directly to VoIP.

[[Cisco_SPA504G_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Gigaset A510 IP==

[[File:Gigaset_a510_IP.jpg#file|300px|thumb|left]]

'''Product:'''Gigaset A510 IP

'''Company:'''Gigaset

'''Overview:'''The Gigaset A510 and C610 IP phones are fitting solutions if you are looking for the flexibility of VoIP and the convenience of using a cordless handset.

[[Gigaset_A510_IP| See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Grandstream DP715/DP710==

[[File:Grandstream 715-710.png|300px|thumb|left]]

'''Product:''' Grandstream DP715/DP710

'''Company:''' Grandstream

'''Overview:''' The DP715/710 is the next generation of powerful, affordable, high quality and simple to configure DECT Cordless IPPhone for
small business and residential users. Their compact size, superb voice quality, rich feature set, market
leading price-performance and wide range radio coverage enable consumers to maximize the power of IP
voice application and mobility for a minimum investment.

[[Grandstream_DP715/DP710| See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Grandstream GXP2200==

[[File:Grandstream_GXP2200.png|300px|thumb|left|Grandstream GXP2200]]

'''Product:''' Grandstream GXP2200

'''Company:''' Grandstream

'''Overview:''' The GXP2200 is one of the most advanced AndroidTM desktop IP phones available on the market today. The innovative phone includes the AndroidTM version 2.3 operating system with a 4.3 inch capacitive touchscreen LCD and the ability to host 6 SIP accounts. Web applications such as news, social media sites, and games can be downloaded directly via Google Play Store, and applications can be created to fit any need and downloaded directly to the phone for customized use.

[[Grandstream_GXP2200|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Grandstream GXP2120 IP Phone==

[[File:Gxp2110.png|300px|thumb|left|Grandstream GXP2120 IP Phone]]

'''Product:''' Grandstream GXP2120 IP Phone

'''Company:''' Grandstream

'''Overview:''' The Grandstream GXP2120 is a 6 line SIP Phone which features HD Voice hardware and software support and a large 320 x 160 backlit graphical LCD. The GXP2120 can handle 6 SIP accounts represented by 6 dual-color line keys and 4 XML programmable context-sensitive soft keys. In addition, the GXP2120 has 7 dual-color BLF extension keys for the most common calls and transfers making it an ideal phone for an office user with moderate to heavy interoffice calling needs.

[[Grandstream_GXP2120_IP_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Grandstream HandyTone 486==

[[File:HT486.jpg|300px|thumb|left|Grandstream HandyTone 486]]

'''Product:''' Grandstream HandyTone 486

'''Company:''' Grandstream

'''Overview:''' Grandstream's award-wining HandyTone-486 is an all-in-one VoIP integrated access device that features superb audio quality, rich functionalities, high level of integration, compactness and ultraaffordability. The HandyTone-486 is fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market.

Grandstream HandyTone-486 has been awarded the Best of Show product in 2004 Internet Telephony Conference and Expo.

[[Grandstream_HandyTone_486|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Grandstream HandyTone 502 - HT502==

[[File:Ht502.jpg|300px|thumb|left|Grandstream HandyTone 502 - HT502]]

'''Product:''' Grandstream HandyTone 502 - HT502

'''Company:''' Grandstream

'''Overview:''' The HT502 is a powerful VoIP router. The product's inclusion of an integrated high performance NAT router and 10Mbps Ethernet WAN and LAN ports enables a shared broadband connection between multiple ethernet devices.

In addition to being SIP 2.0 standard compliant, the product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles, and advanced telephony features. It provides two lines (FxS), supporting both tone and [[pulse dial]] for compatibility with any standard telephone.

Enhanced security

Automated provisioning using symmetric and asymmetric voice

Support for a broad range of popular voice codec

[[Grandstream_HandyTone_502_-_HT502|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Grandstream HandyTone 702 - HT702==

[[File:Ht702.jpg|300px|thumb|left|Grandstream HandyTone 702 - HT702]]

'''Product:''' Grandstream HandyTone 702 - HT702

'''Company:''' Grandstream

'''Overview:''' The HT702 is Grandstream´s Newest powerful VoIP router. The product's inclusion of an integrated high performance NAT router and 10Mbps Ethernet WAN and LAN ports enables a shared broadband connection between multiple ethernet devices.

In addition to being SIP 2.0 standard compliant, the product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles/Phones and advanced telephony features.

Enhanced Security

Automated provisioning using symmetric and asymmetric voice

Support for a broad range of popular voice codec

[[Grandstream_HandyTone_702_-_HT702|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Mediatrix 4100 Series==

[[File:Mediatrix_4100_1_media5.jpg|300px|thumb|left|Mediatrix 4102]]

'''Product:''' : Mediatrix 4100 Series running DGW 2.0 firmware

'''Company:''' Media5 Corporation

'''Overview:''' The Mediatrix® 4100 Series features VoIP adaptors that interconnect up to 24 analog telephones, faxes, and modems ports into SIP-based systems. The Series allows operators to shorten deployment time to enable cloud telephony services into branch offices and SMBs. The Mediatrix 4100 also offers security features such as SIP over TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signaling, and media transmission aspects.

[[Media5 Mediatrix 4100|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Netgear WGR615V==

[[File:NetgearWGR615V.jpg|300px|thumb|left|Netgear WGR615V]]

'''Product:''' WGR615V

'''Company:''' Netgear

'''Overview:''' The WGR615V was never sold to directly to the public when it was new but since the model is discontinued, lots of them are being sold.
It is a wireless (802.11 G) router that happens to have a built-in ATA.
It is possible to use just the ATA part.

[[Netgear WGR615V|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==OBi 100/110 & OBi 200==

[[File:OBi110-ATA.jpg|300px|thumb|left|OBi110]]

'''Product:''' OBi 100/110 & OBi 200

'''Company:''' OBIHAI Technology Inc

'''Overview:''' With OBi devices you are in control of your communications life. From the OBi's on-board connections as well as via the Internet, you have the power to bridge mobile, fixed line and Internet telephone services. The OBi100 & OBi110 are stand-alone, dedicated devices, built with a high-performance "system on a chip" platform to ensure high-quality voice conversations. Every OBi device has high availability and reliability. It is 'always-on' to make or receive a call. With an OBi device, a computer is not required and a computer does not need to be on to talk on the phone or use the OBi's powerful service bridging features. To get started, all you need is a phone, power and a connection to the Internet.

The OBi110 has one phone port and one line port. Use the OBi110 when you need an analog line connection to a traditional telephone network or service. Think of the OBi110 LINE port as a gateway to a traditional phone service. If you do not have a traditional phone service at home, then the OBi100 is probably the right product to get.

[[OBi 100/110 & OBi 200|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Panasonic KX-TGP 550==

[[File:Pana550b.jpg|300px|thumb|left|Panasonic KX-TGP 550]]

'''Product:''' Panasonic KX-TGP 550

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-TGP550 responses to the needs of SIP IP-Centrix/Hosted PBX systems and Asterisk users. Conveniently, no need to set up a system telephone at every base. This system also enables you to use a range of convenient services provided by the carrier such as Call Forward, Voice Mail etc.

*Support for 3 simultaneous network conversations
*Up to 6 DECT cordless handsets
*Support for up to 8 SIP registrations (e.g. up to 8 DID lines or extensions)
*Hands-free speaker phone on cordless handset

[[Panasonic KX-TGP 550|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Pirelli DP-L10==

[[File:Pirelli1.jpg|300px|thumb|left|Pirelli DP-L10]]

'''Product:''' Pirelli DP-L10

'''Company:''' Pirelli

'''Overview:''' Pirelli's DualPhone DP-L10 phone combines GSM tri-band capabilities with SIP-based Wireless VoIP via an integrated WLAN 802.11b/g interface, combining cell phone technology with the added advantage of transporting phone calls over the Internet.

The Dual Phone enables mobile phone features such as Internet browsing, e-mail, SMS, and MMS, which can also be used through a fixed-broadband connection, providing the same mobile services with the added benefits of greater bandwidth and cost savings.

[[Pirelli DP-L10|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Polycom SoundStation IP 4000 Conference Phone==

[[File:Polycom-4000-0.gif|300px|thumb|left|Polycom SoundStation IP 4000 Conference Phone]]

'''Product:''' Polycom SoundStation IP 4000 Conference Phone

'''Company:''' Polycom

'''Overview:''' the SoundStation IP 4000 SIP voice conferencing unit is the answer for organizations that are ready for the
benefits and versatility of a SIP enabled business. Designed for offices or small to medium sized conference rooms, the SoundStation IP 4000 SIP provides remarkable room coverage. You can speak naturally from up to 10-feet away from a microphone and still be heard clearly on the far end of the call. Need coverage for a larger room? The optional extension microphones offer an increased pickup for larger rooms. Plus, with gated microphone technology, echo and background noise is almost entirely eliminated.

The SoundStation IP 4000 SIP also provides familiar features that are easy to use. The menu driven user interface, viewable on a high-resolution backlit LCD, offers convenient access to business telephony functions you use most including transfer, hold, redial, and conference.

[[Polycom_SoundStation_IP_4000_Conference_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Polycom SoundPoint IP 501, 550, 650, etc.==

[[File:Polycom501.jpg|258px|thumb|left|Polycom SoundPoint IP 501]]

'''Product:''' Polycom SoundPoint IP 501, 550, 650, etc.

'''Company:''' Polycom

'''Overview:''' The SoundPoint IP series are multi-line (3-6 lines depending on model) Voice over IP telephones that seamlessly integrate with IP PBX and softswitch vendors’ IP solutions.

As protocols develop and standards evolve, it’s easy to update the phone in the field via a software download, thus enabling new features and functionality for the phone and protecting your investment.

Whether you deploy MGCP or SIP standards, Polycom offers a solution that fits all your business communication needs.

[[Polycom_SoundPoint_IP_501|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Polycom SoundPoint IP 601==

[[File:Voipms-polycom601.jpg|258px|thumb|left|Polycom SoundPoint IP 601]]

'''Product:''' Polycom SoundPoint IP 601

'''Company:''' Polycom

'''Overview:''' The 6-line Polycom® SoundPoint IP™ 601 offers industry-leading functionality and call handling unmatched voice quality an intuitive user interface & expandability to 12 lines!

[[Polycom_SoundPoint_IP_601|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Siemens Gigaset C450-Ip==

[[File:Siemens_gigaset_c450_IP.jpg|300px|thumb|left|Siemens Gigaset C450-Ip]]

'''Product:''' Siemens Gigaset C450-Ip

'''Company:''' Siemens

'''Overview:''' The Gigaset C450 is an entry phone in the Siemens Gigaset product line, the addition of IP connectivity via a standard Ethernet 10/100 BaseT interface and SIP protocol within a reasonable price bracket, opens large applications where wireless operation and SIP are required.
The Ethernet connectivity allows the unit to work without a PC.

The dual mode (SIP/PSTN) enables incoming call on legacy interface with existing directory number and safe emergency outgoing calls.

The presentation of the unit with an independent base and a dedicated phone charger/support will simplify cabling.

[[Siemens Gigaset C450-Ip|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==SNOM 320 VoIP Phone==

[[File:Snom320.png|300px|frame|left|Snom 320 VoIP Phone]]

'''Product:''' Snom 320 VoIP Phone

'''Company:''' Snom

'''Overview:'''Ideal for the office and everyone who spends a lot of time on the phone, the snom 320 is an affordable, yet powerful SIP business phone with built-in, full-duplex speakerphone and three-party conference bridging.

[[SNOM 320|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Snom m3 VoIP Phone==

[[File:Snom_m3.png|300px|thumb|left|Snom m3 VoIP Phone]]

'''Product:''' Snom m3 VoIP Phone

'''Company:''' Snom

'''Overview:''' The snom m9 is the next generation DECT handheld (CAT-iq) that empowers users with the convenience of wireless communication along with the widely accepted benefits and feature richness of Voice-over-IP telephony.

The snom m9 promises to deliver excellent speech quality through digital and wideband audio (klarVOICE).

By combining professional functions of versatile business communication with the intuitive features of the mobile-carrier world, the snom m9 is ideally suited for professional and private use alike. With features such as hands free mode, calling line identification (CLI) by displaying name, number and image of the caller as well as typical mobile-phone features such as Address Book, Calendar, Calculator and Alarm function, the snom m9 provides the perfect blend of mobility and accessibility.

[[Snom_m3_VoIP_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Telco AC-211==

[[File:Ac211n.jpg|300px|thumb|left|Telco AC-211]]

'''Product:''' Telco AC-211

'''Company:''' Telco

'''Overview:''' Telco AC-211 is a SIP ATA supporting two lines and numerous configuration options. This ATA is most commonly found as the AC-211-SR & AC-211N from the now defunct SunRocket service. This device works well with voip.ms once configured.

[[Telco_AC-211|See Configuration Details]]

<div style="width:100%;overflow:hidden;clear:both"></div>

==TP-Link TD-VG3631==

[[File:Tplink vg3631 front.png|300px|thumb|left|TPLink VG3631]]

'''Product:''' TP-Link TD-VG3631

'''Company:''' TP-Link

'''Overview:''' High speed DSL modem, NAT router and wireless access point in one device providing a one-stop networking solution Wireless N speed up to 300Mbps makes it ideal for heavy bandwidth consuming or interruption sensitive applications like online gaming, Internet calls and even the HD video streaming Supporting both traditional land lines and VoIP network offers you a multiple choice when making phone calls. Various call features such as caller ID, call waiting, call holding, call forwarding, 3-way conference calls and voicemail USB 2.0 port supports Storage Sharing (Samba or FTP), Media Server, Print Server and multi accounts Ethernet WAN (EWAN) offers another broadband connectivity option for connecting to Cable, VDSL or Fiber modems IP based Bandwidth Control makes it easier for you to manage the bandwidth of devices that connected to the modem router Wi-Fi On/Off Button allows users to simply turn their wireless radio on or off.

[[TP-Link TD-VG3631|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Yealink SIP-T28P (VSRF)==

[[File:Yealink-sip-t28p.jpg|300px|thumb|left|Yealink SIP-T28P]]

'''Product:''' Yealink SIP-T28P

'''Company:''' Yealink

'''Overview:''' Yealink SIP-T28P represents the next generation VoIP phone which designed for the business user who needs rich telephony features, friendly UI and super voice quality. It is equipped with the TI TITAN chipset, offers high definition voice quality through TI voice engine, HD handset, HD speaker and HD codec (G.722).

By the large, high-resolution graphical display, and together with all the 48 keys, SIP-T28P offers an excellent user experience to configure, make calls, express XML browser, etc. Moreover, to avoid problems with unwanted violations of your audio data, Yealink SIP-T28P supports the security standards TLS, SRTP, HTTPS, 802.1x, Open VPN and AES encryption which are necessary to protect against electronic eavesdropping and data theft.

[[Yealink_SIP-T28P|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>
[[Category:Devices]]

==Zycoo ZP502==

[[File:ZP502.jpg|left|Zycoo ZP502]]

'''Product:''' Zycoo ZP502

'''Company:''' Zycoo ZP502

'''Overview:''' ZP502 VoIP Phone is ZYCOO business IP phone terminal which adopts SIP and IAX2 protocols.

With Broadcom solution,compatible with various Platforms such as Asterisk, FreePBX, Broadsoft, Cisco call manager etc.

[[Zycoo ZP502|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

SNOM 320

2017-04-11T16:43:03Z

Rod:

This article makes reference to the basic configuration for the SNOM 320 phone using a [http://www.voip.ms VoIP.ms] account.

[[File:snom320.jpg|center|320px]]

== '''Documentation:''' ==
[http://downloads.snom.net/documentation/UM_snom3xx_V2.2_en.pdf User's manual]

== '''Configuration:''' ==

The initial configuration is very simple, you will need to plug the SNOM 320 phone to the internet source (modem, router), this will assign the phone with a IP address.

*Launch the web browser on your computer
*Enter the IP Address in the address field in your preferred Web Browser, e.g. http://192.168.1.10
*Press the return key

'''You can find the SNOM's IP Address by using the menu display on the phone.'''<br />
The default login details are: Username: admin, Password: admin.

The SNOM configuration page should appear.

[[File:SNOM320_1.png]]

From the menu screen click on Setup -> Identity/Line 1.

Make sure to use your own credentials such as:

Username/SIP UserID (your six digit [http://www.voip.ms VoIP.ms] sip number, or the sub account that you wish to register)
Password.
Preferred Server (The [http://www.voip.ms VoIP.ms] server of your choice)

[[File:SNOM_2.png]]

==Using the "Retrieve" [http://wiki.voip.ms/article/Voicemail Voice mail] Key==

''In the 'Function Keys' page on the web-interface of your phone, select 'Speed Dial' as the type, in the 'Retrieve' row, 'Number' field, right next to the 'Type' field, enter *97''

'''Also make sure your accounts/sub accounts does have the MWI active (Mailbox assigned)'''
[[category:SIP phones]]

== Outgoing call issues in firmware version 8 ==

Since firmware version 8 the phone is sending additional information in the SIP header which can cause SIP packets to exceed the maximum size of an IP packet that can be transmitted without fragmentation. Some routers or DSL lines cannot handle such large packets which causes the call to fail.

Depending on the MTU size your network components can process, in some cases large packets are required to be fragmented. You can find more information on how to solve this on SNOM's official wiki here http://wiki.snom.com/FAQ/How_to_solve_problems_with_outgoing_calls_in_firmware_version_8%3F

Some firmware versions of SNOM devices have all or several Audio Codecs enabled by default, this may overload the headers causing the INVITE to exceed 1518 bytes in size. To prevent this issue, try removing any codecs that are not supported by VoIP.ms. Go to Identity> RTP > Codec, and remove almost all the codecs from the test field leaving only pcmu, apply the change and reboot the phone.

Reseller Basic Guide

2017-04-03T16:25:00Z

Rod:

The Reseller feature is intended for users who want to initiate a business in the VoIP world. This useful tool can help you create and manage accounts for your own clients, you will be able to assign specific rates for them and also provide credentials to register a DID number. You will be able to handle the billing for your customers depending on your needs and how much you want to earn with your business, setting rates and adding credits for them. Additionally we will provide you with information about how to customize your Reseller portal to have a more personalized site.

The Reports section will allow you to check the Call detail records from the customers and review the Financial Reports related to their accounts. To activate your Reseller section, you need first to login to the Customer Portal then go to Reseller >> Reseller Main >> Activate

This guide will focus on the basic information to run the Reseller section properly. Here you can learn how to:

'''1. Manage Rates and Packages'''

'''2. Manage your clients'''

'''3. Use the client branding and access'''

'''4. Generate Call and Profit Reports'''

== Manage Rates and Packages. ==

This section will focus on how to create packages for the reseller feature, how to edit the rates depending on the destination and how to set a benefit for the calls your clients make. It is important to note that all the calls your customers make are charged to you as per our regulars fees, however on the reseller portal, they will see the charges according to the rates you set here.

=== Create a new package ===

From the Customer Portal go to Reseller >> Manage Rates and Packages >> Create a new Package.

[[File:CreatePackage2.png|right|650px]]

'''Package name''': Set a package name.

'''Outbound Markup''': Here you set the benefit you want to have on the calls that your clients make, you can set a fixed amount or a percentage, for this example if you set a 20%, your package will be set as follows ([cost rate] + $ fixed amount) + 20 %.

Example : for 1 cent per minute, the sell rate will be (0.01 + 0.0000) + 20 % = 0.012

'''Pulse''': This is the charge increments that you will use for this package. For example, a pulse of 60 will make calls charged by the minute. You can read more about this in our [[Calls Cost]] article.

'''Canada Route & International Route''': These are the routes that the package will use on outgoing calls. For more information between the differences between Value and Premium, please read our article [[Value vs Premium]]

'''Voip.ms to Voip.ms Calls''': This option will make it so calls between VoIP.ms DID numbers are free. Applies to calls to USA and Canada VoIP.ms Numbers.

Under the '''Fees''' Tab you will be able to enter a value that you would charge as monthly base rate and a setup rate when connecting the package to a client.

The '''Free Minutes''' Tab will let you set a total of free calls, inbound & outbound for the package. If a vlue is entered in this field, the client will have free minutes (inbound & outbound calculated) up to the number of minutes entered here. Free minutes will be calculated from the invoice date for the package and will be calculated in total rounded minutes( for example, 3 minutes 11 seconds will be calculated as 4 minutes). The destinations added to the ''Free Zones'' at Reseller > Manage Rates & Packages > Edit Rates > Manage free zone will be able to spend these Free Minutes.

The '''Reseller System Configuration''' tab offers several advanced options for the package:

* '''User can edit Sub Account''': If enabled, the end user will be able to modify details of the sub account such as Password, Device Type, Authentication Type, CallerID Override, Music On Hold, Allowed Codecs, DTMF Mode and NAT.

* '''Type of Configuration''': This option determines the configuration of the services that will be shown in the Reseller Customer Portal. If the Default Configuration is set, the Services displayed will be the ones configured at Reseller > Client Branding and Access > Services tab. If the 'Package Configuration' is set, you will be able to set them in the Services section below.

* '''View Setting''': You can set if you want to display the information in the Client Interface on ''Simple View'' or ''Standard View'' here. Simple View is a feature that allows you to show your voipinterface.voip.ms client inferface in a simpler way to your clients. This changes the balance owned to amount due, and hides some features like :
**Call Types (busy, answered, failed & noanswer) filters in the CDR
**This month and today`s amount spent on the Balance page
**Services specific information (Register type, Allowed codecs, etc) on the services page
**And some other information that would not be useful for "vonage style" clients

* '''Services''': In this section you can select which services will be available and which services will not be available for your customers, if you are using the "Package Configuration" option in the Type of Configuration section. (These Options are only available in the New Version of the reseller system)

=== Edit Rates ===

If you want to set different rates depending on the destination of the call, you can do this on the Edit Rates section from the package, all the custom rates you enter there will override the global markup. You can set a custom rate for all the calls to USA or Canada, by country or Calling code. You can also set your '''Free Zones''' here, which will be the destinations that will be included in the Free Minutes offered in the package.

[[File:EditRatesPackage.png|center|thumb|600px]]
<br clear="all" />

== Manage your clients ==

Now that you have a package, you need to add a client, this section will focus on how to create a client and associate it with a [http://wiki.voip.ms/article/Sub_Accounts sub account], as well as adding payments and charges for each client, assigning a DID number and also the proper way to delete a client account that is not longer needed.

[[File:ManageClientsAccounts.png|center|thumb|800px]]
<br clear="all" />

=== Adding a client ===

From the customer portal go to Reseller menu >> Manage Clients acct >> Create a new client account.

[[File:CreateClient.png|center|600px|thumb]]
<br clear="all" />

Fill the form with the client information and choose between soft and hard balance, when finished click on create account. A customer in HARD balance will be suspended if he is on zero or negative balance, while a customer in SOFT balance will be able to keep making use of the service even if their balance is zero or negative.

On the Manage section, you can add a new package or a DID number. To add numbers to your clients you need to order the number from your Customer Portal first. Don't forget to set a Per Minute rate and/or Monthly fee for your number, as incoming calls are charged independently from the outgoing calls.

Once the client account has been created, you will need to create a [http://wiki.voip.ms/article/Sub_Accounts sub account] for it as well.

Go to Sub accounts menu >> Create Sub account. During the Sub Account creation process, you will reach the Reseller configuration section, here you can choose:

[[File:ResellerConfiguration.png|right]]

*Reseller client

*Reseller client Package

*Next Billing date

*And if you decide to charge the set up fees now, make sure to check the appropriate box.

If you already have a [http://wiki.voip.ms/article/Sub_Accounts sub account] created, you can modify it at Sub Accounts >> Edit Sub Account >> Manage and add a package there.

=== Delete a client ===

You can delete a client on the Edit section from the Client's account, however note that before doing this you need to be sure to unconnect any package or DID number related to the client, you can do this from the Manage section on Client´s account.

=== Adding Payments manually ===

You can add payments manually from the customer portal >> Reseller menu >> Manage Clients acct >> Manage Client, then select the Payment button under the Balance panel.

=== Add Charges manually ===

You can add charges manually from the customer portal >> Reseller menu >> Manage Clients acct >> Manage Client, then select the Charge button under the Balance panel.

=== Add phone numbers to your clients (DIDs) ===

You can add phone numbers to your clients from the Customer portal >> Reseller menu >> Manage Clients acct >> Manage Client >> Package for this clients >> Add Phone number. Please note that you need to order a DID and have it on your account before you can assign it to a client. Don't forget to set a Per Minute rate and/or Monthly fee for your number, as incoming calls are charged independently from the outgoing calls.

[[File:AddPhone.png|center|thumb|600px]]
<br clear="all" />

== Use the client branding and access ==

It is always important to show a good brand image to your customer, for that reason we offer the option to customize our reseller interface to have the best presentation possible. There is also a useful feature on the site that can allow us to integrate your client's payments with a paypal account, so they can be added automatically as funds to the client.

=== Branded Interface ===

At the top of the screen you will see two available options, these are:

'''Version of the reseller system:''' This is the version of the Reseller System that will be displayed to your clients. The New Version of the Reseller System offers several customization options as well access to most of the VoIP.ms services, while the Old Version offers limited customization and access to basic services such as direct SIP routing and [[Call Forwarding]]

'''Configure Reseller Version:''' Determines which version of the Reseller System you will be configuring on this section. Enabling the New Version will offer you the option to modify the color of the panels, buttons and background of your Client Portal, as well as previewing it before making any changes.

=== Interface and Access ===

'''Clients Interface Access:''' This section will list the URL that you can give your clients to access their Account.

'''Auto Connect Form for Clients:''' This is an HTML form that you can use to automatically send your clients to their interface without needing authorization.

'''DID Access:''' You can set the pricing here for any DIDs that your clients purchase using the option at Services > Order DIDs on their Client Portal, if you have enabled this service for them.

=== Branding Management ===

In this section you will be able to change the Title that will appear in your Reseller Customer Portal page, as well as any of the colors of the interface, including the buttons* and the panels*, as well as selecting a Header, logo* and login background* for the interface.

<nowiki>*Options only available in the New Version of the reseller system.</nowiki>

=== Display Elements ===

Here you can choose whether or not you'd like to display the SIP and IAX password(s) in the Reseller Interface, as well as allowing your customers the option to reset their passwords in the Login Reseller Interface.

Lastly, you will be presented with an option to integrate an automated signup form into your website. If so, you can link your signup menu option to the URL presented there. You can also select which fields will be shown in order to sign up with a new account with you.

=== Services ===

View Setting: You can set if you want to display the information in the Client Interface on 'Simple View' here. Simple view is a feature that allows you to show your voipinterface.voip.ms client inferface in a simpler way to your clients. This changes the balance owned to amount due, and hides some features like :

*Call Types (busy, answered, failed & noanswer) filters in the CDR
*This month and today`s amount spent on the Balance page
*Services specific information (Register type, Allowed codecs, etc) on the services page
*And some other information that would not be useful for "vonage style" clients

Services*: In this section you can select which services will be available and which services will not be available for your customers, if you are using the "Default Configuration" option in the Type of Configuration section of the Package settings.

<nowiki>*Options only available in the New Version of the reseller system</nowiki>

=== Text and Messages ===

In this tab you will be able to personalize the Welcome Text that will appear in the Reseller Customer Portal. You are also able to customize the Emails that will be sent when a Customer's Balance is running low, as well as when they request and complete a Password Reset.

== Payment ==

If you want your clients to deposit directly in your paypal account, you can activate the paypal integration. All payments will be sent to your paypal account and automatically credited in your client interface.

[[File:PaymentIntegration.png|center|thumb|600px]]
<br clear="all" />

Instructions :
#Enter your paypal email address in the "paypal email account" field
#Enter the amounts that you want to offer, without decimal and separated by a coma
#In your paypal account, go to "Profile" , and under the "Selling preference" section, activate the Instant payment notification(IPN). Enter the url : http://www.voipinterface.net/paynotify.php .
#We strongly recommend to make a live test with an amount of $1.

== Host Name ==

In order to personalize the interface with a hostname

#Configure a host with an A RECORD in your dns that points to 174.120.50.58
#Enter in the field bellow the fully qualified host + domain name (e.g: interface.mydomain.com)
#Click "submit"

There is a one time fee of $10.00 for adding your hostname to our system.

== Reports ==

Here you can see the Call Detail Records for the calls your customers have made and received, with the rates we have established on the packages created, showing a net total of the benefit earned as well.

Reports are divided in the following categories:

*'''Global CDR''': Display cdr report for all your reseller clients calls. You can see all the calls (incoming and outgoing) from all your clients.

*'''Per User CDR''': Display cdr report for a specific client. This filter help you to search calls only for one specific client.

*'''Financial report''': Month to month revenues ,deposits, costs related to the reseller clients. You can see the financial information by month.

*'''Sub accounts report''': List of all sub accounts connected to the reseller clients ,with related information. Here you will find a list of your clients, [http://wiki.voip.ms/article/Sub_Accounts sub account], package and DID number associated with each one.

*'''DID(s) Report''': List of all DID's connected to the reseller clients, with related information.

[[category:guides]]

Voicemail

2017-03-29T15:11:27Z

Rod: /* Voicemail Greeting Customization */

VoIP.ms has an Advanced Voicemail feature that is free to use, and you also have the option to forward your messages to your email address as an attachment.

In order to use the Voicemail feature with Voip.ms you will have to create a Voicemail entry and then assign your entry to one of your DIDs or Accounts.

*The total allowed voicemail messages on a Single Mailbox is 100.
*Please white list the mail domain names *@voip.ms and *@voipinterface.net.
*The Maximum Time for a Voicemail message is 3 minutes.
*5 Seconds of Silence will end a Voicemail Recording.

== Voicemail Accounts ==

From the Customer Portal refer to DID Numbers -> Voicemail You will see the following screen. In there you will be able to see a list of any existing Voicemail Accounts, or create a new Voicemail Account (Mailbox) by clicking on the "Create New Voicemail Account" Button.

[[File:VoicemailAccts.png|900px]]

== Creating a Voicemail mailbox ==

Pressing the "Create New Voicemail Account" button will display a new window.

[[File:NewVoicemailAcc.png|370px]]

You will be prompted with the following information:

'''Voicemail Number:''' This will be used as a unique identifier for your mail box. The minimum is adding one digit to up to ten digits, for example you can set 1 or 5432897.

'''Name:''' This can be used as a note or description to easily identify your mail boxes.

'''Password:''' The password is used to enter your mailbox options such as listen to your messages or record your greeting.

'''Skip Password Prompt:''' If set to Yes, when dialing *97 from an account associated to this mailbox, it will skip the password prompt and login directly.

'''Notification Email:''' If an email address is entered here, the Mailbox system will send an Email notification every time you receive a new message. For the moment you can only set 1 email address, however you can optionally configure an Email forward between your email accounts as a work around.

'''Voicemail Language:''' This sets the language you and the caller will hear when instructions or menus are played by the voicemail.

Pressing the "View Advanced Mode" button will display further options to modify your mailbox.

[[File:NewVoicemailAdv.png|900px]]

'''Unavailable Message Recording:''' This is the greeting that the system will play to the callers that reach your voicemail. There are three options available for this recording:

*System Default - The system will use the generic greeting message recording when the callers reach the Voicemail.

*Set by Phone - The system will use the recording set by the phone by the user via dialing *97 / *98 and choosing option 0 to record it.

*Recordings - The system will use the associated [[Recordings]] previously uploaded via the Customer Portal tool at DID Numbers > Recordings.

'''Attach message to email:''' If set to YES, the Mailbox will attached a .WAV file containing the new message every time it sends an Email notification.

'''Delete Voicemail Message''' If set to YES, the Mailbox will delete the new message automatically after sending the Email notification with attachment.

'''Attachment Format:''' You can select between WAV49, recommended because of its smaller size and easiness to handle; WAV which is uncompressed, causing a bigger file size; and mp3 which offers great compatibility and is very small.

'''Say Instructions:''' If set to YES, the caller will hear instructions on how to leave a message to your Mailbox before the beep sound.

'''Say Time / Date:''' If set to YES, when checking your messages you will hear the date and time when the message was received.

'''Time Zone:''' The time envelope will use this time zone to provide the correct date and time of the message's reception.

'''Say Caller ID:''' If set to YES, when checking your messages you will hear the Caller ID of the message sender.

== Assigning your Voicemail to your DID==

After you have created your Voice Mail entry, you can assign it to any of your DIDs from your main portal. Please refer to DID Numbers -> [[Manage DID]] -> Edit DID -> Voicemail. Also under the same screen you can set the Ring Time (The maximum amount of time a call to your DID can stay in "Ringing State" before we cancel the call to no answer). Please note that 30s equals to 6 rings.

[[File:selectVoicemail.png|750px|thumb|left|Select Voicemail - click to enlarge]]
<br clear="all" />

== Assigning your Voicemail to your Account==

If you would like to assign a Voicemail entry to your Main Account, please from your main portal refer to Main Menu -> [[Account Settings]] -> General -> Voicemail Associated to the Main Account.

[[File:Mainvoicemail.jpg]]

If you need to assign a Voicemail to a specific subaccount, you need to go to the [[Sub Accounts]] Edit page, following the route, Subaccounts >> Manage Subaccounts >> Edit, from the menu tabs.
You will see at the bottom of the page the "Internal Extension Voicemail" option. Here you can set it.

[[File:Subvoicemail.jpg]]

== Manage Voicemail and Messages ==

[[File:VoicemailAccts.png|800px]]

Your mailbox will appear under '''Voicemail Accounts''' after it is created through your Customer Portal>>DID Numbers>>Voicemail page. In there, you can click the '''Delete''' button to delete the Mailbox completely or '''Delete Msgs''' to delete all of the messages in the given mailbox.

If you click '''Edit''', a new window will be displayed where you will be able to change the settings in your voicemail. However, please note that you cannot change the Mailbox ID once it has been created.

You can also manage your voicemail messages by clicking on the '''List''' icon next to your mailbox. Within this new window, you will be able to listen to a message, mark it as Urgent, Forward this message to an email address, download the audio file, mark it as "read", store it in a folder or delete it.

[[File:Voicemail List.png|800px]]

== Voicemail Greeting Customization ==

If you want to change the default voicemail greeting on your mailbox you will have 2 options.

* The '''First''' option is to do it directly from your Phone, once you have created the Mailbox and you have assigned it to your DID number and to your account, dial *97 from your phone and at the Voicemail menu select the option "0" (Voicemail options) and then the option "1" (Unavailable message) there you will be able to record your greeting, and to save it. Any greeting set this way will take precedence over any greeting options, including uploading a Recording or using the System Default greeting.

* The '''Second''' option, is in case that you already have the recording that you want to use as your voicemail greeting. You can upload this recording to DID Numbers > [[Recordings]] in the Customer Portal. Your uploaded recording will then appear as a greeting option at DID Numbers > Voicemail > Edit Mailbox > View Advanced Mode (Upper Right) > Unavailable Message Recording.

*Please note that you must change the '''Unavailable Message Recording''' option of your Voicemail account to the desired greeting for the changes to be reflected.

*Recording your greeting following the First option, using your phone and the audible Voicemail system, will take precedence over any Recording uploaded or the System Default.
If you wish to use an uploaded Recording or the System Default greeting, please delete your recorded greeting first.

== Voicemail Access Codes: ==

You can access your voicemail with any device/system connected directly with your account or subaccount to VoIP.ms using the codes below:

*<nowiki>*97</nowiki> (Asterisk 97) is used to access directly the Mailbox associated to the account you are dialing from. If you would like to check which mailbox is associated to your account refer to [[Voicemail#Assigning_your_Voicemail_to_your_Account|Assign Voicemail]]

*<nowiki>*98</nowiki> (Asterisk 98) is used to access your Voicemail and choose one of your Mailbox accounts. (Will prompt for Mailbox ID and Password)

If for any reason you do not have access to our VoIP network, you can check your Voicemail by just dialing your DID. Once the Voicemail system answers your call, press the asterisk key (*). You will be prompted for the Mailbox ID (or hear the mailbox automatically selected, depending on the server you are using) and Password, once logged in to your Voicemail, press 0 (zero) for options. You can also record your greeting from there by selecting option 4 (four). Please be aware this is considered a regular incoming call and will be charged according to your monthly DID plan.

== Navigate the Voicemail Menu ==

Once you access your voicemail you're going to be prompted with the number of new and/or old messages you have in the mailbox. Here's the list of options you have with the voicemail system of VoIP.ms

* 1 - Play the first new/old message available in your mailbox.

* 2 - Change folders. This option allows you to change to another folder in order to hear the messages you have stored in that folder. At the moment it's not possible to change the name of the folders.
0 - New Messages
1 - Old Messages
2 - Work
3 - Family
4 - Friends
# - Cancel

* 0 - Voicemail options. In here you can change your greetings and record your name, also you can change the password for your voicemail.

1 - Unavailable message
2 - Busy message
3 - Name.
4 - Temporary message
5 - Change Password
* - Return to the Main Menu

'''Notes:'''
* The Temporary Message will override the Voicemail message option set in the Customer Portal until it is deleted through your phone by dialing *97, 0 for options, 4 for Temporary Message and then 2 to Delete it.
* To Delete a Recording you can just record again, a moment of silence, or create a new recording overwriting the old.
* Please note that if you change the password over the phone ('''Option 5'''), it will not change at your customer portal>>Voicemail options,
and you will have to manually change the password there. If you don't don't change it at the customer portal and you modify any other value
later, when you apply the changes the old password will be set again to your mailbox, since the modify option apply all the values set at
the voicemail options.

'''The following options are available when you're listening to your messages.'''

* 3 - Advanced Options
1 - Send a reply. Currently not available.
2 - Message Envelope. Speak the date and time at which the message was received.
* - Return to the Main menu.

* 4 - Play the previous message.

* 5 - Repeat the message.

* 6 - Play the next message.

* 7 - Delete the current message, without confirmation.

* 8 - Forward message to another user. When prompted for an extension, you must enter the full mailbox ID of the intended destination (e.g. 52739100).
1 - Prepend the message with a recording.
2 - Send the message without a prepending message.
* - Return to the Main Menu

== Known Voicemail Issues or Frequent Questions ==

'''I Can No Longer Listen to My Voicemail on my Android Phone.'''

VoIP.ms uses a compressed WAV format for the voicemail messages (WAV49).

This helps server stability and lessen the bandwidth used by VoIP.ms customers to download the file.
Please download another FREE Audio Player like Remote Wave and this will resolve this issue instantly.
You will be given a choice of what application to use to listen to the file or you can make it your Default Audio Application.
If this doesn't work for you, you can always go back to uncompressed WAV format, on your Manage Mailbox options.

'''My Voicemail Messages keep stopping at 3 minutes.'''

The Current Maximum Length, a voicemail message can be, is 3 minutes.

'''I Keep Receiving Cut Off Voicemails from People'''
While recording a message the system listens for 5 seconds of silence. If the systems detects no sounds for 5 seconds it will
Stop the VM Recording and send a person to VM options.

'''How Can I Delete All Voicemail Messages in My Mailbox.'''

You can only delete all messages from your Customer Portal>> DID Numbers>> Voicemail page. Please refer to [[Voicemail#Manage_Voicemail|Manage Voicemail]].

'''I have Voicemail Messages but when I login the system tells me I Have No Voicemails'''

It is possible that you have selected the option to Delete Voicemail Message once it has been sent to email.
Please see the [[Voicemail#Creating_a_Voicemail_mailbox|Creating a Voicemail Mailbox Section]] to change this option.

'''I Dial *97 But the System Still Asks for Mailbox Number'''
In order for *97 to work you have to make sure the device or softphone you are using is connecting to the servers with a SIP account associated with that Mailbox.
So if you are connecting with your main account or a sub account, please see [[Voicemail#Assigning_your_Voicemail_to_your_Account|Assigning Your Voicemail to Your Account]].
If you are connecting with a phone not connected to VoIP.ms service then for security reasons you will have to enter the mailbox number and password.

[[category:guides]]

Choosing Server

2017-03-22T18:10:40Z

Rod:

[[File:Serverlocation3.png]]

= Choosing a Server =

[http://www.voip.ms VoIP.ms] offers many different servers, but which one should you choose? One misconception is that you should pick the closest to your location, however this is not needed most of the time. For example, if you are in the USA, any of the US servers will provide a really good latency and service quality. The newest server within a city is indicated with the highest number attached to the name, as they are classified in ascending order. Also worth noting is that there is a network tool that will help you when deciding which server you want to use, generally named a "ping", which will provide you the latency between you and the server. Therefore the server which provides you less latency should be used.

=== IPs ===

*Amsterdam, NL ('''amsterdam.voip.ms''') 37.58.88.242
*Atlanta 1, GA ('''atlanta.voip.ms''') 75.127.65.130
*Atlanta 2, GA ('''atlanta2.voip.ms''') 72.9.246.170
*Chicago 1, IL ('''chicago.voip.ms''') 208.100.39.52
*Chicago 2, IL ('''chicago2.voip.ms''') 208.100.39.53
*Chicago 3, IL ('''chicago3.voip.ms''') 208.100.39.54
*Chicago 4, IL ('''chicago4.voip.ms''') 208.100.39.55
*Dallas, TX ('''dallas.voip.ms''') 158.85.149.162
*Dallas 2, TX ('''dallas2.voip.ms''') 158.85.149.163
*Denver 1, CO ('''denver.voip.ms''') 173.248.161.90
*Denver 2, CO ('''denver2.voip.ms''') 173.248.159.210
*Houston, TX ('''houston.voip.ms''') 173.193.85.18
*Houston 2, TX ('''houston2.voip.ms''') 173.193.85.19
*London, UK ('''london.voip.ms''') 159.8.157.212
*Los Angeles 1, CA ('''losangeles.voip.ms''') 96.44.149.186
*Los Angeles 2, CA ('''losangeles2.voip.ms''') 96.44.149.202
*Melbourne, AU ('''melbourne.voip.ms''') 168.1.73.84
*Montreal 1, QC ('''montreal.voip.ms''') 67.205.74.184
*Montreal 2, QC ('''montreal2.voip.ms''') 67.205.74.187
*Montreal 3, QC ('''montreal3.voip.ms''') 192.175.96.68
*Montreal 4, QC ('''montreal4.voip.ms''') 67.205.74.179
*Montreal 5, QC ('''montreal5.voip.ms''') 192.175.96.69
*Montreal 6, QC ('''montreal6.voip.ms''') 192.175.96.70
*Montreal 7, QC ('''montreal7.voip.ms''') 192.175.96.71
*Montreal 8, QC ('''montreal8.voip.ms''') 192.175.96.72
*New York 1, NY ('''newyork.voip.ms''') 74.63.41.218
*New York 2, NY ('''newyork2.voip.ms''') 107.6.67.236
*New York 3, NY ('''newyork3.voip.ms''') 107.6.67.237
*New York 4, NY ('''newyork4.voip.ms''') 107.6.67.238
*New York 5, NY ('''newyork5.voip.ms''') 23.29.136.28
*New York 6, NY ('''newyork6.voip.ms''') 23.29.136.29
*New York 7, NY ('''newyork7.voip.ms''') 23.29.136.38
*New York 8, NY ('''newyork8.voip.ms''') 23.29.136.40
*Paris, FR ('''paris.voip.ms''') 159.8.85.180
*San Jose, CA ('''sanjose.voip.ms''') 23.246.247.146
*San Jose 2, CA ('''sanjose2.voip.ms''') 23.246.247.147
*Seattle 1, WA ('''seattle.voip.ms''') 50.23.160.50
*Seattle 2, WA ('''seattle2.voip.ms''') 50.23.160.51
*Seattle 3, WA ('''seattle3.voip.ms''') 50.23.160.52
*Tampa, FL ('''tampa.voip.ms''') 162.254.144.173
*Tampa 2, FL ('''tampa2.voip.ms''') 162.254.144.176
*Toronto 1, ON ('''toronto.voip.ms''') 158.85.70.148
*Toronto 2, ON ('''toronto2.voip.ms''') 158.85.70.149
*Toronto 3, ON ('''toronto3.voip.ms''') 158.85.70.150
*Toronto 4, ON ('''toronto4.voip.ms''') 158.85.70.151
*Toronto 5, ON ('''toronto5.voip.ms''') 184.75.215.106
*Toronto 6, ON ('''toronto6.voip.ms''') 184.75.215.114
*Toronto 7, ON ('''toronto7.voip.ms''') 184.75.215.146
*Toronto 8, ON ('''toronto8.voip.ms''') 184.75.213.210
*Vancouver 1, BC ('''vancouver.voip.ms''') 162.213.157.82
*Vancouver 2, BC ('''vancouver2.voip.ms''') 162.213.157.117
*Washington 1, DC ('''washington.voip.ms''') 208.43.234.226
*Washington 2, DC ('''washington2.voip.ms''') 208.43.234.227

===Server Realms===

For IOS, Please click here [http://wiki.voip.ms/article/Server_Realms Server Realms] to get the Realm Name for the server you plan on using, this can differ from the Domain Name being used.

Please note that the following servers will not be available to select as a DID Point of Presence for newer accounts:
Atlanta 1, Chicago 1, New York 1, New York 4, Seattle 1, Montreal 1, Montreal 2, Montreal 3, Montreal 4, Toronto, Toronto 2, Toronto 3, Toronto 4.

= What is a Ping? =

Ping is a standard tool used to test network connections. It is mostly used to determine if a server or device can be reached across the network and the latency of the response(the time it takes to send a packet to the destination and for it to return to your computer).

Ping tools are part of Windows, Mac OS X and Linux as well as some routers.

== How does the ping work? ==

It sends request messages to a target network address or DNS names at periodic intervals and measures the time it takes for a response message to arrive and return(better known as latency).

==How to ping on a PC==

Pinging is a command which tells you if the connection between your computer and a particular domain is working correctly.

In Windows, select Start > Programs > Accessories > Command Prompt. This will give you a window like the one below.

Enter the word ping, followed by a space, then the domain name.(montreal.voip.ms) in this case domain is our server name.

If the results show a series of replies, the connection is working. The time shows you how fast the connection is. If you see a "timed out" error instead of a reply, there is a breakdown somewhere between your computer and the domain.

[[File:Ping.gif]]

==How to ping on a Mac Computer==

1- Click on Finder in the dock.

2- Click on Applications.

3- Click on Utilities.

4- Double-click on Network Utility. *

* In OS X Mavericks (10.9.x) this utility app changed location. Launch it from spotlight instead, either press "command"+"space bar" or click on spotlight directly (magnifying glass icon at top right of screen), type "network utility" and hit "return"

5- In the Network Utility window, click on the Ping tab

6- In the field under "Please enter the network address to ping," like montreal.voip.ms

''If pings results are not consistent, you may have an issue with Jitter. You can work on this issue by adjusting the "Network Jitter Level" setting on your VoIP device. Usually a ping of under 150 ms is recommended in order to have good quality. The latency time to the server is important, however there are also other factors that could affect the quality of the calls such as packet loss (VoIP communications are very sensitive to this), and the Jitter level of your Internet connection.''

The following is the output of running ping with the target losangeles.voip.ms.

#ping losangeles.voip.ms
Ping to losangeles.voip.ms [67.215.241.250] with 32 bytes de datos:
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
Response from 67.215.241.250: bytes=32 time=69ms TTL=52
Response from 67.215.241.250: bytes=32 time=68ms TTL=52
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
ping statistics from 67.215.241.250:
4 packets transmitted, 4 received, 0% packet lost. rtt min/avg/max/mdev = 67ms, 69ms, 67ms

Sample ping output in windows:
C:\Windows\system32>ping montreal.voip.ms

Pinging montreal.voip.ms [67.205.74.184] with 32 bytes of data:
Reply from 67.205.74.184: bytes=32 time=85ms TTL=49
Reply from 67.205.74.184: bytes=32 time=86ms TTL=49
Reply from 67.205.74.184: bytes=32 time=86ms TTL=49
Reply from 67.205.74.184: bytes=32 time=85ms TTL=49

Ping statistics for 67.205.74.184:
Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 85ms, Maximum = 86ms, Average = 85ms

= Latency Testing Scripts (User Submitted) =
<p>All the following scripts were produced by voip.ms users who felt others might also benefit from the output of their efforts. They were written over a span of Years and probably need adjusting before you use them, to cater for changes in servers over time and changes in policies (like not testing heavily subscribed servers which are not open to new registrations)</p>
<p>If you aren't satisfied that the scripts are safe or simply don’t like the way they syntactically appear, you can still manually ping a selection of servers and choose a server based on the best latency. The following scripts are essentially just wrappers around the ping command which support lists of servers to feed to ping and present the output in a readable format.</p>
<p>If you feel you have a simpler cleaner script that works for another platform or language, please do add it to this wiki via a support ticket.
</p>
=== Bash Script To Handle The Mac Ping Output Format ===

<p>To make use of this script (1) save as a plain text file (2) set permissions on the file to executable (3) invoke script
e.g. Save script below using your favourite editor as pingVoipMS.sh (2) chmod u+x pingVoipMS.sh (3) ./pingVoipMS.sh
This is a bash 3.x script, so it also works in Linux, just change the ping packet loss field from 7 to 6 in the final loop below (or wherever the loss field is in your ping output format). Depending upon your distro curl might need to change to wget.
</p>
<pre>
#!/bin/bash
# Ping several servers and display Latency, Jitter and Packet Loss
# Usage: [-c <count>][-i <wait time>][-r test restricted servers][<server list file>]
#
# The optional text file should be formatted with one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
# If no args are supplied, this script will scrape a ping server list from voip.ms
#
USER_FILE=""
COUNT=3; INTERVAL=5; RESTRICTED=0
restrictedList=(atlanta.voip.ms chicago.voip.ms
montreal.voip.ms montreal2.voip.ms montreal3.voip.ms montreal4.voip.ms
newyork.voip.ms newyork4.voip.ms seattle.voip.ms
toronto.voip.ms toronto2.voip.ms toronto3.voip.ms toronto4.voip.ms)

# Handle any passed in script arguments
while getopts c:i:r parm
do
case $parm in
c)count_opt=$OPTARG;;
i)interval_opt=$OPTARG;;
r)RESTRICTED=1;;
*)echo -e "Invalid arg\nUsage:\t[ -c <count of ECHO_REQUESTs to Tx, default 3> ] \
\n\t[ -i <wait time (s) between datagrams, default 5> ] \
\n\t[ -r ] Include restricted servers in latency test \
\n\t[FILE <ping server list> ]";exit 1;;
esac
done

# Test if an option was specified and whether it's a +ve non-zero integer
[[ -n $count_opt && ($count_opt =~ ^[[:digit:]]+$) && $count_opt -gt 0 ]] && COUNT=$count_opt
[[ -n $interval_opt && ($interval_opt =~ ^[[:digit:]]+$) && $interval_opt -gt 0 ]] && INTERVAL=$interval_opt

shift $(($OPTIND -1))

# Validate supplied file (server list)
[[ -n $1 && !(-f $1 && -r $1) ]] && { echo "\"$1\" file does not exist or is not readable"; exit 1; }
[[ -n $1 && -f $1 && -r $1 ]] && USER_FILE="$1"

if [[ -n $USER_FILE ]]
then
# Bash 3.x in macOS does not support readarray, need to do cumbersome array loops instead
while IFS= read -r line; do
serverList+=( "$line" )
done < <(grep -e '[[:alpha:]]*[[:alnum:]]\.voip\.ms' $USER_FILE | grep -v '^\s*#' | awk NF)
else
# N.B. The script looks for the html boldface tags <b> </b> inside a bracket
# If the website alters and the parse fails, manually create the list and
# supply as a script arg (or perhaps update the parsing to work again :)
while IFS= read -r line; do
serverList+=( "$line" )
done < <(curl --silent http://wiki.voip.ms/article/Choosing_Server | \
grep '(<b>[[:alpha:]]*[[:alnum:]]\.voip\.ms</b>)' | \
tr "<>" " " | awk '{print $(NF-3)}' )
fi

# Newer voip.ms clients can't register onto these over-subscribed servers
# Don't test the restricted list unless explicitly asked (with the -r cmd line option)
if [[ $RESTRICTED -eq 0 ]]
then
for server in ${restrictedList[@]}
do
ix=$(printf "%s\n" "${serverList[@]}" | grep -n "^${server}" | cut -d ":" -f1)

while IFS= read -r idx; do
keys+=($idx)
done < <([[ $ix -gt 0 ]] && echo $((ix-1)))
done
for ((i=${#keys[@]} - 1; i >= 0; i--)); do unset serverList[keys[i]]; done
fi

if [[ ${#serverList[@]} -eq 0 ]]
then
echo "No unrestricted Voip.ms servers could be found, please supply a server list"
exit 1
fi

runTime=$(( $COUNT * $INTERVAL * ${#serverList[@]} ))

echo "PING will send $COUNT packet(s) with a wait of $INTERVAL sec(s) between each packet"
echo "Change the PING options by invoking this script with -c and/or -i, default \"-c 3 -i 5\""
echo "Over $((${#serverList[@]})) server(s) the estimated script Run Time will be $runTime seconds"
echo "================================================================"
printf "%-20s %-18s %7s %8s %6s %s\n" "VoIP Server" "IP Address" "Latency" "Jitter" "Loss" "Countdown"
echo "================================================================ (seconds)"

for myLn in ${serverList[@]}
do
while IFS=$'\n' read -r line; do
pingList+=( "$line" )
printf "%-64s %5d %2d/%-2d\n" "$line" "$((runTime - $COUNT * $INTERVAL * ${#pingList[@]}))" "${#pingList[@]}" "${#serverList[@]}"
done < <( ping -c $COUNT -i $INTERVAL -q $myLn | awk '\
/^PING / {myH=$2}
/^PING / {
IP = substr($3,2,15)
split(IP,myIP,")") }
/packet loss/ {myPL=$7}
/min\/avg\/max/ {
split($4,myS,"/")
printf("%-20s %-18s %7.3f %8.3f %6s\n",
myH, myIP[1], myS[2], myS[4], myPL ) }
' )
done

echo "================================================================"
echo -e "\nMost appropriate server listed in order of best latency\n"
echo "================================================================"
printf "%-20s %-18s %7s %8s %6s\n" "VoIP Server" "IP Address" "Latency" "Jitter" "Loss"
echo "================================================================"
printf "%s\n" "${pingList[@]}" | LC_ALL=C sort -n -k 3,3 -k 5,5 -k 4,4 | awk '{printf("%s $%2d$\n",$0, NR)}'
echo "================================================================"
</pre>

=== Perl Script ===
Pings list of voip.ms servers round robin with optional output csv file.

# usage ping_voip.ms.pl <number of times> <seconds in between> <output.csv>
use Net::Ping;
use Time::HiRes;
use strict;

# input list
my @hosts = qw(
atlanta.voip.ms
atlanta2.voip.ms
chicago.voip.ms
chicago2.voip.ms
chicago3.voip.ms
chicago4.voip.ms
dallas.voip.ms
denver.voip.ms
denver2.voip.ms
houston.voip.ms
losangeles.voip.ms
losangeles2.voip.ms
newyork.voip.ms
newyork2.voip.ms
newyork3.voip.ms
newyork4.voip.ms
seattle.voip.ms
seattle2.voip.ms
seattle3.voip.ms
tampa.voip.ms
washington.voip.ms
washington2.voip.ms
montreal.voip.ms
montreal2.voip.ms
montreal3.voip.ms
montreal4.voip.ms
toronto2.voip.ms
toronto3.voip.ms
toronto4.voip.ms
toronto.voip.ms
london.voip.ms
);

$| = 1; #autoflush
# High precision syntax (requires Time::HiRes)
my $p = Net::Ping->new("icmp",1);
$p->hires();
my $max_name_length = (reverse sort { $a <=> $b } map { length($_) } @hosts)[0];
my $count = 4; # number of times to ping
my $interval = 5; # seconds between ping rounds
my $output_file = "";
my @data;

# check for arguments
my $num_args = @ARGV;
if ($num_args >= 1) {$count = $ARGV[0];}
if ($num_args >= 2) {$interval = $ARGV[1];}
if ($num_args >= 3) {$output_file = $ARGV[2];}

# check argument validity
$0 =~ /^.*\\(.*)$/;
my $script = $1;
if ($count !~ /^\d+$/ or $interval !~ /^\d+$/) {die "Usage: $script <number of rounds> <seconds between rounds> <output.csv>\n";}
if (length($output_file) > 0 and $output_file !~ /\.csv$/) {$output_file .= ".csv";}

# main loop
for my $i (1..$count)
{
sleep $interval unless $i == 1;
print "Round $i\n";
my $host_num=0;
foreach my $host (@hosts)
{
(my $ret, my $duration, my $ip) = $p->ping($host);
$ip =~ /(\d+)\.(\d+)\.(\d+)\.(\d+)/;
if ($ret)
{
printf("%*s [ip: %3s.%3s.%3s.%3s] is alive (%6.2f ms)\n", $max_name_length, $host, $1, $2, $3, $4, $duration*1000);
$data[$host_num][$i]=$duration*1000;
}
else
{
printf("%*s [ip: %3s.%3s.%3s.%3s] is dead\n", $max_name_length, $host, $1, $2, $3, $4);
}
$host_num++;
}
print "\n";
}
$p->close();

# if output file name given
if (length($output_file)>0)
{
# print output to file
open FILE, ">$output_file" or die "$!\n";

# print column headers
print FILE "Server\\Round";
for my $i (1..$count)
{
print FILE ", $i";
}
print FILE ", Average\n";

# print data
my $i = 0;
foreach my $host (@hosts)
{
print FILE "$host";
my $sum = 0;
for my $j (1..$count)
{
$sum += $data[$i][$j];
printf FILE ", %8.4f",$data[$i][$j];
}
printf FILE ", %8.4f\n",$sum/$count;
$i++;
}

close FILE;
print "Data written to $output_file\n";
}

# print summary to screen
my $i = 0;
printf("%-*s Average (ms)\n", $max_name_length, "Server");
foreach my $host (@hosts)
{
my $sum = 0;
for my $j (1..$count)
{
$sum += $data[$i][$j];
}
printf("%-*s %8.4f\n", $max_name_length+1, $host, $sum/$count);
$i++;
}

Output:
Round 1
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.97 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.99 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.70 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.76 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.53 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.73 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 94.99 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.05 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.13 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (102.87 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 64.92 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 63.41 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (131.75 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (120.64 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (120.49 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (111.43 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.25 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.86 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 90.85 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (123.29 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.71 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.19 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 81.82 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 86.13 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 77.09 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.18 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.70 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (131.27 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (125.13 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.26 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (152.77 ms)

Round 2
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.14 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.86 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 50.03 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.44 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.33 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.22 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 95.58 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.94 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.29 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (102.73 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.59 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 64.27 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (112.74 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (121.22 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (121.34 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.75 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.06 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.33 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 91.58 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (122.94 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.28 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.40 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 81.91 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 85.64 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 75.15 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.79 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.10 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (150.85 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.40 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.45 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (170.79 ms)

Round 3
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.76 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.86 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.65 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.01 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.05 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.53 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 95.82 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.02 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.60 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (103.35 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.79 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 64.05 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (113.01 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (121.41 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (122.23 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.62 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 93.65 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.19 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 90.75 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (125.12 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.19 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.98 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 80.16 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 87.16 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 76.54 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 97.51 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.18 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (142.81 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.95 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.78 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (153.14 ms)

Round 4
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 89.19 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.98 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.21 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.50 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.68 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.18 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 93.93 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.22 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.10 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (103.67 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.58 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 63.60 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (114.76 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (120.44 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (121.05 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.51 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.04 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 96.92 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 91.23 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (123.28 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.45 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (100.94 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 82.33 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 85.02 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 76.85 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.32 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.22 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (148.33 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (141.61 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (105.91 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (152.85 ms)

Server Average (ms)
atlanta.voip.ms 88.7630
atlanta2.voip.ms 92.9233
chicago.voip.ms 49.6477
chicago2.voip.ms 59.9305
chicago3.voip.ms 59.3972
chicago4.voip.ms 49.9152
dallas.voip.ms 95.0790
denver.voip.ms 94.8077
denver2.voip.ms 85.2797
houston.voip.ms 103.1562
losangeles.voip.ms 65.4693
losangeles2.voip.ms 63.8347
newyork.voip.ms 118.0643
newyork2.voip.ms 120.9265
newyork3.voip.ms 121.2778
newyork4.voip.ms 110.8275
seattle.voip.ms 93.9993
seattle2.voip.ms 95.8267
seattle3.voip.ms 91.1035
tampa.voip.ms 123.6570
washington.voip.ms 98.4065
washington2.voip.ms 101.3774
montreal.voip.ms 81.5525
montreal2.voip.ms 85.9863
montreal3.voip.ms 76.4058
montreal4.voip.ms 96.7013
toronto2.voip.ms 103.7986
toronto3.voip.ms 143.3156
toronto4.voip.ms 136.0254
toronto.voip.ms 104.1012
london.voip.ms 157.3885

=== Powershell ===

# Usage: Copy and paste the following code into a powershell window
# To run it from a command prompt, save this file with extension ps1. Then run Powershell.exe -file "pathtothisscript.ps1"
#Get the list of servers into an array
$Servers =
@(“atlanta.voip.ms”,”atlanta2.voip.ms”,”chicago.voip.ms”,”chicago2.voip.ms”,”chicago3.voip.ms”,”chicago4.voip.ms”,”dallas.voip.ms”,
”denver.voip.ms”,”denver2.voip.ms”,”houston.voip.ms”,”london.voip.ms”,”losangeles.voip.ms”,”losangeles2.voip.ms”,”montreal.voip.ms”,
”montreal2.voip.ms”,”montreal3.voip.ms”,”montreal4.voip.ms”,”newyork.voip.ms”,” newyork2.voip.ms”,”newyork3.voip.ms”,”newyork4.voip.ms”,
”seattle.voip.ms”,”seattle2.voip.ms”,”seattle3.voip.ms”,”tampa.voip.ms”,”toronto.voip.ms”,”toronto2.voip.ms”,”toronto3.voip.ms”,
”toronto4.voip.ms”,”vancouver.voip.ms”,”vancouver2.voip.ms”,”washington.voip.ms”,”washington2.voip.ms”)
$k = 0 #Counting variable so we know what server number we are testing
#num of servers to test
$servercount = $servers.length
#Do the following code for each server in our array
For Each($server in $servers){
$k++ #Add one to the counting variable....we are on server #1...then server 2, then server 3 etc...
Write-Progress -Activity "Testing Server: ${server}" -status "Testing Server $k out of $servercount" -percentComplete ($k /
$servercount*100) #Update the progress bar
$i = 0 #Counting variable for number of times we tried to ping a given server
Do{
$pingsuccess = $false #assume a failure
$i++ #Add one to the counting variable.....1st try....2nd try....3rd try etc...
Try{
$currentping = (test-connection $server -count 1 -ErrorAction Stop).responsetime #Try to ping
$pingsuccess = $True #If success full, set success variable
}Catch { $pingsuccess = $false } #Catch the failure and set the success variable to false
}While($pingsuccess -eq $false -and $i -le 5) #Try everything between Do and While up to 5 times, or while $pingsuccess is not
true
If($pingsuccess -and ($currentping -lt $bestping -or (!($bestping)))){ #Compare the last ping test with the best known ping
test....if there is no known best ping test, assume this one is the best $bestping = $currentping #If this is the best ping...save it
$bestserver = $server #Save the best server
}
write-host "tested: $server at $currentping ms after $i attempts" #write the results of the test for this server
}
write-host "The server with the best ping is: $bestserver at $bestping ms" #write the end result

=== Linux Shell Script ===
Pings several voip.ms servers

#!/bin/sh
# Ping several servers and display Latency, Jitter and Packet Loss
#
# First, create a text file with all servers you want to ping - one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
myHF="voip_ping_hosts.txt"
# Sample file:
# toronto.voip.ms
# montreal.voip.ms
# seattle.voip.ms
# chicago.voip.ms
# newyork.voip.ms
#
echo "============================================"
printf "%-20s %7s %8s %6s\n" "VoIP Server" "Latency" "Jitter" "Loss"
echo "============================================"
cat ${myHF} |\
while read myLn
do
ping -c 3 -i 5 -q $myLn |\
awk '/^PING / {myH=$2}
/packet loss/ {myPL=$6}
/min\/avg\/max/ {
split($4,myS,"/")
printf( "%-20s %3.1f %1.3f %4s\n", myH, myS[2], myS[4], myPL)
}'
done
echo "============================================"

Output:

============================================
VoIP Server Latency Jitter Loss
============================================
toronto.voip.ms 68.3 0.439 0%
montreal.voip.ms 89.6 0.197 0%
seattle.voip.ms 71.2 0.387 0%
chicago.voip.ms 71.6 0.084 0%
newyork.voip.ms 79.1 0.411 0%
============================================

= Latency and its importance =

Latency is very important for Voip, this will determine the time that will take for the data package transmission to reach the destination. A high latency will lead to a delay and echoes in the communication.

Latency is measured in milliseconds (ms) For example: a latency of 150ms is barely noticeable, thus acceptable. Higher than that, quality starts to suffer. When it gets higher than 300 ms, it becomes unacceptable.

[[category:guides]]

Choosing Server

2017-03-22T13:07:31Z

Rod: /* Latency Testing Scripts (User Submitted) */

[[File:Serverlocation3.png]]

= Choosing a Server =

[http://www.voip.ms VoIP.ms] offers many different servers, but which one should you choose? One misconception is that you should pick the closest to your location, however this is not needed most of the time. For example, if you are in the USA, any of the US servers will provide a really good latency and service quality. The newest server within a city is indicated with the highest number attached to the name, as they are classified in ascending order. Also worth noting is that there is a network tool that will help you when deciding which server you want to use, generally named a "ping", which will provide you the latency between you and the server. Therefore the server which provides you less latency should be used.

=== IPs ===

*Amsterdam, NL ('''amsterdam.voip.ms''') 37.58.88.242
*Atlanta 1, GA ('''atlanta.voip.ms''') 75.127.65.130
*Atlanta 2, GA ('''atlanta2.voip.ms''') 72.9.246.170
*Chicago 1, IL ('''chicago.voip.ms''') 208.100.39.52
*Chicago 2, IL ('''chicago2.voip.ms''') 208.100.39.53
*Chicago 3, IL ('''chicago3.voip.ms''') 208.100.39.54
*Chicago 4, IL ('''chicago4.voip.ms''') 208.100.39.55
*Dallas, TX ('''dallas.voip.ms''') 158.85.149.162
*Dallas 2, TX ('''dallas2.voip.ms''') 158.85.149.163
*Denver 1, CO ('''denver.voip.ms''') 173.248.161.90
*Denver 2, CO ('''denver2.voip.ms''') 173.248.159.210
*Houston, TX ('''houston.voip.ms''') 173.193.85.18
*Houston 2, TX ('''houston2.voip.ms''') 173.193.85.19
*London, UK ('''london.voip.ms''') 159.8.157.212
*Los Angeles 1, CA ('''losangeles.voip.ms''') 96.44.149.186
*Los Angeles 2, CA ('''losangeles2.voip.ms''') 96.44.149.202
*Melbourne, AU ('''melbourne.voip.ms''') 168.1.73.84
*Montreal 1, QC ('''montreal.voip.ms''') 67.205.74.184
*Montreal 2, QC ('''montreal2.voip.ms''') 67.205.74.187
*Montreal 3, QC ('''montreal3.voip.ms''') 192.175.96.68
*Montreal 4, QC ('''montreal4.voip.ms''') 67.205.74.179
*Montreal 5, QC ('''montreal5.voip.ms''') 192.175.96.69
*Montreal 6, QC ('''montreal6.voip.ms''') 192.175.96.70
*Montreal 7, QC ('''montreal7.voip.ms''') 192.175.96.71
*Montreal 8, QC ('''montreal8.voip.ms''') 192.175.96.72
*New York 1, NY ('''newyork.voip.ms''') 74.63.41.218
*New York 2, NY ('''newyork2.voip.ms''') 107.6.67.236
*New York 3, NY ('''newyork3.voip.ms''') 107.6.67.237
*New York 4, NY ('''newyork4.voip.ms''') 107.6.67.238
*New York 5, NY ('''newyork5.voip.ms''') 23.29.136.28
*New York 6, NY ('''newyork6.voip.ms''') 23.29.136.29
*New York 7, NY ('''newyork7.voip.ms''') 23.29.136.38
*New York 8, NY ('''newyork8.voip.ms''') 23.29.136.40
*Paris, FR ('''paris.voip.ms''') 159.8.85.180
*San Jose, CA ('''sanjose.voip.ms''') 23.246.247.146
*San Jose 2, CA ('''sanjose2.voip.ms''') 23.246.247.147
*Seattle 1, WA ('''seattle.voip.ms''') 50.23.160.50
*Seattle 2, WA ('''seattle2.voip.ms''') 50.23.160.51
*Seattle 3, WA ('''seattle3.voip.ms''') 50.23.160.52
*Tampa, FL ('''tampa.voip.ms''') 162.254.144.173
*Tampa 2, FL ('''tampa2.voip.ms''') 162.254.144.176
*Toronto 1, ON ('''toronto.voip.ms''') 158.85.70.148
*Toronto 2, ON ('''toronto2.voip.ms''') 158.85.70.149
*Toronto 3, ON ('''toronto3.voip.ms''') 158.85.70.150
*Toronto 4, ON ('''toronto4.voip.ms''') 158.85.70.151
*Toronto 5, ON ('''toronto5.voip.ms''') 184.75.215.106
*Toronto 6, ON ('''toronto6.voip.ms''') 184.75.215.114
*Toronto 7, ON ('''toronto7.voip.ms''') 184.75.215.146
*Toronto 8, ON ('''toronto8.voip.ms''') 184.75.213.210
*Vancouver 1, BC ('''vancouver.voip.ms''') 162.213.157.82
*Vancouver 2, BC ('''vancouver2.voip.ms''') 162.213.157.117
*Washington 1, DC ('''washington.voip.ms''') 208.43.234.226
*Washington 2, DC ('''washington2.voip.ms''') 208.43.234.227

===Server Realms===

For IOS, Please click here [http://wiki.voip.ms/article/Server_Realms Server Realms] to get the Realm Name for the server you plan on using, this can differ from the Domain Name being used.

Please note that the following servers will not be available to select as a DID Point of Presence for newer accounts:
Atlanta 1, Chicago 1, New York 1, New York 4, Seattle 1, Montreal 1, Montreal 2, Montreal 3, Montreal 4, Toronto, Toronto 2, Toronto 3, Toronto 4.

= What is a Ping? =

Ping is a standard tool used to test network connections. It is mostly used to determine if a server or device can be reached across the network and the latency of the response(the time it takes to send a packet to the destination and for it to return to your computer).

Ping tools are part of Windows, Mac OS X and Linux as well as some routers.

== How does the ping work? ==

It sends request messages to a target network address or DNS names at periodic intervals and measures the time it takes for a response message to arrive and return(better known as latency).

==How to ping on a PC==

Pinging is a command which tells you if the connection between your computer and a particular domain is working correctly.

In Windows, select Start > Programs > Accessories > Command Prompt. This will give you a window like the one below.

Enter the word ping, followed by a space, then the domain name.(montreal.voip.ms) in this case domain is our server name.

If the results show a series of replies, the connection is working. The time shows you how fast the connection is. If you see a "timed out" error instead of a reply, there is a breakdown somewhere between your computer and the domain.

[[File:Ping.gif]]

==How to ping on a Mac Computer==

1- Click on Finder in the dock.

2- Click on Applications.

3- Click on Utilities.

4- Double-click on Network Utility. *

* In OS X Mavericks (10.9.x) this utility app changed location. Launch it from spotlight instead, either press "command"+"space bar" or click on spotlight directly (magnifying glass icon at top right of screen), type "network utility" and hit "return"

5- In the Network Utility window, click on the Ping tab

6- In the field under "Please enter the network address to ping," like montreal.voip.ms

''If pings results are not consistent, you may have an issue with Jitter. You can work on this issue by adjusting the "Network Jitter Level" setting on your VoIP device. Usually a ping of under 150 ms is recommended in order to have good quality. The latency time to the server is important, however there are also other factors that could affect the quality of the calls such as packet loss (VoIP communications are very sensitive to this), and the Jitter level of your Internet connection.''

The following is the output of running ping with the target losangeles.voip.ms.

#ping losangeles.voip.ms
Ping to losangeles.voip.ms [67.215.241.250] with 32 bytes de datos:
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
Response from 67.215.241.250: bytes=32 time=69ms TTL=52
Response from 67.215.241.250: bytes=32 time=68ms TTL=52
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
ping statistics from 67.215.241.250:
4 packets transmitted, 4 received, 0% packet lost. rtt min/avg/max/mdev = 67ms, 69ms, 67ms

Sample ping output in windows:
C:\Windows\system32>ping montreal.voip.ms

Pinging montreal.voip.ms [67.205.74.184] with 32 bytes of data:
Reply from 67.205.74.184: bytes=32 time=85ms TTL=49
Reply from 67.205.74.184: bytes=32 time=86ms TTL=49
Reply from 67.205.74.184: bytes=32 time=86ms TTL=49
Reply from 67.205.74.184: bytes=32 time=85ms TTL=49

Ping statistics for 67.205.74.184:
Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 85ms, Maximum = 86ms, Average = 85ms

=== Sample Linux Shell Script ===
Pings several voip.ms servers

#!/bin/sh
# Ping several servers and display Latency, Jitter and Packet Loss
#
# First, create a text file with all servers you want to ping - one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
myHF="voip_ping_hosts.txt"
# Sample file:
# toronto.voip.ms
# montreal.voip.ms
# seattle.voip.ms
# chicago.voip.ms
# newyork.voip.ms
#
echo "============================================"
printf "%-20s %7s %8s %6s\n" "VoIP Server" "Latency" "Jitter" "Loss"
echo "============================================"
cat ${myHF} |\
while read myLn
do
ping -c 3 -i 5 -q $myLn |\
awk '/^PING / {myH=$2}
/packet loss/ {myPL=$6}
/min\/avg\/max/ {
split($4,myS,"/")
printf( "%-20s %3.1f %1.3f %4s\n", myH, myS[2], myS[4], myPL)
}'
done
echo "============================================"

Output:

============================================
VoIP Server Latency Jitter Loss
============================================
toronto.voip.ms 68.3 0.439 0%
montreal.voip.ms 89.6 0.197 0%
seattle.voip.ms 71.2 0.387 0%
chicago.voip.ms 71.6 0.084 0%
newyork.voip.ms 79.1 0.411 0%
============================================

= Latency Testing Scripts (User Submitted) =

=== Bash Script To Handle The Mac Ping Output Format ===

<p>To make use of this script (1) save as a plain text file (2) set permissions on the file to executable (3) invoke script
e.g. Save script below using your favourite editor as pingVoipMS.sh (2) chmod u+x pingVoipMS.sh (3) ./pingVoipMS.sh
This is a bash 3.x script, so it also works in Linux, just change the ping packet loss field from 7 to 6 in the final loop below (or wherever the loss field is in your ping output format). Depending upon your distro curl might need to change to wget.
</p>
<pre>
#!/bin/bash
# Ping several servers and display Latency, Jitter and Packet Loss
# Usage: [-c <count>][-i <wait time>][-r test restricted servers][<server list file>]
#
# The optional text file should be formatted with one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
# If no args are supplied, this script will scrape a ping server list from voip.ms
#
USER_FILE=""
COUNT=3; INTERVAL=5; RESTRICTED=0
restrictedList=(atlanta.voip.ms chicago.voip.ms
montreal.voip.ms montreal2.voip.ms montreal3.voip.ms montreal4.voip.ms
newyork.voip.ms newyork4.voip.ms seattle.voip.ms
toronto.voip.ms toronto2.voip.ms toronto3.voip.ms toronto4.voip.ms)

# Handle any passed in script arguments
while getopts c:i:r parm
do
case $parm in
c)count_opt=$OPTARG;;
i)interval_opt=$OPTARG;;
r)RESTRICTED=1;;
*)echo -e "Invalid arg\nUsage:\t[ -c <count of ECHO_REQUESTs to Tx, default 3> ] \
\n\t[ -i <wait time (s) between datagrams, default 5> ] \
\n\t[ -r ] Include restricted servers in latency test \
\n\t[FILE <ping server list> ]";exit 1;;
esac
done

# Test if an option was specified and whether it's a +ve non-zero integer
[[ -n $count_opt && ($count_opt =~ ^[[:digit:]]+$) && $count_opt -gt 0 ]] && COUNT=$count_opt
[[ -n $interval_opt && ($interval_opt =~ ^[[:digit:]]+$) && $interval_opt -gt 0 ]] && INTERVAL=$interval_opt

shift $(($OPTIND -1))

# Validate supplied file (server list)
[[ -n $1 && !(-f $1 && -r $1) ]] && { echo "\"$1\" file does not exist or is not readable"; exit 1; }
[[ -n $1 && -f $1 && -r $1 ]] && USER_FILE="$1"

if [[ -n $USER_FILE ]]
then
# Bash 3.x in macOS does not support readarray, need to do cumbersome array loops instead
while IFS= read -r line; do
serverList+=( "$line" )
done < <(grep -e '[[:alpha:]]*[[:alnum:]]\.voip\.ms' $USER_FILE | grep -v '^\s*#' | awk NF)
else
# N.B. The script looks for the html boldface tags <b> </b> inside a bracket
# If the website alters and the parse fails, manually create the list and
# supply as a script arg (or perhaps update the parsing to work again :)
while IFS= read -r line; do
serverList+=( "$line" )
done < <(curl --silent http://wiki.voip.ms/article/Choosing_Server | \
grep '(<b>[[:alpha:]]*[[:alnum:]]\.voip\.ms</b>)' | \
tr "<>" " " | awk '{print $(NF-3)}' )
fi

# Newer voip.ms clients can't register onto these over-subscribed servers
# Don't test the restricted list unless explicitly asked (with the -r cmd line option)
if [[ $RESTRICTED -eq 0 ]]
then
for server in ${restrictedList[@]}
do
ix=$(printf "%s\n" "${serverList[@]}" | grep -n "^${server}" | cut -d ":" -f1)

while IFS= read -r idx; do
keys+=($idx)
done < <([[ $ix -gt 0 ]] && echo $((ix-1)))
done
for ((i=${#keys[@]} - 1; i >= 0; i--)); do unset serverList[keys[i]]; done
fi

if [[ ${#serverList[@]} -eq 0 ]]
then
echo "No unrestricted Voip.ms servers could be found, please supply a server list"
exit 1
fi

runTime=$(( $COUNT * $INTERVAL * ${#serverList[@]} ))

echo "PING will send $COUNT packet(s) with a wait of $INTERVAL sec(s) between each packet"
echo "Change the PING options by invoking this script with -c and/or -i, default \"-c 3 -i 5\""
echo "Over $((${#serverList[@]})) server(s) the estimated script Run Time will be $runTime seconds"
echo "================================================================"
printf "%-20s %-18s %7s %8s %6s %s\n" "VoIP Server" "IP Address" "Latency" "Jitter" "Loss" "Countdown"
echo "================================================================ (seconds)"

for myLn in ${serverList[@]}
do
while IFS=$'\n' read -r line; do
pingList+=( "$line" )
printf "%-64s %5d %2d/%-2d\n" "$line" "$((runTime - $COUNT * $INTERVAL * ${#pingList[@]}))" "${#pingList[@]}" "${#serverList[@]}"
done < <( ping -c $COUNT -i $INTERVAL -q $myLn | awk '\
/^PING / {myH=$2}
/^PING / {
IP = substr($3,2,15)
split(IP,myIP,")") }
/packet loss/ {myPL=$7}
/min\/avg\/max/ {
split($4,myS,"/")
printf("%-20s %-18s %7.3f %8.3f %6s\n",
myH, myIP[1], myS[2], myS[4], myPL ) }
' )
done

echo "================================================================"
echo -e "\nMost appropriate server listed in order of best latency\n"
echo "================================================================"
printf "%-20s %-18s %7s %8s %6s\n" "VoIP Server" "IP Address" "Latency" "Jitter" "Loss"
echo "================================================================"
printf "%s\n" "${pingList[@]}" | LC_ALL=C sort -n -k 3,3 -k 5,5 -k 4,4 | awk '{printf("%s $%2d$\n",$0, NR)}'
echo "================================================================"
</pre>

=== Perl Script ===
Pings list of voip.ms servers round robin with optional output csv file.

# usage ping_voip.ms.pl <number of times> <seconds in between> <output.csv>
use Net::Ping;
use Time::HiRes;
use strict;

# input list
my @hosts = qw(
atlanta.voip.ms
atlanta2.voip.ms
chicago.voip.ms
chicago2.voip.ms
chicago3.voip.ms
chicago4.voip.ms
dallas.voip.ms
denver.voip.ms
denver2.voip.ms
houston.voip.ms
losangeles.voip.ms
losangeles2.voip.ms
newyork.voip.ms
newyork2.voip.ms
newyork3.voip.ms
newyork4.voip.ms
seattle.voip.ms
seattle2.voip.ms
seattle3.voip.ms
tampa.voip.ms
washington.voip.ms
washington2.voip.ms
montreal.voip.ms
montreal2.voip.ms
montreal3.voip.ms
montreal4.voip.ms
toronto2.voip.ms
toronto3.voip.ms
toronto4.voip.ms
toronto.voip.ms
london.voip.ms
);

$| = 1; #autoflush
# High precision syntax (requires Time::HiRes)
my $p = Net::Ping->new("icmp",1);
$p->hires();
my $max_name_length = (reverse sort { $a <=> $b } map { length($_) } @hosts)[0];
my $count = 4; # number of times to ping
my $interval = 5; # seconds between ping rounds
my $output_file = "";
my @data;

# check for arguments
my $num_args = @ARGV;
if ($num_args >= 1) {$count = $ARGV[0];}
if ($num_args >= 2) {$interval = $ARGV[1];}
if ($num_args >= 3) {$output_file = $ARGV[2];}

# check argument validity
$0 =~ /^.*\\(.*)$/;
my $script = $1;
if ($count !~ /^\d+$/ or $interval !~ /^\d+$/) {die "Usage: $script <number of rounds> <seconds between rounds> <output.csv>\n";}
if (length($output_file) > 0 and $output_file !~ /\.csv$/) {$output_file .= ".csv";}

# main loop
for my $i (1..$count)
{
sleep $interval unless $i == 1;
print "Round $i\n";
my $host_num=0;
foreach my $host (@hosts)
{
(my $ret, my $duration, my $ip) = $p->ping($host);
$ip =~ /(\d+)\.(\d+)\.(\d+)\.(\d+)/;
if ($ret)
{
printf("%*s [ip: %3s.%3s.%3s.%3s] is alive (%6.2f ms)\n", $max_name_length, $host, $1, $2, $3, $4, $duration*1000);
$data[$host_num][$i]=$duration*1000;
}
else
{
printf("%*s [ip: %3s.%3s.%3s.%3s] is dead\n", $max_name_length, $host, $1, $2, $3, $4);
}
$host_num++;
}
print "\n";
}
$p->close();

# if output file name given
if (length($output_file)>0)
{
# print output to file
open FILE, ">$output_file" or die "$!\n";

# print column headers
print FILE "Server\\Round";
for my $i (1..$count)
{
print FILE ", $i";
}
print FILE ", Average\n";

# print data
my $i = 0;
foreach my $host (@hosts)
{
print FILE "$host";
my $sum = 0;
for my $j (1..$count)
{
$sum += $data[$i][$j];
printf FILE ", %8.4f",$data[$i][$j];
}
printf FILE ", %8.4f\n",$sum/$count;
$i++;
}

close FILE;
print "Data written to $output_file\n";
}

# print summary to screen
my $i = 0;
printf("%-*s Average (ms)\n", $max_name_length, "Server");
foreach my $host (@hosts)
{
my $sum = 0;
for my $j (1..$count)
{
$sum += $data[$i][$j];
}
printf("%-*s %8.4f\n", $max_name_length+1, $host, $sum/$count);
$i++;
}

Output:
Round 1
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.97 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.99 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.70 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.76 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.53 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.73 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 94.99 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.05 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.13 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (102.87 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 64.92 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 63.41 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (131.75 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (120.64 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (120.49 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (111.43 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.25 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.86 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 90.85 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (123.29 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.71 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.19 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 81.82 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 86.13 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 77.09 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.18 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.70 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (131.27 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (125.13 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.26 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (152.77 ms)

Round 2
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.14 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.86 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 50.03 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.44 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.33 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.22 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 95.58 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.94 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.29 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (102.73 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.59 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 64.27 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (112.74 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (121.22 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (121.34 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.75 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.06 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.33 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 91.58 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (122.94 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.28 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.40 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 81.91 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 85.64 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 75.15 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.79 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.10 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (150.85 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.40 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.45 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (170.79 ms)

Round 3
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.76 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.86 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.65 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.01 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.05 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.53 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 95.82 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.02 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.60 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (103.35 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.79 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 64.05 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (113.01 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (121.41 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (122.23 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.62 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 93.65 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.19 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 90.75 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (125.12 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.19 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.98 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 80.16 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 87.16 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 76.54 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 97.51 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.18 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (142.81 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.95 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.78 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (153.14 ms)

Round 4
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 89.19 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.98 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.21 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.50 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.68 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.18 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 93.93 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.22 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.10 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (103.67 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.58 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 63.60 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (114.76 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (120.44 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (121.05 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.51 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.04 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 96.92 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 91.23 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (123.28 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.45 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (100.94 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 82.33 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 85.02 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 76.85 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.32 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.22 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (148.33 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (141.61 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (105.91 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (152.85 ms)

Server Average (ms)
atlanta.voip.ms 88.7630
atlanta2.voip.ms 92.9233
chicago.voip.ms 49.6477
chicago2.voip.ms 59.9305
chicago3.voip.ms 59.3972
chicago4.voip.ms 49.9152
dallas.voip.ms 95.0790
denver.voip.ms 94.8077
denver2.voip.ms 85.2797
houston.voip.ms 103.1562
losangeles.voip.ms 65.4693
losangeles2.voip.ms 63.8347
newyork.voip.ms 118.0643
newyork2.voip.ms 120.9265
newyork3.voip.ms 121.2778
newyork4.voip.ms 110.8275
seattle.voip.ms 93.9993
seattle2.voip.ms 95.8267
seattle3.voip.ms 91.1035
tampa.voip.ms 123.6570
washington.voip.ms 98.4065
washington2.voip.ms 101.3774
montreal.voip.ms 81.5525
montreal2.voip.ms 85.9863
montreal3.voip.ms 76.4058
montreal4.voip.ms 96.7013
toronto2.voip.ms 103.7986
toronto3.voip.ms 143.3156
toronto4.voip.ms 136.0254
toronto.voip.ms 104.1012
london.voip.ms 157.3885

=== Powershell ===

# Usage: Copy and paste the following code into a powershell window
# To run it from a command prompt, save this file with extension ps1. Then run Powershell.exe -file "pathtothisscript.ps1"
#Get the list of servers into an array
$Servers =
@(“atlanta.voip.ms”,”atlanta2.voip.ms”,”chicago.voip.ms”,”chicago2.voip.ms”,”chicago3.voip.ms”,”chicago4.voip.ms”,”dallas.voip.ms”,
”denver.voip.ms”,”denver2.voip.ms”,”houston.voip.ms”,”london.voip.ms”,”losangeles.voip.ms”,”losangeles2.voip.ms”,”montreal.voip.ms”,
”montreal2.voip.ms”,”montreal3.voip.ms”,”montreal4.voip.ms”,”newyork.voip.ms”,” newyork2.voip.ms”,”newyork3.voip.ms”,”newyork4.voip.ms”,
”seattle.voip.ms”,”seattle2.voip.ms”,”seattle3.voip.ms”,”tampa.voip.ms”,”toronto.voip.ms”,”toronto2.voip.ms”,”toronto3.voip.ms”,
”toronto4.voip.ms”,”vancouver.voip.ms”,”vancouver2.voip.ms”,”washington.voip.ms”,”washington2.voip.ms”)
$k = 0 #Counting variable so we know what server number we are testing
#num of servers to test
$servercount = $servers.length
#Do the following code for each server in our array
For Each($server in $servers){
$k++ #Add one to the counting variable....we are on server #1...then server 2, then server 3 etc...
Write-Progress -Activity "Testing Server: ${server}" -status "Testing Server $k out of $servercount" -percentComplete ($k /
$servercount*100) #Update the progress bar
$i = 0 #Counting variable for number of times we tried to ping a given server
Do{
$pingsuccess = $false #assume a failure
$i++ #Add one to the counting variable.....1st try....2nd try....3rd try etc...
Try{
$currentping = (test-connection $server -count 1 -ErrorAction Stop).responsetime #Try to ping
$pingsuccess = $True #If success full, set success variable
}Catch { $pingsuccess = $false } #Catch the failure and set the success variable to false
}While($pingsuccess -eq $false -and $i -le 5) #Try everything between Do and While up to 5 times, or while $pingsuccess is not
true
If($pingsuccess -and ($currentping -lt $bestping -or (!($bestping)))){ #Compare the last ping test with the best known ping
test....if there is no known best ping test, assume this one is the best $bestping = $currentping #If this is the best ping...save it
$bestserver = $server #Save the best server
}
write-host "tested: $server at $currentping ms after $i attempts" #write the results of the test for this server
}
write-host "The server with the best ping is: $bestserver at $bestping ms" #write the end result

= Latency and its importance =

Latency is very important for Voip, this will determine the time that will take for the data package transmission to reach the destination. A high latency will lead to a delay and echoes in the communication.

Latency is measured in milliseconds (ms) For example: a latency of 150ms is barely noticeable, thus acceptable. Higher than that, quality starts to suffer. When it gets higher than 300 ms, it becomes unacceptable.

[[category:guides]]

Choosing Server

2017-03-22T13:06:58Z

Rod:

[[File:Serverlocation3.png]]

= Choosing a Server =

[http://www.voip.ms VoIP.ms] offers many different servers, but which one should you choose? One misconception is that you should pick the closest to your location, however this is not needed most of the time. For example, if you are in the USA, any of the US servers will provide a really good latency and service quality. The newest server within a city is indicated with the highest number attached to the name, as they are classified in ascending order. Also worth noting is that there is a network tool that will help you when deciding which server you want to use, generally named a "ping", which will provide you the latency between you and the server. Therefore the server which provides you less latency should be used.

=== IPs ===

*Amsterdam, NL ('''amsterdam.voip.ms''') 37.58.88.242
*Atlanta 1, GA ('''atlanta.voip.ms''') 75.127.65.130
*Atlanta 2, GA ('''atlanta2.voip.ms''') 72.9.246.170
*Chicago 1, IL ('''chicago.voip.ms''') 208.100.39.52
*Chicago 2, IL ('''chicago2.voip.ms''') 208.100.39.53
*Chicago 3, IL ('''chicago3.voip.ms''') 208.100.39.54
*Chicago 4, IL ('''chicago4.voip.ms''') 208.100.39.55
*Dallas, TX ('''dallas.voip.ms''') 158.85.149.162
*Dallas 2, TX ('''dallas2.voip.ms''') 158.85.149.163
*Denver 1, CO ('''denver.voip.ms''') 173.248.161.90
*Denver 2, CO ('''denver2.voip.ms''') 173.248.159.210
*Houston, TX ('''houston.voip.ms''') 173.193.85.18
*Houston 2, TX ('''houston2.voip.ms''') 173.193.85.19
*London, UK ('''london.voip.ms''') 159.8.157.212
*Los Angeles 1, CA ('''losangeles.voip.ms''') 96.44.149.186
*Los Angeles 2, CA ('''losangeles2.voip.ms''') 96.44.149.202
*Melbourne, AU ('''melbourne.voip.ms''') 168.1.73.84
*Montreal 1, QC ('''montreal.voip.ms''') 67.205.74.184
*Montreal 2, QC ('''montreal2.voip.ms''') 67.205.74.187
*Montreal 3, QC ('''montreal3.voip.ms''') 192.175.96.68
*Montreal 4, QC ('''montreal4.voip.ms''') 67.205.74.179
*Montreal 5, QC ('''montreal5.voip.ms''') 192.175.96.69
*Montreal 6, QC ('''montreal6.voip.ms''') 192.175.96.70
*Montreal 7, QC ('''montreal7.voip.ms''') 192.175.96.71
*Montreal 8, QC ('''montreal8.voip.ms''') 192.175.96.72
*New York 1, NY ('''newyork.voip.ms''') 74.63.41.218
*New York 2, NY ('''newyork2.voip.ms''') 107.6.67.236
*New York 3, NY ('''newyork3.voip.ms''') 107.6.67.237
*New York 4, NY ('''newyork4.voip.ms''') 107.6.67.238
*New York 5, NY ('''newyork5.voip.ms''') 23.29.136.28
*New York 6, NY ('''newyork6.voip.ms''') 23.29.136.29
*New York 7, NY ('''newyork7.voip.ms''') 23.29.136.38
*New York 8, NY ('''newyork8.voip.ms''') 23.29.136.40
*Paris, FR ('''paris.voip.ms''') 159.8.85.180
*San Jose, CA ('''sanjose.voip.ms''') 23.246.247.146
*San Jose 2, CA ('''sanjose2.voip.ms''') 23.246.247.147
*Seattle 1, WA ('''seattle.voip.ms''') 50.23.160.50
*Seattle 2, WA ('''seattle2.voip.ms''') 50.23.160.51
*Seattle 3, WA ('''seattle3.voip.ms''') 50.23.160.52
*Tampa, FL ('''tampa.voip.ms''') 162.254.144.173
*Tampa 2, FL ('''tampa2.voip.ms''') 162.254.144.176
*Toronto 1, ON ('''toronto.voip.ms''') 158.85.70.148
*Toronto 2, ON ('''toronto2.voip.ms''') 158.85.70.149
*Toronto 3, ON ('''toronto3.voip.ms''') 158.85.70.150
*Toronto 4, ON ('''toronto4.voip.ms''') 158.85.70.151
*Toronto 5, ON ('''toronto5.voip.ms''') 184.75.215.106
*Toronto 6, ON ('''toronto6.voip.ms''') 184.75.215.114
*Toronto 7, ON ('''toronto7.voip.ms''') 184.75.215.146
*Toronto 8, ON ('''toronto8.voip.ms''') 184.75.213.210
*Vancouver 1, BC ('''vancouver.voip.ms''') 162.213.157.82
*Vancouver 2, BC ('''vancouver2.voip.ms''') 162.213.157.117
*Washington 1, DC ('''washington.voip.ms''') 208.43.234.226
*Washington 2, DC ('''washington2.voip.ms''') 208.43.234.227

===Server Realms===

For IOS, Please click here [http://wiki.voip.ms/article/Server_Realms Server Realms] to get the Realm Name for the server you plan on using, this can differ from the Domain Name being used.

Please note that the following servers will not be available to select as a DID Point of Presence for newer accounts:
Atlanta 1, Chicago 1, New York 1, New York 4, Seattle 1, Montreal 1, Montreal 2, Montreal 3, Montreal 4, Toronto, Toronto 2, Toronto 3, Toronto 4.

= What is a Ping? =

Ping is a standard tool used to test network connections. It is mostly used to determine if a server or device can be reached across the network and the latency of the response(the time it takes to send a packet to the destination and for it to return to your computer).

Ping tools are part of Windows, Mac OS X and Linux as well as some routers.

== How does the ping work? ==

It sends request messages to a target network address or DNS names at periodic intervals and measures the time it takes for a response message to arrive and return(better known as latency).

==How to ping on a PC==

Pinging is a command which tells you if the connection between your computer and a particular domain is working correctly.

In Windows, select Start > Programs > Accessories > Command Prompt. This will give you a window like the one below.

Enter the word ping, followed by a space, then the domain name.(montreal.voip.ms) in this case domain is our server name.

If the results show a series of replies, the connection is working. The time shows you how fast the connection is. If you see a "timed out" error instead of a reply, there is a breakdown somewhere between your computer and the domain.

[[File:Ping.gif]]

==How to ping on a Mac Computer==

1- Click on Finder in the dock.

2- Click on Applications.

3- Click on Utilities.

4- Double-click on Network Utility. *

* In OS X Mavericks (10.9.x) this utility app changed location. Launch it from spotlight instead, either press "command"+"space bar" or click on spotlight directly (magnifying glass icon at top right of screen), type "network utility" and hit "return"

5- In the Network Utility window, click on the Ping tab

6- In the field under "Please enter the network address to ping," like montreal.voip.ms

''If pings results are not consistent, you may have an issue with Jitter. You can work on this issue by adjusting the "Network Jitter Level" setting on your VoIP device. Usually a ping of under 150 ms is recommended in order to have good quality. The latency time to the server is important, however there are also other factors that could affect the quality of the calls such as packet loss (VoIP communications are very sensitive to this), and the Jitter level of your Internet connection.''

The following is the output of running ping with the target losangeles.voip.ms.

#ping losangeles.voip.ms
Ping to losangeles.voip.ms [67.215.241.250] with 32 bytes de datos:
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
Response from 67.215.241.250: bytes=32 time=69ms TTL=52
Response from 67.215.241.250: bytes=32 time=68ms TTL=52
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
ping statistics from 67.215.241.250:
4 packets transmitted, 4 received, 0% packet lost. rtt min/avg/max/mdev = 67ms, 69ms, 67ms

Sample ping output in windows:
C:\Windows\system32>ping montreal.voip.ms

Pinging montreal.voip.ms [67.205.74.184] with 32 bytes of data:
Reply from 67.205.74.184: bytes=32 time=85ms TTL=49
Reply from 67.205.74.184: bytes=32 time=86ms TTL=49
Reply from 67.205.74.184: bytes=32 time=86ms TTL=49
Reply from 67.205.74.184: bytes=32 time=85ms TTL=49

Ping statistics for 67.205.74.184:
Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 85ms, Maximum = 86ms, Average = 85ms

=== Sample Linux Shell Script ===
Pings several voip.ms servers

#!/bin/sh
# Ping several servers and display Latency, Jitter and Packet Loss
#
# First, create a text file with all servers you want to ping - one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
myHF="voip_ping_hosts.txt"
# Sample file:
# toronto.voip.ms
# montreal.voip.ms
# seattle.voip.ms
# chicago.voip.ms
# newyork.voip.ms
#
echo "============================================"
printf "%-20s %7s %8s %6s\n" "VoIP Server" "Latency" "Jitter" "Loss"
echo "============================================"
cat ${myHF} |\
while read myLn
do
ping -c 3 -i 5 -q $myLn |\
awk '/^PING / {myH=$2}
/packet loss/ {myPL=$6}
/min\/avg\/max/ {
split($4,myS,"/")
printf( "%-20s %3.1f %1.3f %4s\n", myH, myS[2], myS[4], myPL)
}'
done
echo "============================================"

Output:

============================================
VoIP Server Latency Jitter Loss
============================================
toronto.voip.ms 68.3 0.439 0%
montreal.voip.ms 89.6 0.197 0%
seattle.voip.ms 71.2 0.387 0%
chicago.voip.ms 71.6 0.084 0%
newyork.voip.ms 79.1 0.411 0%
============================================

== Latency Testing Scripts (User Submitted) ==

=== Bash Script To Handle The Mac Ping Output Format ===

<p>To make use of this script (1) save as a plain text file (2) set permissions on the file to executable (3) invoke script
e.g. Save script below using your favourite editor as pingVoipMS.sh (2) chmod u+x pingVoipMS.sh (3) ./pingVoipMS.sh
This is a bash 3.x script, so it also works in Linux, just change the ping packet loss field from 7 to 6 in the final loop below (or wherever the loss field is in your ping output format). Depending upon your distro curl might need to change to wget.
</p>
<pre>
#!/bin/bash
# Ping several servers and display Latency, Jitter and Packet Loss
# Usage: [-c <count>][-i <wait time>][-r test restricted servers][<server list file>]
#
# The optional text file should be formatted with one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
# If no args are supplied, this script will scrape a ping server list from voip.ms
#
USER_FILE=""
COUNT=3; INTERVAL=5; RESTRICTED=0
restrictedList=(atlanta.voip.ms chicago.voip.ms
montreal.voip.ms montreal2.voip.ms montreal3.voip.ms montreal4.voip.ms
newyork.voip.ms newyork4.voip.ms seattle.voip.ms
toronto.voip.ms toronto2.voip.ms toronto3.voip.ms toronto4.voip.ms)

# Handle any passed in script arguments
while getopts c:i:r parm
do
case $parm in
c)count_opt=$OPTARG;;
i)interval_opt=$OPTARG;;
r)RESTRICTED=1;;
*)echo -e "Invalid arg\nUsage:\t[ -c <count of ECHO_REQUESTs to Tx, default 3> ] \
\n\t[ -i <wait time (s) between datagrams, default 5> ] \
\n\t[ -r ] Include restricted servers in latency test \
\n\t[FILE <ping server list> ]";exit 1;;
esac
done

# Test if an option was specified and whether it's a +ve non-zero integer
[[ -n $count_opt && ($count_opt =~ ^[[:digit:]]+$) && $count_opt -gt 0 ]] && COUNT=$count_opt
[[ -n $interval_opt && ($interval_opt =~ ^[[:digit:]]+$) && $interval_opt -gt 0 ]] && INTERVAL=$interval_opt

shift $(($OPTIND -1))

# Validate supplied file (server list)
[[ -n $1 && !(-f $1 && -r $1) ]] && { echo "\"$1\" file does not exist or is not readable"; exit 1; }
[[ -n $1 && -f $1 && -r $1 ]] && USER_FILE="$1"

if [[ -n $USER_FILE ]]
then
# Bash 3.x in macOS does not support readarray, need to do cumbersome array loops instead
while IFS= read -r line; do
serverList+=( "$line" )
done < <(grep -e '[[:alpha:]]*[[:alnum:]]\.voip\.ms' $USER_FILE | grep -v '^\s*#' | awk NF)
else
# N.B. The script looks for the html boldface tags <b> </b> inside a bracket
# If the website alters and the parse fails, manually create the list and
# supply as a script arg (or perhaps update the parsing to work again :)
while IFS= read -r line; do
serverList+=( "$line" )
done < <(curl --silent http://wiki.voip.ms/article/Choosing_Server | \
grep '(<b>[[:alpha:]]*[[:alnum:]]\.voip\.ms</b>)' | \
tr "<>" " " | awk '{print $(NF-3)}' )
fi

# Newer voip.ms clients can't register onto these over-subscribed servers
# Don't test the restricted list unless explicitly asked (with the -r cmd line option)
if [[ $RESTRICTED -eq 0 ]]
then
for server in ${restrictedList[@]}
do
ix=$(printf "%s\n" "${serverList[@]}" | grep -n "^${server}" | cut -d ":" -f1)

while IFS= read -r idx; do
keys+=($idx)
done < <([[ $ix -gt 0 ]] && echo $((ix-1)))
done
for ((i=${#keys[@]} - 1; i >= 0; i--)); do unset serverList[keys[i]]; done
fi

if [[ ${#serverList[@]} -eq 0 ]]
then
echo "No unrestricted Voip.ms servers could be found, please supply a server list"
exit 1
fi

runTime=$(( $COUNT * $INTERVAL * ${#serverList[@]} ))

echo "PING will send $COUNT packet(s) with a wait of $INTERVAL sec(s) between each packet"
echo "Change the PING options by invoking this script with -c and/or -i, default \"-c 3 -i 5\""
echo "Over $((${#serverList[@]})) server(s) the estimated script Run Time will be $runTime seconds"
echo "================================================================"
printf "%-20s %-18s %7s %8s %6s %s\n" "VoIP Server" "IP Address" "Latency" "Jitter" "Loss" "Countdown"
echo "================================================================ (seconds)"

for myLn in ${serverList[@]}
do
while IFS=$'\n' read -r line; do
pingList+=( "$line" )
printf "%-64s %5d %2d/%-2d\n" "$line" "$((runTime - $COUNT * $INTERVAL * ${#pingList[@]}))" "${#pingList[@]}" "${#serverList[@]}"
done < <( ping -c $COUNT -i $INTERVAL -q $myLn | awk '\
/^PING / {myH=$2}
/^PING / {
IP = substr($3,2,15)
split(IP,myIP,")") }
/packet loss/ {myPL=$7}
/min\/avg\/max/ {
split($4,myS,"/")
printf("%-20s %-18s %7.3f %8.3f %6s\n",
myH, myIP[1], myS[2], myS[4], myPL ) }
' )
done

echo "================================================================"
echo -e "\nMost appropriate server listed in order of best latency\n"
echo "================================================================"
printf "%-20s %-18s %7s %8s %6s\n" "VoIP Server" "IP Address" "Latency" "Jitter" "Loss"
echo "================================================================"
printf "%s\n" "${pingList[@]}" | LC_ALL=C sort -n -k 3,3 -k 5,5 -k 4,4 | awk '{printf("%s $%2d$\n",$0, NR)}'
echo "================================================================"
</pre>

=== Perl Script ===
Pings list of voip.ms servers round robin with optional output csv file.

# usage ping_voip.ms.pl <number of times> <seconds in between> <output.csv>
use Net::Ping;
use Time::HiRes;
use strict;

# input list
my @hosts = qw(
atlanta.voip.ms
atlanta2.voip.ms
chicago.voip.ms
chicago2.voip.ms
chicago3.voip.ms
chicago4.voip.ms
dallas.voip.ms
denver.voip.ms
denver2.voip.ms
houston.voip.ms
losangeles.voip.ms
losangeles2.voip.ms
newyork.voip.ms
newyork2.voip.ms
newyork3.voip.ms
newyork4.voip.ms
seattle.voip.ms
seattle2.voip.ms
seattle3.voip.ms
tampa.voip.ms
washington.voip.ms
washington2.voip.ms
montreal.voip.ms
montreal2.voip.ms
montreal3.voip.ms
montreal4.voip.ms
toronto2.voip.ms
toronto3.voip.ms
toronto4.voip.ms
toronto.voip.ms
london.voip.ms
);

$| = 1; #autoflush
# High precision syntax (requires Time::HiRes)
my $p = Net::Ping->new("icmp",1);
$p->hires();
my $max_name_length = (reverse sort { $a <=> $b } map { length($_) } @hosts)[0];
my $count = 4; # number of times to ping
my $interval = 5; # seconds between ping rounds
my $output_file = "";
my @data;

# check for arguments
my $num_args = @ARGV;
if ($num_args >= 1) {$count = $ARGV[0];}
if ($num_args >= 2) {$interval = $ARGV[1];}
if ($num_args >= 3) {$output_file = $ARGV[2];}

# check argument validity
$0 =~ /^.*\\(.*)$/;
my $script = $1;
if ($count !~ /^\d+$/ or $interval !~ /^\d+$/) {die "Usage: $script <number of rounds> <seconds between rounds> <output.csv>\n";}
if (length($output_file) > 0 and $output_file !~ /\.csv$/) {$output_file .= ".csv";}

# main loop
for my $i (1..$count)
{
sleep $interval unless $i == 1;
print "Round $i\n";
my $host_num=0;
foreach my $host (@hosts)
{
(my $ret, my $duration, my $ip) = $p->ping($host);
$ip =~ /(\d+)\.(\d+)\.(\d+)\.(\d+)/;
if ($ret)
{
printf("%*s [ip: %3s.%3s.%3s.%3s] is alive (%6.2f ms)\n", $max_name_length, $host, $1, $2, $3, $4, $duration*1000);
$data[$host_num][$i]=$duration*1000;
}
else
{
printf("%*s [ip: %3s.%3s.%3s.%3s] is dead\n", $max_name_length, $host, $1, $2, $3, $4);
}
$host_num++;
}
print "\n";
}
$p->close();

# if output file name given
if (length($output_file)>0)
{
# print output to file
open FILE, ">$output_file" or die "$!\n";

# print column headers
print FILE "Server\\Round";
for my $i (1..$count)
{
print FILE ", $i";
}
print FILE ", Average\n";

# print data
my $i = 0;
foreach my $host (@hosts)
{
print FILE "$host";
my $sum = 0;
for my $j (1..$count)
{
$sum += $data[$i][$j];
printf FILE ", %8.4f",$data[$i][$j];
}
printf FILE ", %8.4f\n",$sum/$count;
$i++;
}

close FILE;
print "Data written to $output_file\n";
}

# print summary to screen
my $i = 0;
printf("%-*s Average (ms)\n", $max_name_length, "Server");
foreach my $host (@hosts)
{
my $sum = 0;
for my $j (1..$count)
{
$sum += $data[$i][$j];
}
printf("%-*s %8.4f\n", $max_name_length+1, $host, $sum/$count);
$i++;
}

Output:
Round 1
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.97 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.99 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.70 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.76 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.53 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.73 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 94.99 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.05 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.13 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (102.87 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 64.92 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 63.41 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (131.75 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (120.64 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (120.49 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (111.43 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.25 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.86 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 90.85 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (123.29 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.71 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.19 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 81.82 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 86.13 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 77.09 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.18 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.70 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (131.27 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (125.13 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.26 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (152.77 ms)

Round 2
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.14 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.86 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 50.03 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.44 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.33 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.22 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 95.58 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.94 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.29 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (102.73 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.59 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 64.27 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (112.74 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (121.22 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (121.34 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.75 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.06 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.33 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 91.58 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (122.94 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.28 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.40 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 81.91 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 85.64 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 75.15 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.79 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.10 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (150.85 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.40 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.45 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (170.79 ms)

Round 3
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.76 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.86 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.65 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.01 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.05 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.53 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 95.82 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.02 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.60 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (103.35 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.79 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 64.05 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (113.01 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (121.41 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (122.23 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.62 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 93.65 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.19 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 90.75 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (125.12 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.19 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.98 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 80.16 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 87.16 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 76.54 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 97.51 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.18 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (142.81 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.95 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.78 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (153.14 ms)

Round 4
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 89.19 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.98 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.21 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.50 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.68 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.18 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 93.93 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.22 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.10 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (103.67 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.58 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 63.60 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (114.76 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (120.44 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (121.05 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.51 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.04 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 96.92 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 91.23 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (123.28 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.45 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (100.94 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 82.33 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 85.02 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 76.85 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.32 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.22 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (148.33 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (141.61 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (105.91 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (152.85 ms)

Server Average (ms)
atlanta.voip.ms 88.7630
atlanta2.voip.ms 92.9233
chicago.voip.ms 49.6477
chicago2.voip.ms 59.9305
chicago3.voip.ms 59.3972
chicago4.voip.ms 49.9152
dallas.voip.ms 95.0790
denver.voip.ms 94.8077
denver2.voip.ms 85.2797
houston.voip.ms 103.1562
losangeles.voip.ms 65.4693
losangeles2.voip.ms 63.8347
newyork.voip.ms 118.0643
newyork2.voip.ms 120.9265
newyork3.voip.ms 121.2778
newyork4.voip.ms 110.8275
seattle.voip.ms 93.9993
seattle2.voip.ms 95.8267
seattle3.voip.ms 91.1035
tampa.voip.ms 123.6570
washington.voip.ms 98.4065
washington2.voip.ms 101.3774
montreal.voip.ms 81.5525
montreal2.voip.ms 85.9863
montreal3.voip.ms 76.4058
montreal4.voip.ms 96.7013
toronto2.voip.ms 103.7986
toronto3.voip.ms 143.3156
toronto4.voip.ms 136.0254
toronto.voip.ms 104.1012
london.voip.ms 157.3885

=== Powershell ===

# Usage: Copy and paste the following code into a powershell window
# To run it from a command prompt, save this file with extension ps1. Then run Powershell.exe -file "pathtothisscript.ps1"
#Get the list of servers into an array
$Servers =
@(“atlanta.voip.ms”,”atlanta2.voip.ms”,”chicago.voip.ms”,”chicago2.voip.ms”,”chicago3.voip.ms”,”chicago4.voip.ms”,”dallas.voip.ms”,
”denver.voip.ms”,”denver2.voip.ms”,”houston.voip.ms”,”london.voip.ms”,”losangeles.voip.ms”,”losangeles2.voip.ms”,”montreal.voip.ms”,
”montreal2.voip.ms”,”montreal3.voip.ms”,”montreal4.voip.ms”,”newyork.voip.ms”,” newyork2.voip.ms”,”newyork3.voip.ms”,”newyork4.voip.ms”,
”seattle.voip.ms”,”seattle2.voip.ms”,”seattle3.voip.ms”,”tampa.voip.ms”,”toronto.voip.ms”,”toronto2.voip.ms”,”toronto3.voip.ms”,
”toronto4.voip.ms”,”vancouver.voip.ms”,”vancouver2.voip.ms”,”washington.voip.ms”,”washington2.voip.ms”)
$k = 0 #Counting variable so we know what server number we are testing
#num of servers to test
$servercount = $servers.length
#Do the following code for each server in our array
For Each($server in $servers){
$k++ #Add one to the counting variable....we are on server #1...then server 2, then server 3 etc...
Write-Progress -Activity "Testing Server: ${server}" -status "Testing Server $k out of $servercount" -percentComplete ($k /
$servercount*100) #Update the progress bar
$i = 0 #Counting variable for number of times we tried to ping a given server
Do{
$pingsuccess = $false #assume a failure
$i++ #Add one to the counting variable.....1st try....2nd try....3rd try etc...
Try{
$currentping = (test-connection $server -count 1 -ErrorAction Stop).responsetime #Try to ping
$pingsuccess = $True #If success full, set success variable
}Catch { $pingsuccess = $false } #Catch the failure and set the success variable to false
}While($pingsuccess -eq $false -and $i -le 5) #Try everything between Do and While up to 5 times, or while $pingsuccess is not
true
If($pingsuccess -and ($currentping -lt $bestping -or (!($bestping)))){ #Compare the last ping test with the best known ping
test....if there is no known best ping test, assume this one is the best $bestping = $currentping #If this is the best ping...save it
$bestserver = $server #Save the best server
}
write-host "tested: $server at $currentping ms after $i attempts" #write the results of the test for this server
}
write-host "The server with the best ping is: $bestserver at $bestping ms" #write the end result

= Latency and its importance =

Latency is very important for Voip, this will determine the time that will take for the data package transmission to reach the destination. A high latency will lead to a delay and echoes in the communication.

Latency is measured in milliseconds (ms) For example: a latency of 150ms is barely noticeable, thus acceptable. Higher than that, quality starts to suffer. When it gets higher than 300 ms, it becomes unacceptable.

[[category:guides]]

Choosing Server

2017-03-22T13:06:28Z

Rod:

[[File:Serverlocation3.png]]

= Choosing a Server =

[http://www.voip.ms VoIP.ms] offers many different servers, but which one should you choose? One misconception is that you should pick the closest to your location, however this is not needed most of the time. For example, if you are in the USA, any of the US servers will provide a really good latency and service quality. The newest server within a city is indicated with the highest number attached to the name, as they are classified in ascending order. Also worth noting is that there is a network tool that will help you when deciding which server you want to use, generally named a "ping", which will provide you the latency between you and the server. Therefore the server which provides you less latency should be used.

=== IPs ===

*Amsterdam, NL ('''amsterdam.voip.ms''') 37.58.88.242
*Atlanta 1, GA ('''atlanta.voip.ms''') 75.127.65.130
*Atlanta 2, GA ('''atlanta2.voip.ms''') 72.9.246.170
*Chicago 1, IL ('''chicago.voip.ms''') 208.100.39.52
*Chicago 2, IL ('''chicago2.voip.ms''') 208.100.39.53
*Chicago 3, IL ('''chicago3.voip.ms''') 208.100.39.54
*Chicago 4, IL ('''chicago4.voip.ms''') 208.100.39.55
*Dallas, TX ('''dallas.voip.ms''') 158.85.149.162
*Dallas 2, TX ('''dallas2.voip.ms''') 158.85.149.163
*Denver 1, CO ('''denver.voip.ms''') 173.248.161.90
*Denver 2, CO ('''denver2.voip.ms''') 173.248.159.210
*Houston, TX ('''houston.voip.ms''') 173.193.85.18
*Houston 2, TX ('''houston2.voip.ms''') 173.193.85.19
*London, UK ('''london.voip.ms''') 159.8.157.212
*Los Angeles 1, CA ('''losangeles.voip.ms''') 96.44.149.186
*Los Angeles 2, CA ('''losangeles2.voip.ms''') 96.44.149.202
*Melbourne, AU ('''melbourne.voip.ms''') 168.1.73.84
*Montreal 1, QC ('''montreal.voip.ms''') 67.205.74.184
*Montreal 2, QC ('''montreal2.voip.ms''') 67.205.74.187
*Montreal 3, QC ('''montreal3.voip.ms''') 192.175.96.68
*Montreal 4, QC ('''montreal4.voip.ms''') 67.205.74.179
*Montreal 5, QC ('''montreal5.voip.ms''') 192.175.96.69
*Montreal 6, QC ('''montreal6.voip.ms''') 192.175.96.70
*Montreal 7, QC ('''montreal7.voip.ms''') 192.175.96.71
*Montreal 8, QC ('''montreal8.voip.ms''') 192.175.96.72
*New York 1, NY ('''newyork.voip.ms''') 74.63.41.218
*New York 2, NY ('''newyork2.voip.ms''') 107.6.67.236
*New York 3, NY ('''newyork3.voip.ms''') 107.6.67.237
*New York 4, NY ('''newyork4.voip.ms''') 107.6.67.238
*New York 5, NY ('''newyork5.voip.ms''') 23.29.136.28
*New York 6, NY ('''newyork6.voip.ms''') 23.29.136.29
*New York 7, NY ('''newyork7.voip.ms''') 23.29.136.38
*New York 8, NY ('''newyork8.voip.ms''') 23.29.136.40
*Paris, FR ('''paris.voip.ms''') 159.8.85.180
*San Jose, CA ('''sanjose.voip.ms''') 23.246.247.146
*San Jose 2, CA ('''sanjose2.voip.ms''') 23.246.247.147
*Seattle 1, WA ('''seattle.voip.ms''') 50.23.160.50
*Seattle 2, WA ('''seattle2.voip.ms''') 50.23.160.51
*Seattle 3, WA ('''seattle3.voip.ms''') 50.23.160.52
*Tampa, FL ('''tampa.voip.ms''') 162.254.144.173
*Tampa 2, FL ('''tampa2.voip.ms''') 162.254.144.176
*Toronto 1, ON ('''toronto.voip.ms''') 158.85.70.148
*Toronto 2, ON ('''toronto2.voip.ms''') 158.85.70.149
*Toronto 3, ON ('''toronto3.voip.ms''') 158.85.70.150
*Toronto 4, ON ('''toronto4.voip.ms''') 158.85.70.151
*Toronto 5, ON ('''toronto5.voip.ms''') 184.75.215.106
*Toronto 6, ON ('''toronto6.voip.ms''') 184.75.215.114
*Toronto 7, ON ('''toronto7.voip.ms''') 184.75.215.146
*Toronto 8, ON ('''toronto8.voip.ms''') 184.75.213.210
*Vancouver 1, BC ('''vancouver.voip.ms''') 162.213.157.82
*Vancouver 2, BC ('''vancouver2.voip.ms''') 162.213.157.117
*Washington 1, DC ('''washington.voip.ms''') 208.43.234.226
*Washington 2, DC ('''washington2.voip.ms''') 208.43.234.227

===Server Realms===

For IOS, Please click here [http://wiki.voip.ms/article/Server_Realms Server Realms] to get the Realm Name for the server you plan on using, this can differ from the Domain Name being used.

Please note that the following servers will not be available to select as a DID Point of Presence for newer accounts:
Atlanta 1, Chicago 1, New York 1, New York 4, Seattle 1, Montreal 1, Montreal 2, Montreal 3, Montreal 4, Toronto, Toronto 2, Toronto 3, Toronto 4.

= What is a Ping? =

Ping is a standard tool used to test network connections. It is mostly used to determine if a server or device can be reached across the network and the latency of the response(the time it takes to send a packet to the destination and for it to return to your computer).

Ping tools are part of Windows, Mac OS X and Linux as well as some routers.

== How does the ping work? ==

It sends request messages to a target network address or DNS names at periodic intervals and measures the time it takes for a response message to arrive and return(better known as latency).

==How to ping on a PC==

Pinging is a command which tells you if the connection between your computer and a particular domain is working correctly.

In Windows, select Start > Programs > Accessories > Command Prompt. This will give you a window like the one below.

Enter the word ping, followed by a space, then the domain name.(montreal.voip.ms) in this case domain is our server name.

If the results show a series of replies, the connection is working. The time shows you how fast the connection is. If you see a "timed out" error instead of a reply, there is a breakdown somewhere between your computer and the domain.

[[File:Ping.gif]]

==How to ping on a Mac Computer==

1- Click on Finder in the dock.

2- Click on Applications.

3- Click on Utilities.

4- Double-click on Network Utility. *

* In OS X Mavericks (10.9.x) this utility app changed location. Launch it from spotlight instead, either press "command"+"space bar" or click on spotlight directly (magnifying glass icon at top right of screen), type "network utility" and hit "return"

5- In the Network Utility window, click on the Ping tab

6- In the field under "Please enter the network address to ping," like montreal.voip.ms

''If pings results are not consistent, you may have an issue with Jitter. You can work on this issue by adjusting the "Network Jitter Level" setting on your VoIP device. Usually a ping of under 150 ms is recommended in order to have good quality. The latency time to the server is important, however there are also other factors that could affect the quality of the calls such as packet loss (VoIP communications are very sensitive to this), and the Jitter level of your Internet connection.''

The following is the output of running ping with the target losangeles.voip.ms.

#ping losangeles.voip.ms
Ping to losangeles.voip.ms [67.215.241.250] with 32 bytes de datos:
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
Response from 67.215.241.250: bytes=32 time=69ms TTL=52
Response from 67.215.241.250: bytes=32 time=68ms TTL=52
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
ping statistics from 67.215.241.250:
4 packets transmitted, 4 received, 0% packet lost. rtt min/avg/max/mdev = 67ms, 69ms, 67ms

Sample ping output in windows:
C:\Windows\system32>ping montreal.voip.ms

Pinging montreal.voip.ms [67.205.74.184] with 32 bytes of data:
Reply from 67.205.74.184: bytes=32 time=85ms TTL=49
Reply from 67.205.74.184: bytes=32 time=86ms TTL=49
Reply from 67.205.74.184: bytes=32 time=86ms TTL=49
Reply from 67.205.74.184: bytes=32 time=85ms TTL=49

Ping statistics for 67.205.74.184:
Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 85ms, Maximum = 86ms, Average = 85ms

=== Sample Linux Shell Script ===
Pings several voip.ms servers

#!/bin/sh
# Ping several servers and display Latency, Jitter and Packet Loss
#
# First, create a text file with all servers you want to ping - one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
myHF="voip_ping_hosts.txt"
# Sample file:
# toronto.voip.ms
# montreal.voip.ms
# seattle.voip.ms
# chicago.voip.ms
# newyork.voip.ms
#
echo "============================================"
printf "%-20s %7s %8s %6s\n" "VoIP Server" "Latency" "Jitter" "Loss"
echo "============================================"
cat ${myHF} |\
while read myLn
do
ping -c 3 -i 5 -q $myLn |\
awk '/^PING / {myH=$2}
/packet loss/ {myPL=$6}
/min\/avg\/max/ {
split($4,myS,"/")
printf( "%-20s %3.1f %1.3f %4s\n", myH, myS[2], myS[4], myPL)
}'
done
echo "============================================"

Output:

============================================
VoIP Server Latency Jitter Loss
============================================
toronto.voip.ms 68.3 0.439 0%
montreal.voip.ms 89.6 0.197 0%
seattle.voip.ms 71.2 0.387 0%
chicago.voip.ms 71.6 0.084 0%
newyork.voip.ms 79.1 0.411 0%
============================================

== Latency Testing Scripts (User Submitted) ==

=== Bash Script To Handle The Mac Ping Output Format ===

<p>To make use of this script (1) save as a plain text file (2) set permissions on the file to executable (3) invoke script
e.g. Save script below using your favourite editor as pingVoipMS.sh (2) chmod u+x pingVoipMS.sh (3) ./pingVoipMS.sh
This is a bash 3.x script, so it also works in Linux, just change the ping packet loss field from 7 to 6 in the final loop below (or wherever the loss field is in your ping output format). Depending upon your distro curl might need to change to wget.
</p>
<pre>
#!/bin/bash
# Ping several servers and display Latency, Jitter and Packet Loss
# Usage: [-c <count>][-i <wait time>][-r test restricted servers][<server list file>]
#
# The optional text file should be formatted with one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
# If no args are supplied, this script will scrape a ping server list from voip.ms
#
USER_FILE=""
COUNT=3; INTERVAL=5; RESTRICTED=0
restrictedList=(atlanta.voip.ms chicago.voip.ms
montreal.voip.ms montreal2.voip.ms montreal3.voip.ms montreal4.voip.ms
newyork.voip.ms newyork4.voip.ms seattle.voip.ms
toronto.voip.ms toronto2.voip.ms toronto3.voip.ms toronto4.voip.ms)

# Handle any passed in script arguments
while getopts c:i:r parm
do
case $parm in
c)count_opt=$OPTARG;;
i)interval_opt=$OPTARG;;
r)RESTRICTED=1;;
*)echo -e "Invalid arg\nUsage:\t[ -c <count of ECHO_REQUESTs to Tx, default 3> ] \
\n\t[ -i <wait time (s) between datagrams, default 5> ] \
\n\t[ -r ] Include restricted servers in latency test \
\n\t[FILE <ping server list> ]";exit 1;;
esac
done

# Test if an option was specified and whether it's a +ve non-zero integer
[[ -n $count_opt && ($count_opt =~ ^[[:digit:]]+$) && $count_opt -gt 0 ]] && COUNT=$count_opt
[[ -n $interval_opt && ($interval_opt =~ ^[[:digit:]]+$) && $interval_opt -gt 0 ]] && INTERVAL=$interval_opt

shift $(($OPTIND -1))

# Validate supplied file (server list)
[[ -n $1 && !(-f $1 && -r $1) ]] && { echo "\"$1\" file does not exist or is not readable"; exit 1; }
[[ -n $1 && -f $1 && -r $1 ]] && USER_FILE="$1"

if [[ -n $USER_FILE ]]
then
# Bash 3.x in macOS does not support readarray, need to do cumbersome array loops instead
while IFS= read -r line; do
serverList+=( "$line" )
done < <(grep -e '[[:alpha:]]*[[:alnum:]]\.voip\.ms' $USER_FILE | grep -v '^\s*#' | awk NF)
else
# N.B. The script looks for the html boldface tags <b> </b> inside a bracket
# If the website alters and the parse fails, manually create the list and
# supply as a script arg (or perhaps update the parsing to work again :)
while IFS= read -r line; do
serverList+=( "$line" )
done < <(curl --silent http://wiki.voip.ms/article/Choosing_Server | \
grep '(<b>[[:alpha:]]*[[:alnum:]]\.voip\.ms</b>)' | \
tr "<>" " " | awk '{print $(NF-3)}' )
fi

# Newer voip.ms clients can't register onto these over-subscribed servers
# Don't test the restricted list unless explicitly asked (with the -r cmd line option)
if [[ $RESTRICTED -eq 0 ]]
then
for server in ${restrictedList[@]}
do
ix=$(printf "%s\n" "${serverList[@]}" | grep -n "^${server}" | cut -d ":" -f1)

while IFS= read -r idx; do
keys+=($idx)
done < <([[ $ix -gt 0 ]] && echo $((ix-1)))
done
for ((i=${#keys[@]} - 1; i >= 0; i--)); do unset serverList[keys[i]]; done
fi

if [[ ${#serverList[@]} -eq 0 ]]
then
echo "No unrestricted Voip.ms servers could be found, please supply a server list"
exit 1
fi

runTime=$(( $COUNT * $INTERVAL * ${#serverList[@]} ))

echo "PING will send $COUNT packet(s) with a wait of $INTERVAL sec(s) between each packet"
echo "Change the PING options by invoking this script with -c and/or -i, default \"-c 3 -i 5\""
echo "Over $((${#serverList[@]})) server(s) the estimated script Run Time will be $runTime seconds"
echo "================================================================"
printf "%-20s %-18s %7s %8s %6s %s\n" "VoIP Server" "IP Address" "Latency" "Jitter" "Loss" "Countdown"
echo "================================================================ (seconds)"

for myLn in ${serverList[@]}
do
while IFS=$'\n' read -r line; do
pingList+=( "$line" )
printf "%-64s %5d %2d/%-2d\n" "$line" "$((runTime - $COUNT * $INTERVAL * ${#pingList[@]}))" "${#pingList[@]}" "${#serverList[@]}"
done < <( ping -c $COUNT -i $INTERVAL -q $myLn | awk '\
/^PING / {myH=$2}
/^PING / {
IP = substr($3,2,15)
split(IP,myIP,")") }
/packet loss/ {myPL=$7}
/min\/avg\/max/ {
split($4,myS,"/")
printf("%-20s %-18s %7.3f %8.3f %6s\n",
myH, myIP[1], myS[2], myS[4], myPL ) }
' )
done

echo "================================================================"
echo -e "\nMost appropriate server listed in order of best latency\n"
echo "================================================================"
printf "%-20s %-18s %7s %8s %6s\n" "VoIP Server" "IP Address" "Latency" "Jitter" "Loss"
echo "================================================================"
printf "%s\n" "${pingList[@]}" | LC_ALL=C sort -n -k 3,3 -k 5,5 -k 4,4 | awk '{printf("%s $%2d$\n",$0, NR)}'
echo "================================================================"
</pre>

=== Perl Script ===
Pings list of voip.ms servers round robin with optional output csv file.

# usage ping_voip.ms.pl <number of times> <seconds in between> <output.csv>
use Net::Ping;
use Time::HiRes;
use strict;

# input list
my @hosts = qw(
atlanta.voip.ms
atlanta2.voip.ms
chicago.voip.ms
chicago2.voip.ms
chicago3.voip.ms
chicago4.voip.ms
dallas.voip.ms
denver.voip.ms
denver2.voip.ms
houston.voip.ms
losangeles.voip.ms
losangeles2.voip.ms
newyork.voip.ms
newyork2.voip.ms
newyork3.voip.ms
newyork4.voip.ms
seattle.voip.ms
seattle2.voip.ms
seattle3.voip.ms
tampa.voip.ms
washington.voip.ms
washington2.voip.ms
montreal.voip.ms
montreal2.voip.ms
montreal3.voip.ms
montreal4.voip.ms
toronto2.voip.ms
toronto3.voip.ms
toronto4.voip.ms
toronto.voip.ms
london.voip.ms
);

$| = 1; #autoflush
# High precision syntax (requires Time::HiRes)
my $p = Net::Ping->new("icmp",1);
$p->hires();
my $max_name_length = (reverse sort { $a <=> $b } map { length($_) } @hosts)[0];
my $count = 4; # number of times to ping
my $interval = 5; # seconds between ping rounds
my $output_file = "";
my @data;

# check for arguments
my $num_args = @ARGV;
if ($num_args >= 1) {$count = $ARGV[0];}
if ($num_args >= 2) {$interval = $ARGV[1];}
if ($num_args >= 3) {$output_file = $ARGV[2];}

# check argument validity
$0 =~ /^.*\\(.*)$/;
my $script = $1;
if ($count !~ /^\d+$/ or $interval !~ /^\d+$/) {die "Usage: $script <number of rounds> <seconds between rounds> <output.csv>\n";}
if (length($output_file) > 0 and $output_file !~ /\.csv$/) {$output_file .= ".csv";}

# main loop
for my $i (1..$count)
{
sleep $interval unless $i == 1;
print "Round $i\n";
my $host_num=0;
foreach my $host (@hosts)
{
(my $ret, my $duration, my $ip) = $p->ping($host);
$ip =~ /(\d+)\.(\d+)\.(\d+)\.(\d+)/;
if ($ret)
{
printf("%*s [ip: %3s.%3s.%3s.%3s] is alive (%6.2f ms)\n", $max_name_length, $host, $1, $2, $3, $4, $duration*1000);
$data[$host_num][$i]=$duration*1000;
}
else
{
printf("%*s [ip: %3s.%3s.%3s.%3s] is dead\n", $max_name_length, $host, $1, $2, $3, $4);
}
$host_num++;
}
print "\n";
}
$p->close();

# if output file name given
if (length($output_file)>0)
{
# print output to file
open FILE, ">$output_file" or die "$!\n";

# print column headers
print FILE "Server\\Round";
for my $i (1..$count)
{
print FILE ", $i";
}
print FILE ", Average\n";

# print data
my $i = 0;
foreach my $host (@hosts)
{
print FILE "$host";
my $sum = 0;
for my $j (1..$count)
{
$sum += $data[$i][$j];
printf FILE ", %8.4f",$data[$i][$j];
}
printf FILE ", %8.4f\n",$sum/$count;
$i++;
}

close FILE;
print "Data written to $output_file\n";
}

# print summary to screen
my $i = 0;
printf("%-*s Average (ms)\n", $max_name_length, "Server");
foreach my $host (@hosts)
{
my $sum = 0;
for my $j (1..$count)
{
$sum += $data[$i][$j];
}
printf("%-*s %8.4f\n", $max_name_length+1, $host, $sum/$count);
$i++;
}

Output:
Round 1
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.97 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.99 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.70 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.76 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.53 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.73 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 94.99 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.05 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.13 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (102.87 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 64.92 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 63.41 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (131.75 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (120.64 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (120.49 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (111.43 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.25 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.86 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 90.85 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (123.29 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.71 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.19 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 81.82 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 86.13 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 77.09 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.18 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.70 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (131.27 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (125.13 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.26 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (152.77 ms)

Round 2
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.14 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.86 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 50.03 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.44 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.33 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.22 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 95.58 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.94 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.29 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (102.73 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.59 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 64.27 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (112.74 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (121.22 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (121.34 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.75 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.06 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.33 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 91.58 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (122.94 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.28 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.40 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 81.91 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 85.64 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 75.15 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.79 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.10 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (150.85 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.40 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.45 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (170.79 ms)

Round 3
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.76 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.86 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.65 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.01 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.05 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.53 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 95.82 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.02 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.60 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (103.35 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.79 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 64.05 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (113.01 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (121.41 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (122.23 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.62 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 93.65 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.19 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 90.75 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (125.12 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.19 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.98 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 80.16 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 87.16 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 76.54 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 97.51 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.18 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (142.81 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.95 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.78 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (153.14 ms)

Round 4
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 89.19 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.98 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.21 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.50 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.68 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.18 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 93.93 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.22 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.10 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (103.67 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.58 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 63.60 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (114.76 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (120.44 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (121.05 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.51 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.04 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 96.92 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 91.23 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (123.28 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.45 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (100.94 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 82.33 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 85.02 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 76.85 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.32 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.22 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (148.33 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (141.61 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (105.91 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (152.85 ms)

Server Average (ms)
atlanta.voip.ms 88.7630
atlanta2.voip.ms 92.9233
chicago.voip.ms 49.6477
chicago2.voip.ms 59.9305
chicago3.voip.ms 59.3972
chicago4.voip.ms 49.9152
dallas.voip.ms 95.0790
denver.voip.ms 94.8077
denver2.voip.ms 85.2797
houston.voip.ms 103.1562
losangeles.voip.ms 65.4693
losangeles2.voip.ms 63.8347
newyork.voip.ms 118.0643
newyork2.voip.ms 120.9265
newyork3.voip.ms 121.2778
newyork4.voip.ms 110.8275
seattle.voip.ms 93.9993
seattle2.voip.ms 95.8267
seattle3.voip.ms 91.1035
tampa.voip.ms 123.6570
washington.voip.ms 98.4065
washington2.voip.ms 101.3774
montreal.voip.ms 81.5525
montreal2.voip.ms 85.9863
montreal3.voip.ms 76.4058
montreal4.voip.ms 96.7013
toronto2.voip.ms 103.7986
toronto3.voip.ms 143.3156
toronto4.voip.ms 136.0254
toronto.voip.ms 104.1012
london.voip.ms 157.3885

=== Powershell ===

# Usage: Copy and paste the following code into a powershell window
# To run it from a command prompt, save this file with extension ps1. Then run Powershell.exe -file "pathtothisscript.ps1"
#Get the list of servers into an array
$Servers =
@(“atlanta.voip.ms”,”atlanta2.voip.ms”,”chicago.voip.ms”,”chicago2.voip.ms”,”chicago3.voip.ms”,”chicago4.voip.ms”,”dallas.voip.ms”,
”denver.voip.ms”,”denver2.voip.ms”,”houston.voip.ms”,”london.voip.ms”,”losangeles.voip.ms”,”losangeles2.voip.ms”,”montreal.voip.ms”,
”montreal2.voip.ms”,”montreal3.voip.ms”,”montreal4.voip.ms”,”newyork.voip.ms”,” newyork2.voip.ms”,”newyork3.voip.ms”,”newyork4.voip.ms”,
”seattle.voip.ms”,”seattle2.voip.ms”,”seattle3.voip.ms”,”tampa.voip.ms”,”toronto.voip.ms”,”toronto2.voip.ms”,”toronto3.voip.ms”,
”toronto4.voip.ms”,”vancouver.voip.ms”,”vancouver2.voip.ms”,”washington.voip.ms”,”washington2.voip.ms”)
$k = 0 #Counting variable so we know what server number we are testing
#num of servers to test
$servercount = $servers.length
#Do the following code for each server in our array
For Each($server in $servers){
$k++ #Add one to the counting variable....we are on server #1...then server 2, then server 3 etc...
Write-Progress -Activity "Testing Server: ${server}" -status "Testing Server $k out of $servercount" -percentComplete ($k /
$servercount*100) #Update the progress bar
$i = 0 #Counting variable for number of times we tried to ping a given server
Do{
$pingsuccess = $false #assume a failure
$i++ #Add one to the counting variable.....1st try....2nd try....3rd try etc...
Try{
$currentping = (test-connection $server -count 1 -ErrorAction Stop).responsetime #Try to ping
$pingsuccess = $True #If success full, set success variable
}Catch { $pingsuccess = $false } #Catch the failure and set the success variable to false
}While($pingsuccess -eq $false -and $i -le 5) #Try everything between Do and While up to 5 times, or while $pingsuccess is not
true
If($pingsuccess -and ($currentping -lt $bestping -or (!($bestping)))){ #Compare the last ping test with the best known ping
test....if there is no known best ping test, assume this one is the best $bestping = $currentping #If this is the best ping...save it
$bestserver = $server #Save the best server
}
write-host "tested: $server at $currentping ms after $i attempts" #write the results of the test for this server
}
write-host "The server with the best ping is: $bestserver at $bestping ms" #write the end result

= Latency and it's importance =

Latency is very important for Voip, this will determine the time that will take for the data package transmission to reach the destination. A high latency will lead to a delay and echoes in the communication.

Latency is measured in milliseconds (ms) For example: a latency of 150ms is barely noticeable, thus acceptable. Higher than that, quality starts to suffer. When it gets higher than 300 ms, it becomes unacceptable.

[[category:guides]]

Choosing Server

2017-03-22T13:05:33Z

Rod:

[[File:Serverlocation3.png]]

= Choosing a Server =

[http://www.voip.ms VoIP.ms] offers many different servers, but which one should you choose? One misconception is that you should pick the closest to your location, however this is not needed most of the time. For example, if you are in the USA, any of the US servers will provide a really good latency and service quality. The newest server within a city is indicated with the highest number attached to the name, as they are classified in ascending order. Also worth noting is that there is a network tool that will help you when deciding which server you want to use, generally named a "ping", which will provide you the latency between you and the server. Therefore the server which provides you less latency should be used.

=== IPs ===

*Amsterdam, NL ('''amsterdam.voip.ms''') 37.58.88.242
*Atlanta 1, GA ('''atlanta.voip.ms''') 75.127.65.130
*Atlanta 2, GA ('''atlanta2.voip.ms''') 72.9.246.170
*Chicago 1, IL ('''chicago.voip.ms''') 208.100.39.52
*Chicago 2, IL ('''chicago2.voip.ms''') 208.100.39.53
*Chicago 3, IL ('''chicago3.voip.ms''') 208.100.39.54
*Chicago 4, IL ('''chicago4.voip.ms''') 208.100.39.55
*Dallas, TX ('''dallas.voip.ms''') 158.85.149.162
*Dallas 2, TX ('''dallas2.voip.ms''') 158.85.149.163
*Denver 1, CO ('''denver.voip.ms''') 173.248.161.90
*Denver 2, CO ('''denver2.voip.ms''') 173.248.159.210
*Houston, TX ('''houston.voip.ms''') 173.193.85.18
*Houston 2, TX ('''houston2.voip.ms''') 173.193.85.19
*London, UK ('''london.voip.ms''') 159.8.157.212
*Los Angeles 1, CA ('''losangeles.voip.ms''') 96.44.149.186
*Los Angeles 2, CA ('''losangeles2.voip.ms''') 96.44.149.202
*Melbourne, AU ('''melbourne.voip.ms''') 168.1.73.84
*Montreal 1, QC ('''montreal.voip.ms''') 67.205.74.184
*Montreal 2, QC ('''montreal2.voip.ms''') 67.205.74.187
*Montreal 3, QC ('''montreal3.voip.ms''') 192.175.96.68
*Montreal 4, QC ('''montreal4.voip.ms''') 67.205.74.179
*Montreal 5, QC ('''montreal5.voip.ms''') 192.175.96.69
*Montreal 6, QC ('''montreal6.voip.ms''') 192.175.96.70
*Montreal 7, QC ('''montreal7.voip.ms''') 192.175.96.71
*Montreal 8, QC ('''montreal8.voip.ms''') 192.175.96.72
*New York 1, NY ('''newyork.voip.ms''') 74.63.41.218
*New York 2, NY ('''newyork2.voip.ms''') 107.6.67.236
*New York 3, NY ('''newyork3.voip.ms''') 107.6.67.237
*New York 4, NY ('''newyork4.voip.ms''') 107.6.67.238
*New York 5, NY ('''newyork5.voip.ms''') 23.29.136.28
*New York 6, NY ('''newyork6.voip.ms''') 23.29.136.29
*New York 7, NY ('''newyork7.voip.ms''') 23.29.136.38
*New York 8, NY ('''newyork8.voip.ms''') 23.29.136.40
*Paris, FR ('''paris.voip.ms''') 159.8.85.180
*San Jose, CA ('''sanjose.voip.ms''') 23.246.247.146
*San Jose 2, CA ('''sanjose2.voip.ms''') 23.246.247.147
*Seattle 1, WA ('''seattle.voip.ms''') 50.23.160.50
*Seattle 2, WA ('''seattle2.voip.ms''') 50.23.160.51
*Seattle 3, WA ('''seattle3.voip.ms''') 50.23.160.52
*Tampa, FL ('''tampa.voip.ms''') 162.254.144.173
*Tampa 2, FL ('''tampa2.voip.ms''') 162.254.144.176
*Toronto 1, ON ('''toronto.voip.ms''') 158.85.70.148
*Toronto 2, ON ('''toronto2.voip.ms''') 158.85.70.149
*Toronto 3, ON ('''toronto3.voip.ms''') 158.85.70.150
*Toronto 4, ON ('''toronto4.voip.ms''') 158.85.70.151
*Toronto 5, ON ('''toronto5.voip.ms''') 184.75.215.106
*Toronto 6, ON ('''toronto6.voip.ms''') 184.75.215.114
*Toronto 7, ON ('''toronto7.voip.ms''') 184.75.215.146
*Toronto 8, ON ('''toronto8.voip.ms''') 184.75.213.210
*Vancouver 1, BC ('''vancouver.voip.ms''') 162.213.157.82
*Vancouver 2, BC ('''vancouver2.voip.ms''') 162.213.157.117
*Washington 1, DC ('''washington.voip.ms''') 208.43.234.226
*Washington 2, DC ('''washington2.voip.ms''') 208.43.234.227

===Server Realms===

For IOS, Please click here [http://wiki.voip.ms/article/Server_Realms Server Realms] to get the Realm Name for the server you plan on using, this can differ from the Domain Name being used.

Please note that the following servers will not be available to select as a DID Point of Presence for newer accounts:
Atlanta 1, Chicago 1, New York 1, New York 4, Seattle 1, Montreal 1, Montreal 2, Montreal 3, Montreal 4, Toronto, Toronto 2, Toronto 3, Toronto 4.

= What is a Ping? =

Ping is a standard tool used to test network connections. It is mostly used to determine if a server or device can be reached across the network and the latency of the response(the time it takes to send a packet to the destination and for it to return to your computer).

Ping tools are part of Windows, Mac OS X and Linux as well as some routers.

== How does the ping work? ==

It sends request messages to a target network address or DNS names at periodic intervals and measures the time it takes for a response message to arrive and return(better known as latency).

==How to ping on a PC==

Pinging is a command which tells you if the connection between your computer and a particular domain is working correctly.

In Windows, select Start > Programs > Accessories > Command Prompt. This will give you a window like the one below.

Enter the word ping, followed by a space, then the domain name.(montreal.voip.ms) in this case domain is our server name.

If the results show a series of replies, the connection is working. The time shows you how fast the connection is. If you see a "timed out" error instead of a reply, there is a breakdown somewhere between your computer and the domain.

[[File:Ping.gif]]

==How to ping on a Mac Computer==

1- Click on Finder in the dock.

2- Click on Applications.

3- Click on Utilities.

4- Double-click on Network Utility. *

* In OS X Mavericks (10.9.x) this utility app changed location. Launch it from spotlight instead, either press "command"+"space bar" or click on spotlight directly (magnifying glass icon at top right of screen), type "network utility" and hit "return"

5- In the Network Utility window, click on the Ping tab

6- In the field under "Please enter the network address to ping," like montreal.voip.ms

''If pings results are not consistent, you may have an issue with Jitter. You can work on this issue by adjusting the "Network Jitter Level" setting on your VoIP device. Usually a ping of under 150 ms is recommended in order to have good quality. The latency time to the server is important, however there are also other factors that could affect the quality of the calls such as packet loss (VoIP communications are very sensitive to this), and the Jitter level of your Internet connection.''

The following is the output of running ping with the target losangeles.voip.ms.

#ping losangeles.voip.ms
Ping to losangeles.voip.ms [67.215.241.250] with 32 bytes de datos:
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
Response from 67.215.241.250: bytes=32 time=69ms TTL=52
Response from 67.215.241.250: bytes=32 time=68ms TTL=52
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
ping statistics from 67.215.241.250:
4 packets transmitted, 4 received, 0% packet lost. rtt min/avg/max/mdev = 67ms, 69ms, 67ms

Sample ping output in windows:
C:\Windows\system32>ping montreal.voip.ms

Pinging montreal.voip.ms [67.205.74.184] with 32 bytes of data:
Reply from 67.205.74.184: bytes=32 time=85ms TTL=49
Reply from 67.205.74.184: bytes=32 time=86ms TTL=49
Reply from 67.205.74.184: bytes=32 time=86ms TTL=49
Reply from 67.205.74.184: bytes=32 time=85ms TTL=49

Ping statistics for 67.205.74.184:
Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 85ms, Maximum = 86ms, Average = 85ms

=== Sample Linux Shell Script ===
Pings several voip.ms servers

#!/bin/sh
# Ping several servers and display Latency, Jitter and Packet Loss
#
# First, create a text file with all servers you want to ping - one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
myHF="voip_ping_hosts.txt"
# Sample file:
# toronto.voip.ms
# montreal.voip.ms
# seattle.voip.ms
# chicago.voip.ms
# newyork.voip.ms
#
echo "============================================"
printf "%-20s %7s %8s %6s\n" "VoIP Server" "Latency" "Jitter" "Loss"
echo "============================================"
cat ${myHF} |\
while read myLn
do
ping -c 3 -i 5 -q $myLn |\
awk '/^PING / {myH=$2}
/packet loss/ {myPL=$6}
/min\/avg\/max/ {
split($4,myS,"/")
printf( "%-20s %3.1f %1.3f %4s\n", myH, myS[2], myS[4], myPL)
}'
done
echo "============================================"

Output:

============================================
VoIP Server Latency Jitter Loss
============================================
toronto.voip.ms 68.3 0.439 0%
montreal.voip.ms 89.6 0.197 0%
seattle.voip.ms 71.2 0.387 0%
chicago.voip.ms 71.6 0.084 0%
newyork.voip.ms 79.1 0.411 0%
============================================

== Latency Testing Scripts (User Submitted ==

==Bash Script To Handle The Mac Ping Output Format ==

<p>To make use of this script (1) save as a plain text file (2) set permissions on the file to executable (3) invoke script
e.g. Save script below using your favourite editor as pingVoipMS.sh (2) chmod u+x pingVoipMS.sh (3) ./pingVoipMS.sh
This is a bash 3.x script, so it also works in Linux, just change the ping packet loss field from 7 to 6 in the final loop below (or wherever the loss field is in your ping output format). Depending upon your distro curl might need to change to wget.
</p>
<pre>
#!/bin/bash
# Ping several servers and display Latency, Jitter and Packet Loss
# Usage: [-c <count>][-i <wait time>][-r test restricted servers][<server list file>]
#
# The optional text file should be formatted with one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
# If no args are supplied, this script will scrape a ping server list from voip.ms
#
USER_FILE=""
COUNT=3; INTERVAL=5; RESTRICTED=0
restrictedList=(atlanta.voip.ms chicago.voip.ms
montreal.voip.ms montreal2.voip.ms montreal3.voip.ms montreal4.voip.ms
newyork.voip.ms newyork4.voip.ms seattle.voip.ms
toronto.voip.ms toronto2.voip.ms toronto3.voip.ms toronto4.voip.ms)

# Handle any passed in script arguments
while getopts c:i:r parm
do
case $parm in
c)count_opt=$OPTARG;;
i)interval_opt=$OPTARG;;
r)RESTRICTED=1;;
*)echo -e "Invalid arg\nUsage:\t[ -c <count of ECHO_REQUESTs to Tx, default 3> ] \
\n\t[ -i <wait time (s) between datagrams, default 5> ] \
\n\t[ -r ] Include restricted servers in latency test \
\n\t[FILE <ping server list> ]";exit 1;;
esac
done

# Test if an option was specified and whether it's a +ve non-zero integer
[[ -n $count_opt && ($count_opt =~ ^[[:digit:]]+$) && $count_opt -gt 0 ]] && COUNT=$count_opt
[[ -n $interval_opt && ($interval_opt =~ ^[[:digit:]]+$) && $interval_opt -gt 0 ]] && INTERVAL=$interval_opt

shift $(($OPTIND -1))

# Validate supplied file (server list)
[[ -n $1 && !(-f $1 && -r $1) ]] && { echo "\"$1\" file does not exist or is not readable"; exit 1; }
[[ -n $1 && -f $1 && -r $1 ]] && USER_FILE="$1"

if [[ -n $USER_FILE ]]
then
# Bash 3.x in macOS does not support readarray, need to do cumbersome array loops instead
while IFS= read -r line; do
serverList+=( "$line" )
done < <(grep -e '[[:alpha:]]*[[:alnum:]]\.voip\.ms' $USER_FILE | grep -v '^\s*#' | awk NF)
else
# N.B. The script looks for the html boldface tags <b> </b> inside a bracket
# If the website alters and the parse fails, manually create the list and
# supply as a script arg (or perhaps update the parsing to work again :)
while IFS= read -r line; do
serverList+=( "$line" )
done < <(curl --silent http://wiki.voip.ms/article/Choosing_Server | \
grep '(<b>[[:alpha:]]*[[:alnum:]]\.voip\.ms</b>)' | \
tr "<>" " " | awk '{print $(NF-3)}' )
fi

# Newer voip.ms clients can't register onto these over-subscribed servers
# Don't test the restricted list unless explicitly asked (with the -r cmd line option)
if [[ $RESTRICTED -eq 0 ]]
then
for server in ${restrictedList[@]}
do
ix=$(printf "%s\n" "${serverList[@]}" | grep -n "^${server}" | cut -d ":" -f1)

while IFS= read -r idx; do
keys+=($idx)
done < <([[ $ix -gt 0 ]] && echo $((ix-1)))
done
for ((i=${#keys[@]} - 1; i >= 0; i--)); do unset serverList[keys[i]]; done
fi

if [[ ${#serverList[@]} -eq 0 ]]
then
echo "No unrestricted Voip.ms servers could be found, please supply a server list"
exit 1
fi

runTime=$(( $COUNT * $INTERVAL * ${#serverList[@]} ))

echo "PING will send $COUNT packet(s) with a wait of $INTERVAL sec(s) between each packet"
echo "Change the PING options by invoking this script with -c and/or -i, default \"-c 3 -i 5\""
echo "Over $((${#serverList[@]})) server(s) the estimated script Run Time will be $runTime seconds"
echo "================================================================"
printf "%-20s %-18s %7s %8s %6s %s\n" "VoIP Server" "IP Address" "Latency" "Jitter" "Loss" "Countdown"
echo "================================================================ (seconds)"

for myLn in ${serverList[@]}
do
while IFS=$'\n' read -r line; do
pingList+=( "$line" )
printf "%-64s %5d %2d/%-2d\n" "$line" "$((runTime - $COUNT * $INTERVAL * ${#pingList[@]}))" "${#pingList[@]}" "${#serverList[@]}"
done < <( ping -c $COUNT -i $INTERVAL -q $myLn | awk '\
/^PING / {myH=$2}
/^PING / {
IP = substr($3,2,15)
split(IP,myIP,")") }
/packet loss/ {myPL=$7}
/min\/avg\/max/ {
split($4,myS,"/")
printf("%-20s %-18s %7.3f %8.3f %6s\n",
myH, myIP[1], myS[2], myS[4], myPL ) }
' )
done

echo "================================================================"
echo -e "\nMost appropriate server listed in order of best latency\n"
echo "================================================================"
printf "%-20s %-18s %7s %8s %6s\n" "VoIP Server" "IP Address" "Latency" "Jitter" "Loss"
echo "================================================================"
printf "%s\n" "${pingList[@]}" | LC_ALL=C sort -n -k 3,3 -k 5,5 -k 4,4 | awk '{printf("%s $%2d$\n",$0, NR)}'
echo "================================================================"
</pre>

=== Perl Script ===
Pings list of voip.ms servers round robin with optional output csv file.

# usage ping_voip.ms.pl <number of times> <seconds in between> <output.csv>
use Net::Ping;
use Time::HiRes;
use strict;

# input list
my @hosts = qw(
atlanta.voip.ms
atlanta2.voip.ms
chicago.voip.ms
chicago2.voip.ms
chicago3.voip.ms
chicago4.voip.ms
dallas.voip.ms
denver.voip.ms
denver2.voip.ms
houston.voip.ms
losangeles.voip.ms
losangeles2.voip.ms
newyork.voip.ms
newyork2.voip.ms
newyork3.voip.ms
newyork4.voip.ms
seattle.voip.ms
seattle2.voip.ms
seattle3.voip.ms
tampa.voip.ms
washington.voip.ms
washington2.voip.ms
montreal.voip.ms
montreal2.voip.ms
montreal3.voip.ms
montreal4.voip.ms
toronto2.voip.ms
toronto3.voip.ms
toronto4.voip.ms
toronto.voip.ms
london.voip.ms
);

$| = 1; #autoflush
# High precision syntax (requires Time::HiRes)
my $p = Net::Ping->new("icmp",1);
$p->hires();
my $max_name_length = (reverse sort { $a <=> $b } map { length($_) } @hosts)[0];
my $count = 4; # number of times to ping
my $interval = 5; # seconds between ping rounds
my $output_file = "";
my @data;

# check for arguments
my $num_args = @ARGV;
if ($num_args >= 1) {$count = $ARGV[0];}
if ($num_args >= 2) {$interval = $ARGV[1];}
if ($num_args >= 3) {$output_file = $ARGV[2];}

# check argument validity
$0 =~ /^.*\\(.*)$/;
my $script = $1;
if ($count !~ /^\d+$/ or $interval !~ /^\d+$/) {die "Usage: $script <number of rounds> <seconds between rounds> <output.csv>\n";}
if (length($output_file) > 0 and $output_file !~ /\.csv$/) {$output_file .= ".csv";}

# main loop
for my $i (1..$count)
{
sleep $interval unless $i == 1;
print "Round $i\n";
my $host_num=0;
foreach my $host (@hosts)
{
(my $ret, my $duration, my $ip) = $p->ping($host);
$ip =~ /(\d+)\.(\d+)\.(\d+)\.(\d+)/;
if ($ret)
{
printf("%*s [ip: %3s.%3s.%3s.%3s] is alive (%6.2f ms)\n", $max_name_length, $host, $1, $2, $3, $4, $duration*1000);
$data[$host_num][$i]=$duration*1000;
}
else
{
printf("%*s [ip: %3s.%3s.%3s.%3s] is dead\n", $max_name_length, $host, $1, $2, $3, $4);
}
$host_num++;
}
print "\n";
}
$p->close();

# if output file name given
if (length($output_file)>0)
{
# print output to file
open FILE, ">$output_file" or die "$!\n";

# print column headers
print FILE "Server\\Round";
for my $i (1..$count)
{
print FILE ", $i";
}
print FILE ", Average\n";

# print data
my $i = 0;
foreach my $host (@hosts)
{
print FILE "$host";
my $sum = 0;
for my $j (1..$count)
{
$sum += $data[$i][$j];
printf FILE ", %8.4f",$data[$i][$j];
}
printf FILE ", %8.4f\n",$sum/$count;
$i++;
}

close FILE;
print "Data written to $output_file\n";
}

# print summary to screen
my $i = 0;
printf("%-*s Average (ms)\n", $max_name_length, "Server");
foreach my $host (@hosts)
{
my $sum = 0;
for my $j (1..$count)
{
$sum += $data[$i][$j];
}
printf("%-*s %8.4f\n", $max_name_length+1, $host, $sum/$count);
$i++;
}

Output:
Round 1
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.97 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.99 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.70 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.76 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.53 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.73 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 94.99 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.05 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.13 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (102.87 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 64.92 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 63.41 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (131.75 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (120.64 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (120.49 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (111.43 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.25 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.86 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 90.85 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (123.29 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.71 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.19 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 81.82 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 86.13 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 77.09 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.18 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.70 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (131.27 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (125.13 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.26 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (152.77 ms)

Round 2
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.14 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.86 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 50.03 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.44 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.33 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.22 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 95.58 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.94 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.29 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (102.73 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.59 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 64.27 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (112.74 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (121.22 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (121.34 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.75 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.06 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.33 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 91.58 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (122.94 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.28 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.40 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 81.91 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 85.64 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 75.15 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.79 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.10 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (150.85 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.40 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.45 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (170.79 ms)

Round 3
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.76 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.86 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.65 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.01 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.05 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.53 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 95.82 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.02 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.60 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (103.35 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.79 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 64.05 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (113.01 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (121.41 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (122.23 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.62 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 93.65 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.19 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 90.75 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (125.12 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.19 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.98 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 80.16 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 87.16 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 76.54 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 97.51 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.18 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (142.81 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.95 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.78 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (153.14 ms)

Round 4
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 89.19 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.98 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.21 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.50 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.68 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.18 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 93.93 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.22 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.10 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (103.67 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.58 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 63.60 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (114.76 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (120.44 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (121.05 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.51 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.04 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 96.92 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 91.23 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (123.28 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.45 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (100.94 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 82.33 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 85.02 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 76.85 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.32 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.22 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (148.33 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (141.61 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (105.91 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (152.85 ms)

Server Average (ms)
atlanta.voip.ms 88.7630
atlanta2.voip.ms 92.9233
chicago.voip.ms 49.6477
chicago2.voip.ms 59.9305
chicago3.voip.ms 59.3972
chicago4.voip.ms 49.9152
dallas.voip.ms 95.0790
denver.voip.ms 94.8077
denver2.voip.ms 85.2797
houston.voip.ms 103.1562
losangeles.voip.ms 65.4693
losangeles2.voip.ms 63.8347
newyork.voip.ms 118.0643
newyork2.voip.ms 120.9265
newyork3.voip.ms 121.2778
newyork4.voip.ms 110.8275
seattle.voip.ms 93.9993
seattle2.voip.ms 95.8267
seattle3.voip.ms 91.1035
tampa.voip.ms 123.6570
washington.voip.ms 98.4065
washington2.voip.ms 101.3774
montreal.voip.ms 81.5525
montreal2.voip.ms 85.9863
montreal3.voip.ms 76.4058
montreal4.voip.ms 96.7013
toronto2.voip.ms 103.7986
toronto3.voip.ms 143.3156
toronto4.voip.ms 136.0254
toronto.voip.ms 104.1012
london.voip.ms 157.3885

=== Powershell ===

# Usage: Copy and paste the following code into a powershell window
# To run it from a command prompt, save this file with extension ps1. Then run Powershell.exe -file "pathtothisscript.ps1"
#Get the list of servers into an array
$Servers =
@(“atlanta.voip.ms”,”atlanta2.voip.ms”,”chicago.voip.ms”,”chicago2.voip.ms”,”chicago3.voip.ms”,”chicago4.voip.ms”,”dallas.voip.ms”,
”denver.voip.ms”,”denver2.voip.ms”,”houston.voip.ms”,”london.voip.ms”,”losangeles.voip.ms”,”losangeles2.voip.ms”,”montreal.voip.ms”,
”montreal2.voip.ms”,”montreal3.voip.ms”,”montreal4.voip.ms”,”newyork.voip.ms”,” newyork2.voip.ms”,”newyork3.voip.ms”,”newyork4.voip.ms”,
”seattle.voip.ms”,”seattle2.voip.ms”,”seattle3.voip.ms”,”tampa.voip.ms”,”toronto.voip.ms”,”toronto2.voip.ms”,”toronto3.voip.ms”,
”toronto4.voip.ms”,”vancouver.voip.ms”,”vancouver2.voip.ms”,”washington.voip.ms”,”washington2.voip.ms”)
$k = 0 #Counting variable so we know what server number we are testing
#num of servers to test
$servercount = $servers.length
#Do the following code for each server in our array
For Each($server in $servers){
$k++ #Add one to the counting variable....we are on server #1...then server 2, then server 3 etc...
Write-Progress -Activity "Testing Server: ${server}" -status "Testing Server $k out of $servercount" -percentComplete ($k /
$servercount*100) #Update the progress bar
$i = 0 #Counting variable for number of times we tried to ping a given server
Do{
$pingsuccess = $false #assume a failure
$i++ #Add one to the counting variable.....1st try....2nd try....3rd try etc...
Try{
$currentping = (test-connection $server -count 1 -ErrorAction Stop).responsetime #Try to ping
$pingsuccess = $True #If success full, set success variable
}Catch { $pingsuccess = $false } #Catch the failure and set the success variable to false
}While($pingsuccess -eq $false -and $i -le 5) #Try everything between Do and While up to 5 times, or while $pingsuccess is not
true
If($pingsuccess -and ($currentping -lt $bestping -or (!($bestping)))){ #Compare the last ping test with the best known ping
test....if there is no known best ping test, assume this one is the best $bestping = $currentping #If this is the best ping...save it
$bestserver = $server #Save the best server
}
write-host "tested: $server at $currentping ms after $i attempts" #write the results of the test for this server
}
write-host "The server with the best ping is: $bestserver at $bestping ms" #write the end result

= Latency and it's importance =

Latency is very important for Voip, this will determine the time that will take for the data package transmission to reach the destination. A high latency will lead to a delay and echoes in the communication.

Latency is measured in milliseconds (ms) For example: a latency of 150ms is barely noticeable, thus acceptable. Higher than that, quality starts to suffer. When it gets higher than 300 ms, it becomes unacceptable.

[[category:guides]]

Softphones

2017-03-03T17:27:30Z

Rod:

A softphone is a software program for making telephone calls over the Internet using a general purpose computer, tablet or smartphone, rather than using dedicated hardware. Often a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a phone with a display panel and buttons with which the user can interact. A softphone is usually used with a headset connected to the sound card of the PC, or as an installed app on your phone. See also: [[Smartphone]]

==3CX Phone==

[[File:3cx_3.png|200px|thumb|left|3CX Phone]]

'''Product:''' 3CX

'''Company:''' 3CX

'''OS:''' Windows, IOS, Android

'''Overview:'''
3CXPhone is a free VoIP phone available for Windows, iPhone or any Android based smart phone such as Google Nexus, Sony Xperia, Motorola Droid or Samsung Galaxy.

Earlier versions of 3CXPhone could connect to a VoIP provider or to a VoIP PBX to make calls to any VoIP, mobile or landline number. The current versions (3CXPhone client 11 and 12) for the Android platform only work with a same-version [[3CX Phone System]] server; that server in turn should be able to connect to a standard SIP provider such as VoIP.ms. Another option is to use the Windows 3CX desktop client, which has is no licensing fee and is not tied to a particular VoIP provider.

[[3CX_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Acrobits==

[[File:AcrobitsLogo.png|left|Acrobits]]

'''Product:''' Acrobits

'''Company:''' Acrobits

'''OS:''' IOS, Android

'''Overview:'''
Acrobits is a mobile software development company with a focus on SIP Clients for mobile devices; their SIP client, Acrobits Softphone, is the leading app of it's type on the iTunes App Store.

[[Acrobits|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

== Android Sip Client ==

[[File:android_sip_client.jpg|left]]

'''Product:''' Android

'''Company:''' Google Android

'''OS:''' Android

'''Overview:''' Android SIP Client that is included with Android 4.1.1. We cannot guarantee 100% compatibility with all future Android releases since settings may change. We recommend the use of a Wi-Fi connection with these features, instead of a 3G connection. You also could be billed for your data usage if used over 3G/EDGE.

[[Android_Sip_Client|See Configuration Details]]

==Bria==

[[File:Bria.jpg|left|Bria]]

'''Product:''' Bria

'''Company:''' CounterPath

'''OS:''' Windows, MacOS, Linux (Ubuntu), IOS, Android

'''App Storage Consumption:''' 6.11mb (intl. mem only- SD card unusable. Data uses an additional ~2mb approx.)

'''Overview:'''
Bria 3 is a carrier-grade next generation softphone application that enables you to manage your communications easily and efficiently – all from your computer desktop. Replacing or complementing your hard phone, the Bria softphone allows you to make VoIP and Video calls over IP, see when your contacts are available, send Instant Messages and transfer files with ease and efficiency.

Bria Android Edition is a highly secure, standards-based mobile VoIP softphone that works over both 3G and Wi-Fi networks.

Using the device’s existing contact list, Bria Android Edition facilitates easy and effective communication management with an intuitive interface. Deskphone-class calling functionality includes the ability to swap between two calls, merge calls and perform attended and unattended transfers.

[[Bria|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==CSipSimple==

[[File:Csip-0.png|200px|thumb|left|CSipSimple]]

'''Product:''' CSipSimple

'''Company:''' CSipSimple

'''OS:''' Android

'''Overview:'''
CSipSimple is a Voice over Internet Protocol (VoIP) application for Google Android operating system using the SIP.

CSipSimple is an open source and free software released under the GNU General Public License.

It relies on the pjsip SIP stack and get features provided by this sip stack.

The key features of this software are :

Various codecs support : Speex (narrow band and wideband), G.711 (u-law, a-law), GSM, iLBC, SILK(narrow band, wideband and ultra wideband), G.722

*Can use native audio driver
*STUN and ICE NAT traversal
*Integration with Android operating system with filters and rewriting rules
*A distribution of the software available on the website allow SRTP, SIP over TLS and ZRTP
*SIP-SIMPLE messaging

[[CSipSimple|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Dialer+==

[[File:Dialerplus.png|200px|thumb|left|Dialer+]]

'''Product:''' cd26

'''Company:''' cd26

'''OS:''' IOS

'''Overview:'''
Dialer+ is a SIP-based softphone for iPhone , iPod touch and iPad

Standard Phone Features
* Fast Contact Search – Use phone pad to find contacts by names, initials or any part of the numbers.
• iOS 4.0+ Background Support
• Supports GSM calls when the network is unavailable or slowly
• Multiple Account Support
• Auto switch to best service provider when use multiple account
• Pre-configured VoIP providers list
• Call display
• Speakerphone, Mute and Hold
• Call history
• Contact List and Contact Favorites - leveraging the iPhone Contacts
• Ringtones and contact avatars
• Multiple incoming Call Support - swap between two active calls; merge and split calls
• Audio codecs include G.711,G.722, iLBC ,Speex and GSM, Make an in app purchase to add G.729 for great performance over 3G networks
• Support for RFC 2833 DTMF
• VPN Support
• uPNP Support
• iPhone 4 Retina Display Support

[[Dialer+|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Diax==

[[File:Diax1.png|200px|thumb|left|Diax Softphone]]

'''Product:''' Diax

'''Company:''' Diax

'''OS:''' Windows

'''Overview:'''This is a full featured and very small IAX based software phone for Microsoft Windows platforms (only). The application is distributed as a freeware for personal use.

This is a full featured and very small (just around 300KB for all the files) IAX2 based software phone for Microsoft. It is based on the iaxclient library.

[[Diax_Softphone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Ekiga==

[[File:Ekigaw0.png|200px|thumb|left|Ekiga Softphone]]

'''Product:''' Ekiga

'''Company:''' Ekiga

'''OS:''' Windows, Linux

'''Overview:'''Ekiga is a software phone and video conferencing application. It allows you to hear and see your friends for free using your computer and Internet. You can also chat with them and see if they are online or not. It works on GNU/Linux and Windows.

More specifically, Ekiga is a VoIP, IP Telephony, and Video Conferencing application that allows you to make audio and video calls to remote users with SIP or H.323 compatible hardware and software.

It supports many audio and video codecs and all modern VoIP features for both SIP and H.323. Ekiga is the first Open Source application to support both H.323 and SIP, as well as audio and video. Ekiga was formerly known as GnomeMeeting.

[[Ekiga_Softphone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Express Talk==

[[File:Express0.png|200px|thumb|left|Express Talk]]

'''Product:''' Express Talk

'''Company:''' NCH Software

'''OS:''' Windows, MacOS

'''Overview:''' Express Talk works like a telephone to let you make calls through your computer. Call anyone via the internet who also has a softphone installed and if you sign up with a VoIP gateway service company you call regular telephone numbers as well.

With Express Talk you could potentially call someone using their internet IP (eg. "bob@1.2.3.4") but this is not usually practical. It is much easier to use a friendly address like "bob@myphoneco.com". To have this you need to sign up with a SIP proxy service. This is almost always free to signup for and usually requires nothing more than an email address. See our Recommended SIP Service Providers List for a list of options.

Some of the companies also provide gateway services. This means connecting phone calls to ordinary (non internet or analog PSTN) phone lines. These calls tend to be much less expensive particularly for international calls compared to regular long distance charges since they can offer local call rates.

[[Express_Talk|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==FgVoIP==

[[File:Fgvoip0.jpeg|200px|thumb|left|FgVoIP]]

'''Product:''' FgVoIP

'''Company:''' FG Microtec

'''OS:''' BlackberryOS

'''Overview:''' fg's BlackBerry SIP VoIP Client is a feature rich soft phone application for BlackBerry devices. It can register to any SIP VoIP provider or IP/SIP capable PBX using a WiFi connection in office, campus, home, hotel rooms, public hot-spots etc.

Its much like having a BlackBerry desktop phone in your pocket, providing significantly higher overall productivity while at the same time lowering telephony costs.

Experience the freedom of choosing your own VoIP provider whenever you want, wherever you want. All you need is a WiFi network and a VoIP account.

[[FgVoIP|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Fring==

[[File:Fring00.jpg|left|Fring]]

'''Product:''' Fring

'''Company:''' Fring

'''OS:''' Windows, MacOS, IOS, Linux, Android, Symbian

'''Overview:'''
Fring mobile communication service that gives users internet-rich communication from their mobile phones.

Fring members make FREE video calls, voice calls, live chat to other fringsters and to friends on other social networks.

You can now use your SIP account even from non-SIP enabled phones.

Choose whichever SIP provider you have and in just few clicks, start using this account to make calls to landlines, GSM phones, or to other SIP providers.

Supports Symbian S60, iPhone/ iPod touch, Android, Windows Mobile, MeeGo, Linux, J2MEand proprietary OS.

Fring allows you to easily switch between them, without having to reconfigure your handset through a nightmarish process.

[[Fring|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Grandstream Wave==

[[File:Grandstream Wave.png|200px|Thumb|left|Grandstream Wave]]

'''Product:''' Grandstream Wave

'''Company:''' Grandstream

'''OS:''' Android

'''Overview:'''
Grandstream Wave is a FREE softphone application that allows users to make and receive voice calls through their business or residential SIP accounts on any Android™ device (version 4.0+) from anywhere in the world.

This application supports integration of up to 6 SIP accounts, 6-way voice conferencing, and allows users to monitor their PBX (such as Grandstream's UCM6100 series IP PBX & UCM6510 IP PBX) while utilizing speed dial with up to 24 virtual BLF keys.

Grandstream Wave also supports advanced SIP telephony features including call transfer, LDAP phonebook integration and more.

[[Grandstream Wave|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Idoubs==

[[File:Idoubs2.png|left|Idoubs]]

'''Product:''' Idoubs

'''Company:''' Idoubs

'''OS:''' iOS (iPhone, iPad and iPod Touch) and MAC OS X

'''Overview:'''
iDoubs v2.x is the first fully featured open source 3GPP IMS Client for iOS devices (iPhone, iPod Touch and iPad).

As the SIP follows RFC 3261 and 3GPP TS 24.229 Rel-9 specifications, this will allow you to connect to any compliant SIP registrar. <br>

Crystal clear audio quality (adaptive jitter buffer, echo cancellation, noise suppression, automatic resampling, gain control, ...)<br>

Better video quality (low latency, low cpu usage, ...)<br>

Audio codecs: G.722, G.729AB, AMR-NB, iLBC, GSM, PCMA, PCMU, Speex-NB, Speex-WB, Speex-UWB<br>

Video codecs: . VP8, H264, MP4V-ES, Theora, H.263, H.263-1998<br>

Chat (SMS-like) & Favorites<br>

[[Idoubs Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Jitsi==

[[File:Logo2.png|200px|left|Jitsi]]

'''Product:''' Jitsi

'''Company:''' Jitsi

'''OS:''' Windows, MacOS, Linux

'''Overview:''' Jitsi (formerly SIP Communicator) is an audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, AIM/ICQ, Windows Live, Yahoo! and many other useful features.

Jitsi is an Open Source / Free Software, and is available under the terms of the LGPL.

[[Jitsi|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Kiax==

[[File:Kiax3.png|left|200px|Kiax]]

'''Product:''' Kiax

'''Company:''' Kiax

'''OS:''' Windows, MacOS, Linux

'''Overview:'''
Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.

[[Kiax|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Linphone==

[[File:Linphone.jpg|left|200px|Linphone]]

'''Product''': Linphone

'''Company''': Linphone

'''OS:''' Windows, Linux, MacOSX, IPhone, Android.

'''App Storage Consumption:''' 8.09mb (ability to move app to SD card. Data uses an additional ~0.89mb approx.)

'''Overview:''' Linphone is an internet phone or Voice Over IP phone (VoIP). With linphone you can communicate freely with people over the internet, with voice, video, and text instant messaging. Linphone makes use of the SIP protocol, an open standard for internet telephony. You can use Linphone with any SIP VoIP operator.

[[Linphone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==NetDial Sip Phone==

[[File:NetDial.jpg|left|200px|Linphone]]

'''Product''': NetDial Sip Phone

'''Company''': NeoMecca

'''OS:''' iOS, Android.

[[NetDial_Sip_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Media5-fone==

[[File:Media5Android.jpg|left|Media5Android]]

'''Product:''' Media5-fone

'''Company:''' Media5 Corporation

'''OS:''' Android, iOS

'''Overview:'''
The Media5-fone is a softphone application that runs on the Android and the iOS Operating System. It is a SIP Client (softphone) that enables users to make and receive VoIP calls, enabling them to use their devices as an IP-PBX phone extension in their office or anywhere else in the world.

[[Media5-fone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==MizuPhone==

[[File:Mizuphone3.png|left|200px|MizuPhone]]

'''Product:''' MizuPhone

'''Company:''' Mizutech

'''OS:''' Windows

'''Overview:'''
Mizu SoftPhone is our award winner professional VoIP client application based on the open standard SIP protocol with an easy to use modern interface. With Mizu SoftPhone you can connect to any SIP server on the public internet or on your local area network.

Multiple accounts and multiple SIP server registrations combined with a powerful dial plan can minimize your telecom bills while using the greatest features from the VoIP industry.

Features include built-in encryption, IM, presence, HD Video, history with voice records, skype like voice quality using ultra-wideband codecs and much more.

A free edition is also available based on the same engine as our corporate sip softphone but it has some features disabled (like G.729 and wideband codecs, multiple accounts and file transfer over SIP).

[[MizuPhone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==NinjaLite==

[[File:Ninja6.png|left|200px|NinjaLite]]

'''Product:''' NinjaLite

'''Company:''' Global IP Telecommunications

'''OS:''' Windows

'''Overview:'''
Ninja Lite offers Audio- and video telephony for everybody, Ninja features individual skins and ringtones!, Ninja is natively Vista 32bit /64 bit and Windows7 compatible.

[[NinjaLite|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Nokia E71==

[[File:NokiaE71.jpg|left|150px|PhonerLite]]

'''Product:''' Nokia E71 SIP Client

'''Company:''' Nokia

'''OS:''' Symbian

[[Nokia E71|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Nokia N82==

[[File:Nokia_n82_Black.jpg|left|222px|NokiaN82]]

'''Product:''' Nokia N82 SIP Client

'''Company:''' Nokia

'''OS:''' Symbian OS 9.2, S60 3rd Edition, Feature Pack 1

[[Nokia_N82|Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==PhonerLite==

[[File:PhonerLite.png|left|300px|PhonerLite]]

'''Product:''' PhonerLite

'''Company:''' PhonerLite

'''OS:''' Windows

'''Overview:'''
PhonerLite is a clearly arranged application for Windows. PhonerLite enables your PC to use it for internet telephony (VoIP , Voice over IP ). Pre-conditions are a full-duplex sound card , a microphone and speakers (alternativelya headset), an internet connection and a registration at a provider supporting the protocol SIP .PhonerLite supports several SIP profiles, each configurable independently. In thesame way the integrated phone book and call log are easy to use.

[[PhonerLite|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==PortGo==

[[File:Portgo3.png|left|200px|PortGo]]

'''Product:''' PortGo

'''Company:''' PortSIP Solutions, Inc

'''OS:''' Windows (desktop and mobile)

'''Overview:'''
PortGo is the newest SIP softphone from PortSIP, it's built base on PortSIP VoIP SDK, allowing users to enjoy multimedia communications in a dynamic way.

Featuring an intuitive interface, PortGo is expanding the softphone experience by making it even easier to make VoIP and Video over IP calls, see when your contacts are available and send Instant Messages. PortGo features an IM interface which focuses on your contacts and friends.

[[PortGo|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==QuteCom==

[[File:Qutecom3.png|left|200px|QuteCom]]

'''Product:''' QuteCom

'''Company:''' QuteCom

'''OS:''' Windows,MacOS,Linux

'''Overview:'''
QuteCom is a community of enthusiasts and developers, creating free software products related to communication over IP. The flagship product of the QuteCom project is a softphone which allows you to make free PC to PC video and voice calls, and to integrate all your IM contacts in one place.

[[QuteCom|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==SIP Droid==

[[File:Sipdroid_00.png|200px|thumb|left|SIP Droid]]

'''Product:''' SIP Droid

'''Company:''' SIP Droid

'''OS:''' Android

'''App Storage Consumption:''' 1.85mb (ability to move app to SD card. Data uses an additional ~4kb approx.)

'''Overview:''' SIPDroid is a java based, open source SIP client that has recently been developed for use with mobile devices based on Google’s Android platform.

[[SIP_Droid|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==SJPhone==

[[File:Sjphone0.png|200px|thumb|left|SJPhone]]

'''Product:''' SJPhone

'''Company:''' SJ Labs, Inc.

'''OS:''' Windows,Windows Mobile

'''Overview:'''
SJphone is a VOIP softphone that allows you to speak with any other softphone, any stand-alone IP-phone, or using ITSP with any traditional wired or mobile phone.

It supports both SIP and H.323 standards and is fully inter-operable with most major VOIP vendors and ITSP.

[[SJPhone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Taki==

[[File:blackberry_z30_taki_oncall.png|200px|thumb|left|Taki]]

'''Product:''' Taki

'''Company:''' Taki

'''OS:''' BlackBerry 10, BlackBerry PlayBook

'''Project site:''' http://taki.sourceforge.net/

'''App in the AppWorld:''' http://appworld.blackberry.com/webstore/content/130295

'''Overview:''' Taki is a native SIP softphone for BlackBerry® PlayBook™ and
BlackBerry® 10 platforms.

'''Main Features:'''
- Multiple SIP accounts support
- Multiple simultaneous calls
- Call recording
- Conference calls
- Advanced call control: Transfer, Hold, Mute, Reject, Redial, switch
between multiple active calls
- Speaker phone support
- Address book. Integration with local contacts on BlackBerry 10 phones
- Comprehensive SIP settings (port binding, outbound proxy, registrar,
codecs priority, NAT traversal, STUN, Presence, and more)
- Transports: UDP/TCP
- Audio codecs: G.711a/u, GSM, iLBC, G.722 (HD), Speex (NB, WB, UWB)
- Built for BlackBerry

[[Taki|See Configuration Details]]

<div style="width:100%;overflow:hidden;clear:both"></div>

==Telephone==

[[File:telephone-user-interface.png|200px|thumb|left|Telephone]]

'''Product:''' Telephone

'''Company:''' 64 Characters

'''OS:''' OS X 10.9 or later

'''Project site:''' http://www.64characters.com/telephone/

'''Mac App Store:''' https://itunes.apple.com/us/app/telephone/id406825478

'''Overview:''' Telephone is a fully-featured SIP client with a minimalist user interface.

<br>

'''Main Features:'''

* Multiple SIP accounts
* Multiple simultaneous calls w/conferencing
* Call control: Transfer, Hold, Mute, Reject, Redial, Call Waiting, multiple active calls
* Speaker phone/headset support (via Mac sound device)
* Integration with Contacts app, dial by name, autocomplete, click to dial
* Send DTMF tones
* Languages: English, German, Russian

[[Telephone|See Configuration Details]]

<div style="width:100%;overflow:hidden;clear:both"></div>

==Vimphone==

[[File:Vimphone0.png|200px|thumb|left|Vimphone]]

'''Product:''' Vimphone

'''Company:''' Vimtura

'''OS:''' Android

'''Overview:''' Vimphone is a free app (GPL) which allows Android users to make inexpensive calls using VoIP.ms (or any SIP-compliant provider) and a wi-fi connection.

[[Vimphone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==X-Lite==

[[File:Xlite_00.png|200px|thumb|left|X-Lite]]

'''Product:''' X-Lite

'''Company:''' CounterPath

'''OS:''' Windows, MacOS, Linux

'''Overview:''' CounterPath's X-Lite helps you seamlessly transition from a traditional phone environment into the world of Voice over IP.

The latest release of X-Lite provides a completely redesigned interface that allows for a contact-centric or dialpad-centric user experience, or a combination of the two. It also provides you with some of the most popular features of our fully loaded Bria and eyeBeam softphones so you can take them for a test drive before you make your purchase.

Having a simple voice conversation, you’ll soon see why having a softphone on your desktop or laptop is the ultimate communications experience.

[[X-Lite|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==YakaPhone==

[[File:Yakaphone.png|200px|thumb|left|YakaPhone]]

'''Product:''' YakaPhone

'''Company:''' Bits Valley Ltd

'''OS:''' Windows, MacOS, Linux

'''Overview:''' Yaka Software focuses on providing solutions based on established open source technologies within the Voice-Over-IP sphere. Their flagship products are the customizable IAX-based softphone.It is Open Source and may be modified but any pre or post modification commercial usage requires their prior authorization.

[[YakaPhone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Zoiper Classic==

[[File:Zoiper0.png|300px|thumb|left|Zoiper Classic]]

'''Product:''' Zoiper Classic

'''Company:''' Zoiper

'''OS:''' Windows, Mac OS, Android, iOS

'''Overview:''' IAX & SIP multilanguage softphone is a VoIP soft client, meant to work with any IP-based communications systems and infrastructure.

Zoiper is available for Windows, Mac OS X and Linux and supports the following languages: English, German, Spanish, French, Dutch, Portuguese, Russian, Chinese, Japanese, Italian, Polish, Magyar.

*[[Zoiper_Classic|See Configuration Details for PC/MAC]]<br />
*[[ZoIPer_for_Android|See Configuration Details for ANDROID]]<br/>
*[[ZoIPer_for_iOS|See Configuration Details for iOS]]<br/>
<div style="width:100%;overflow:hidden;clear:both"></div>
[[category:softphones]]

==Zoiper Communicator==

[[File:Zoiperco1.png|200px|thumb|left|Zoiper Communicator]]

'''Product:''' Zoiper Communicator

'''Company:''' Zoiper

'''OS:''' Windows, MacOS, Linux, Solaris

'''Overview:''' Zoiper Communicator IAX & SIP softphone is a converged Internet communication tool combining high-quality voice and video calls, fax, instant messaging and presence through a contact-centric intuitive interface.

[[Zoiper_Communicator|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

OBi 100/110 & OBi 200

2017-03-03T15:28:07Z

Rod: /* General Section */

[[File:OBi110-ATA.jpg|none|200px|center]]

<br/>''The OBi100 & OBi200 are single phone port ATA adapters that support SIP VoIP services. The OBi100 & OBi200 are perfect for customers who do not have a traditional phone service, yet need a similar solution and want the savings and simplicity of using a VoIP service for all their calls. To start configuring your OBi100 or OBi200 device you will need to plug it in to your router/modem via its Internet port with an Ethernet cable and connect a regular handset phone to it's Phone port, then follow the next steps.''<br/>
<br/>
== Manual Configuration Details ==
<br/>
Start by dialing ''' * * * ''' from the connected phone, then press '''1''' to confirm your choice, this will return the IP address of your device being a number similar to '''192.168.xxx.xxx'''.<br/>
Once you get the IP address, enter it in the URL address bar '''"http://"''' of your Internet Browser to get access to the Graphic User Interface of the OBi100.

For an OBi202 please do the following to enable the GUI Web Interface:

Dial *** from the phone connected to the OBi202
Enter 0 For Advanced
Enter 30# Check Mark from
Press 1 to Enter a New Value
Press 1# to Enable
Press 1 to Save
Hang up

If done properly, the following window should appear on your screen:
[[File:ObiLogin.png|300px|thumb|left|Authentication Window - click to enlarge]]
<br clear="all" />

Once you get the Authentication Required window you will be prompted for a User Name and Password, the default credentials are:

'''User Name:''' admin

'''Password:''' admin

After this, you should now be able to see the OBi Web interface.

Now on the left side of the screen please find the next options and follow the next steps:

===Disabling auto-provisioning===

'''**NOTE :''' You may use this guide to configure an OBi110 as well. This is the VoIP.ms recommended configuration versus using the Obihai configuration dashboard (more on this later on this page) and you may also not find all new VoIP.ms servers on the Obihai Dahsboard. In order to make sure there will be no conflicts between this Manual configuration and the Obihai dashboard, please perform the following steps to disable auto-provisioning:

*System Management -> Auto Provisioning -> Auto Firmware Update -> Method : Disabled
*System Management -> Auto Provisioning -> ITSP Provisioning -> Method : Disabled
*System Management -> Auto Provisioning -> OBiTALK Provisioning -> Method : Disabled
*Voice Services -> OBiTALK Service -> Enable : Unchecked

Please note you must remove the check mark from the "default" column, then under "Method" please use the ''''Drop Down Selection'''' and choose '''Disabled'''.

[[File:Step1.png|450px|thumb|left|Disabling Auto Provision - click to enlarge]]
<br clear="all" />
After this, save all changes and you are ready to move on to the actual configuration.
<br />
===Configuring the ITSP Profile===

====General Section====
In this section you will set the name and the DigiMap you will use in the profile you configure. By default you will configure the profile A, unless you use the same device with another provider.

:'''Name''': 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)<br/>
:'''DigitMap''': Copy the line, including parenthesis, in the Digitmap field in the ITSP Profile and replace the "555" digits in the following lines by the area code of your choice:

::Dial Plan (recommended):
(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|911|011xx.|xx.|*xx.|***xxx|4xxx|(Mipd)|[^*]@@.)

Also, some clients have been successful by dialing '''**1 and *97''' for line 1 or '''**2 and *97''' for line 2.

:*If you need to set the dial plan back to Default, you can use this:

(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|011xx.|***xxx|xx.|(Mipd)|[^*]@@.)

[[File:Step2.png|550px|thumb|left|ITSP profile, General - click to enlarge]]
<br clear="all" />

====SIP Section====
In this section you can set the server and the port that you wish to register to.

Please note that in order to change the settings, you need to uncheck the Default box on the right hand side.

*ProxyServer: denver.voip.ms (one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location.)
*ProxyServerPort: 5060
*RegistrarServer: denver.voip.ms (one of VoIP.ms multiple servers [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location.)
*RegistrarServerPort: 5060

[[File:Step3.png|550px|thumb|left|ITSP profile, SIP - click to enlarge]]
<br clear="all" />

Additionally, you may want to change the RegisterExpires value to 300, scroll down, deselect the default box and set the value there from 3600 to 300.

[[File:Step4.png|550px|thumb|left|ITSP profile, SIP (Register Expires)- click to enlarge]]
<br clear="all" />

===Configuring Voice Services===
<br/>
In this section you can set your Main account/sub_account credentials like User name and Password. The Main account password by default is the same password as the Customer Portal.

*AuthUserName: 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)
*AuthPassword: ****** (''Your SIP Account Password'')

[[File:Step5.png|550px|thumb|left|Voice Services (SIP Credentials) - click to enlarge]]
<br clear="all" />

Once you have finished changing all those settings, click on the button ''Submit'' to save the changes and ''reboot your OBi device'', your device should now be registered.

== Configuration Using OBi Dashboard ==

Besides the Manual Configuration previously explained, Obihai also provides us with their own API dashboard where you can add your device, to complete the configuration in easy steps.
Add your device to the OBiTALK service in the OBi Dashboard [http://www.obitalk.com/obinet/]. Instructions for this are included with the OBi110 and are not discussed here.

After the OBi110 is added, edit the device. You can select '''Service Provider 1''' or '''Service Provider 2''' under the '''Configure Voice Services''' heading. This will take you to a page where you can select ''voip.ms''. Follow the instructions and once you are done the configuration will be downloaded to your Obi110.

== Features Star Codes ==
Please check this link to the Star Codes that are available to activate and deactivate some of the features on your device [http://www.obihai.com/docs/OBiFeatureStarCodes.pdf OBI feature star codes]
<br/>

==Known Issues and Resolutions==
=== Ghost Calls ===
<br/>
Some customers have reported receiving constant calls from "100" or "101" as callerID. These calls are going directly to your device and do not pass through our servers, so we cannot filter them. You can read more about these calls, including how to prevent them, here: [[Sip Scanner Ghost Calls]]

You can also follow these suggestions, specific for the OBi brand devices:
*You can just disable (by unchecking Enable) for SP2 and OBiTALK under your Voice Tab (If you are using our service as SP1).

*You can restrict which IP addresses that can connect to your OBi. Going to "Service Providers -> ITSP Profile A -> SIP -> X_AccessList" : voip.ms_ip_address. You can see the IP address of the server you are currently using from this link: [http://wiki.voip.ms/article/Choosing_Server#IPs Server's IPs]

*You can also change your Obi Firewall Setting X_InboundCallRoute to : {(?|x|xx|xxx|xxxx|xxxxx|xxxxxx|un@@.|anon@@.):}, ph
This will only allow 7 digit or greater numbers through.

*Another alternative: OBi Interface>> Voice Services>> SP1 Service>> X_InboundCallRoute: {>('Insert your AuthUserName here'):ph}, example:

{>('100000'):ph} where 100000 is replaced with your own six digit SIP account UserID or the sub-account registered with your device.

By default, OBi devices accept calls destined for any username. The above syntax rejects calls that are not intended for whatever you have configured as AuthUserName.

<br/>

===10 Second Delay Reaching voip.ms Voicemail Attendant when dialing *97 or *98===

The Obi 100, 110 and 202 devices have non-configurable 'short' and 'long' delays if a dialed sequence does not match a digitmap. So you may have a 10 second delay when you dial into your voip.ms voicemail because of the built-in 'long' delay. This can be resolved in a couple of ways. Simply dial a # sign after you dial *97 or *98. Or include literals in your digitmap under the Service Provider / ITSP profile A or B / General / digitmap. Here is an example digitmap with a *97 literal included:

(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|*xx.|'*97'|(Mipd)|[^*#]@@.)

The literal in the example is '*97'. You could also add a literal for '*98'.

Then when you dial *97, the device immediately sends it instead of waiting 10 seconds.

Read more on digitmaps under the topic Digit Map Configuration in the Obi Device Admin Guide.

=== Call Drops ===

If you experience random call drops while in the middle of a call or if the person you talk to remains silent for over a minute (60 seconds by default), OBi will hang up the call. Please go here and check and increase the following setting (Physical interface -> LINE port -> DetectFarEndLongSilence / SilenceTimeThreshold)

=== Enable Message Waiting Indicator MWI ===

To enable MWI please refer to the following section of the OBI web page:

Voice Services -> SP1 Service -> Calling Features ->

'''MWIEnable''' - Uncheck the box at the far right, to be able to check the box at the left, this enables the option.

'''X_VMWIEnable''' - Uncheck the box at the far right, to be able to check the box at the left, this enables the option.

'''MessageWaiting''' - Mark the checkbox

After these steps, the MWI should be active and working.

'''If you are trying to place an outbound call and get a recorded message ¨There is no service to complete your call¨ Please do the following to resolve this.'''
In Your OBi Device please go to Physical Interfaces >> PHONE Port which by default it is PSTN and it needs to be changed to Trunk Group 1

=== Using the OBi Network ===

You can use your OBi device to make calls directly to other OBi devices "''The OBi comes out of the box ready to make FREE calls to other OBi endpoints using the OBiTALK network. Dialing **9 + obi account number will use the OBiTALK feature and does not place calls to regular numbers nor use our network. ''" (you can get more information about [http://www.obihai.com/features-and-set-up here]), be aware that those calls will not pass through our network. If you need assistance with that feature, please contact [http://www.obihai.com/request-support OBI's support].
<br/>
=== Can not dial *98 even if is on your DigiMap ===
By default your OBi device uses *98 as Blind transfer code. If you want to be able to dial *98 from your device, you should change this code. You can achieve this in the settings of your device at: ''Star Codes >> Star code profile (A/B)'', unmark the "default" box and change *98 for something else (like *99)

[[File:Step6.png|550px|thumb|left|Changing *98 default code - click to enlarge]]
<br clear="all" />

=== Using the Phone book of your customer portal ===

If you plan on using the Phone Book in your Customer Portal and Speed Dial *75. Please log into your OBi and change the built-in speed dial code from *75 in the device to something else.

=== An additional note regarding outgoing calls===

In at least one instance it was necessary to specify a non-default outbound calling route in the OBi110 to be able to place calls using the voip.ms service. The default setting had the OBi110 attempting to place calls using the PSTN port on the device. The relevant setting is:

'''Physical Interfaces >> PHONE Port '''
*PrimaryLine: (Select from drop-down)

[[File:ObiPhoneport.JPG|550px|thumb|left|Changing Phone Port - click to enlarge]]
<br clear="all" />

The default is PSTN. Select SP1 Service if you only have one SIP account configured on the device. Select Trunk Group 1 to have it attempt to place calls using SP1 first, then SP2. Additional Trunk groups can be configured under Voice Services >> Gateways and Trunk Groups.

Portions of this article have been taken from [http://www.toao.net/500-mangos-guide-to-configuring-an-obi100-obi110-and-obi202-ata Mango's Guide to Configuring an OBi ATA]. Used with permission.

[[Category:Analog Telephone Adapters]]

Reseller Basic Guide

2017-02-27T16:37:57Z

Rod: /* Manage Rates and Packages. */

The Reseller feature is intended for users who want to initiate a business in the VoIP world. This useful tool can help you create and manage accounts for your own clients, you will be able to assign specific rates for them and also provide credentials to register a DID number. You will be able to handle the billing for your customers depending on your needs and how much you want to earn with your business, setting rates and adding credits for them. Additionally we will provide you with information about how to customize your Reseller portal to have a more personalized site.

The Reports section will allow you to check the Call detail records from the customers and review the Financial Reports related to their accounts. To activate your Reseller section, you need first to login to the Customer Portal then go to Reseller >> Reseller Main >> Activate

This guide will focus on the basic information to run the Reseller section properly. Here you can learn how to:

'''1. Manage Rates and Packages'''

'''2. Manage your clients'''

'''3. Use the client branding and access'''

'''4. Generate Call and Profit Reports'''

== Manage Rates and Packages. ==

This section will focus on how to create packages for the reseller feature, how to edit the rates depending on the destination and how to set a benefit for the calls your clients make. It is important to note that all the calls your customers make are charged to you as per our regulars fees, however on the reseller portal, they will see the charges according to the rates you set here.

=== Create a new package ===

From the Customer Portal go to Reseller >> Manage Rates and Packages >> Create a new Package.

[[File:CreatePackage2.png|right|650px]]

'''Package name''': Set a package name.

'''Outbound Markup''': Here you set the benefit you want to have on the calls that your clients make, you can set a fixed amount or a percentage, for this example if you set a 20%, your package will be set as follows ([cost rate] + $ fixed amount) + 20 %.

Example : for 1 cent per minute, the sell rate will (be 0.01 + 0.0000) + 20 % = 0.012

'''Pulse''': This is the charge increments that you will use for this package. For example, a pulse of 60 will make calls charged by the minute. You can read more about this in our [[Calls Cost]] article.

'''Canada Route & International Route''': These are the routes that the package will use on outgoing calls. For more information between the differences between Value and Premium, please read our article [[Value vs Premium]]

'''Voip.ms to Voip.ms Calls''': This option will make it so calls between VoIP.ms DID numbers are free. Applies to calls to USA and Canada VoIP.ms Numbers.

Under the '''Fees''' Tab you will be able to enter a value that you would charge as monthly base rate and a setup rate when connecting the package to a client.

The '''Free Minutes''' Tab will let you set a total of free calls, inbound & outbound for the package. If a vlue is entered in this field, the client will have free minutes (inbound & outbound calculated) up to the number of minutes entered here. Free minutes will be calculated from the invoice date for the package and will be calculated in total rounded minutes( for example, 3 minutes 11 seconds will be calculated as 4 minutes). The destinations added to the ''Free Zones'' at Reseller > Manage Rates & Packages > Edit Rates > Manage free zone will be able to spend these Free Minutes.

The '''Reseller System Configuration''' tab offers several advanced options for the package:

* '''User can edit Sub Account''': If enabled, the end user will be able to modify details of the sub account such as Password, Device Type, Authentication Type, CallerID Override, Music On Hold, Allowed Codecs, DTMF Mode and NAT.

* '''Type of Configuration''': This option determines the configuration of the services that will be shown in the Reseller Customer Portal. If the Default Configuration is set, the Services displayed will be the ones configured at Reseller > Client Branding and Access > Services tab. If the 'Package Configuration' is set, you will be able to set them in the Services section below.

* '''View Setting''': You can set if you want to display the information in the Client Interface on ''Simple View'' or ''Standard View'' here. Simple View is a feature that allows you to show your voipinterface.voip.ms client inferface in a simpler way to your clients. This hides some features like :
**Call Types (busy, answered, failed & noanswer) filters in the CDR
**This month and today`s amount spent on the Balance page
**Services specific information (Register type, Allowed codecs, etc) on the services page
**And some other information that would not be useful for "vonage style" clients

* '''Services''': In this section you can select which services will be available and which services will not be available for your customers, if you are using the "Package Configuration" option in the Type of Configuration section. (These Options are only available in the New Version of the reseller system)

=== Edit Rates ===

If you want to set different rates depending on the destination of the call, you can do this on the Edit Rates section from the package, all the custom rates you enter there will override the global markup. You can set a custom rate for all the calls to USA or Canada, by country or Calling code. You can also set your '''Free Zones''' here, which will be the destinations that will be included in the Free Minutes offered in the package.

[[File:EditRatesPackage.png|center|thumb|600px]]
<br clear="all" />

== Manage your clients ==

Now that you have a package, you need to add a client, this section will focus on how to create a client and associate it with a [http://wiki.voip.ms/article/Sub_Accounts sub account], as well as adding payments and charges for each client, assigning a DID number and also the proper way to delete a client account that is not longer needed.

[[File:ManageClientsAccounts.png|center|thumb|800px]]
<br clear="all" />

=== Adding a client ===

From the customer portal go to Reseller menu >> Manage Clients acct >> Create a new client account.

[[File:CreateClient.png|center|600px|thumb]]
<br clear="all" />

Fill the form with the client information and choose between soft and hard balance, when finished click on create account. A customer in HARD balance will be suspended if he is on zero or negative balance, while a customer in SOFT balance will be able to keep making use of the service even if their balance is zero or negative.

On the Manage section, you can add a new package or a DID number. To add numbers to your clients you need to order the number from your Customer Portal first. Don't forget to set a Per Minute rate and/or Monthly fee for your number, as incoming calls are charged independently from the outgoing calls.

Once the client account has been created, you will need to create a [http://wiki.voip.ms/article/Sub_Accounts sub account] for it as well.

Go to Sub accounts menu >> Create Sub account. During the Sub Account creation process, you will reach the Reseller configuration section, here you can choose:

[[File:ResellerConfiguration.png|right]]

*Reseller client

*Reseller client Package

*Next Billing date

*And if you decide to charge the set up fees now, make sure to check the appropriate box.

If you already have a [http://wiki.voip.ms/article/Sub_Accounts sub account] created, you can modify it at Sub Accounts >> Edit Sub Account >> Manage and add a package there.

=== Delete a client ===

You can delete a client on the Edit section from the Client's account, however note that before doing this you need to be sure to unconnect any package or DID number related to the client, you can do this from the Manage section on Client´s account.

=== Adding Payments manually ===

You can add payments manually from the customer portal >> Reseller menu >> Manage Clients acct >> Manage Client, then select the Payment button under the Balance panel.

=== Add Charges manually ===

You can add charges manually from the customer portal >> Reseller menu >> Manage Clients acct >> Manage Client, then select the Charge button under the Balance panel.

=== Add phone numbers to your clients (DIDs) ===

You can add phone numbers to your clients from the Customer portal >> Reseller menu >> Manage Clients acct >> Manage Client >> Package for this clients >> Add Phone number. Please note that you need to order a DID and have it on your account before you can assign it to a client. Don't forget to set a Per Minute rate and/or Monthly fee for your number, as incoming calls are charged independently from the outgoing calls.

[[File:AddPhone.png|center|thumb|600px]]
<br clear="all" />

== Use the client branding and access ==

It is always important to show a good brand image to your customer, for that reason we offer the option to customize our reseller interface to have the best presentation possible. There is also a useful feature on the site that can allow us to integrate your client's payments with a paypal account, so they can be added automatically as funds to the client.

=== Branded Interface ===

At the top of the screen you will see two available options, these are:

'''Version of the reseller system:''' This is the version of the Reseller System that will be displayed to your clients. The New Version of the Reseller System offers several customization options as well access to most of the VoIP.ms services, while the Old Version offers limited customization and access to basic services such as direct SIP routing and [[Call Forwarding]]

'''Configure Reseller Version:''' Determines which version of the Reseller System you will be configuring on this section. Enabling the New Version will offer you the option to modify the color of the panels, buttons and background of your Client Portal, as well as previewing it before making any changes.

=== Interface and Access ===

'''Clients Interface Access:''' This section will list the URL that you can give your clients to access their Account.

'''Auto Connect Form for Clients:''' This is an HTML form that you can use to automatically send your clients to their interface without needing authorization.

'''DID Access:''' You can set the pricing here for any DIDs that your clients purchase using the option at Services > Order DIDs on their Client Portal, if you have enabled this service for them.

=== Branding Management ===

In this section you will be able to change the Title that will appear in your Reseller Customer Portal page, as well as any of the colors of the interface, including the buttons* and the panels*, as well as selecting a Header, logo* and login background* for the interface.

<nowiki>*Options only available in the New Version of the reseller system.</nowiki>

=== Display Elements ===

Here you can choose whether or not you'd like to display the SIP and IAX password(s) in the Reseller Interface, as well as allowing your customers the option to reset their passwords in the Login Reseller Interface.

Lastly, you will be presented with an option to integrate an automated signup form into your website. If so, you can link your signup menu option to the URL presented there. You can also select which fields will be shown in order to sign up with a new account with you.

=== Services ===

View Setting: You can set if you want to display the information in the Client Interface on 'Simple View' here. Simple view is a feature that allows you to show your voipinterface.voip.ms client inferface in a simpler way to your clients. This hides some features like :

*Call Types (busy, answered, failed & noanswer) filters in the CDR
*This month and today`s amount spent on the Balance page
*Services specific information (Register type, Allowed codecs, etc) on the services page
*And some other information that would not be useful for "vonage style" clients

Services*: In this section you can select which services will be available and which services will not be available for your customers, if you are using the "Default Configuration" option in the Type of Configuration section of the Package settings.

<nowiki>*Options only available in the New Version of the reseller system</nowiki>

=== Text and Messages ===

In this tab you will be able to personalize the Welcome Text that will appear in the Reseller Customer Portal. You are also able to customize the Emails that will be sent when a Customer's Balance is running low, as well as when they request and complete a Password Reset.

== Payment ==

If you want your clients to deposit directly in your paypal account, you can activate the paypal integration. All payments will be sent to your paypal account and automatically credited in your client interface.

[[File:PaymentIntegration.png|center|thumb|600px]]
<br clear="all" />

Instructions :
#Enter your paypal email address in the "paypal email account" field
#Enter the amounts that you want to offer, without decimal and separated by a coma
#In your paypal account, go to "Profile" , and under the "Selling preference" section, activate the Instant payment notification(IPN). Enter the url : http://www.voipinterface.net/paynotify.php .
#We strongly recommend to make a live test with an amount of $1.

== Host Name ==

In order to personalize the interface with a hostname

#Configure a host with an A RECORD in your dns that points to 174.120.50.58
#Enter in the field bellow the fully qualified host + domain name (e.g: interface.mydomain.com)
#Click "submit"

There is a one time fee of $10.00 for adding your hostname to our system.

== Reports ==

Here you can see the Call Detail Records for the calls your customers have made and received, with the rates we have established on the packages created, showing a net total of the benefit earned as well.

Reports are divided in the following categories:

*'''Global CDR''': Display cdr report for all your reseller clients calls. You can see all the calls (incoming and outgoing) from all your clients.

*'''Per User CDR''': Display cdr report for a specific client. This filter help you to search calls only for one specific client.

*'''Financial report''': Month to month revenues ,deposits, costs related to the reseller clients. You can see the financial information by month.

*'''Sub accounts report''': List of all sub accounts connected to the reseller clients ,with related information. Here you will find a list of your clients, [http://wiki.voip.ms/article/Sub_Accounts sub account], package and DID number associated with each one.

*'''DID(s) Report''': List of all DID's connected to the reseller clients, with related information.

[[category:guides]]

Grandstream HandyTone 702 - HT702

2017-02-22T18:33:14Z

Rod: /* Configuring the Handytone 702 */

== GENERAL INFORMATION ==

The Grandstream HandyTone 702/7XX series is the latest in the HandyTone line. The 702 is a reliable, inexpensive telephone adapter which works with the VoIP.ms service when placed after your broadband internet router.

'''Websites:'''
[http://www.grandstream.com/index.php/products/ip-voice-telephony/consumer-analog-telephone-adaptors/ht702_704 HT702 Product Page]

'''Help / Support:'''
[http://www.grandstream.com/support/firmware Grandstream Support]

[[File:Ht702.jpg‎]]

== Configuring the Handytone 702 ==

These instructions are based on HandyTone 702 software version 1.0.0.18; if you are running a different software version some menus and settings may be different.
<br>
These instructions are also based on using the HandyTone in its factory default configuration, which obtains a dynamic IP address automatically from your router using DHCP. For information on configuring your HandyTone with a Static IP Address, please refer to the HandyTone user´s manual.

Each step is important in assuring that your device works properly.

:''We recommend that you read each step through in its entirety before performing the action indicated in the step.''

'''Step 1'''<br>

Connect your HandyTone to your router with the supplied Ethernet network cable.

Now connect your phone to the HandyTone. Plugging the cable into the correct FXS Port that you configure.

Finally plug the supplied power cable into the HandyTone.
<br>

'''Step 2'''<br>

: Wait 60 seconds after plugging your HT702 in.

Pick up the phone connected to the HT702 and dial *** on it.

Please have a pen and paper ready. You will hear a message - "Enter a menu option", then enter 0 2 on your phone. You will now hear a message giving you the IP address of your HT702 such as - "192.168.001.010" and write this number down.

Open a web browser on your computer such as Chrome or Firefox and enter the IP address you heard in step 4 as the address (I.E. where you would normally enter www.voip.ms).

Please note: Some browsers will require you to remove leading zero's ( 0 's ) in the IP address. For example if you heard "192.168.001.010" you should change this to "192.168.1.10".

The Interface has a timeout so please make changes quickly or save/update your settings every couple of minutes.
<br>

'''Step 3'''<br>

: You should now see a page that looks like this:<br>
[[File:Grandstream_702_01.png‎]]

Enter the password for the HT702 in the password field. The default administrator password for the HT702 is "admin" (without quotes).

After entering the password you should see a screen that looks similar to the one below:

[[File:Grandstream_702_02.png‎]]
<br>

'''Step 4'''<br>

Now, click on FXS PORT1 and configure your settings accordingly (as shown below):

Please use the same server in the Failover SIP Server as your Primary SIP Server or leave the Failover SIP Server field Blank.

[[File:GrandstreamConfigPage.png‎]]

Please use the following settings to configure your VoIP.ms account:
'''
Configuration Page Settings:'''

Primary SIP Server: servername.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server VoIP Servers])

* You can also find this information by logging into your [https://www.voip.ms/m/accountinfo.php Customer portal].

Failover SIP Server: (Please leave this Blank)

Outbound Proxy: servername.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server VoIP Servers])
* You can also find this information by logging into your [https://www.voip.ms/m/accountinfo.php Customer portal].

NAT Traversal (STUN): No, but send keep-alive

SIP User ID: (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)

Authenticate ID: (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)

Authenticate Password: ****** (Use the SIP account password - By default this is the same as the Customer Portal)

DNS Mode: A Record

SIP Registration: Yes

Unregister On Reboot: Yes

Outgoing Call Without Registration: Yes

Register Expiration: 3

Allow Incoming SIP Messages from SIP Proxy Only: Yes

Preferred DTMF method: In-audio, RFC2833

Enable Call Features: No

Dial Plan: {[x*]+}

Preferred Vocoder: PCMU, G729, PCMA
<br>

'''Step 5'''<br>

: Once you have configured the settings above, click the Update button and then the Reboot button to save the configurations.

Your HT702 will power cycle after you click the reboot button. Please wait at least 30 seconds for the unit to finish power cycling. If you see that the Phone 1 LED (or phone 2 LED, depending on which FXS port you've configured our service for) is a solid green color, then your unit is configured and ready to make calls.
<br>

'''Step 6'''<br>

: That's it! You can now make a phone call.

1 + the area code and number for calls to the US & Canada

Or

011 + the country code, area code, and number for calls worldwide (you may also use 00 instead of 011).
<br>

== Preventing Direct IP calls like 100 & 1000 ==

To Prevent Direct IP calls to your device and only allow calls from our service please enable the following 2 options in your FXS Port Configuration Page.

'''Check SIP User ID for incoming INVITE''' - Default is No. Check the incoming SIP User ID in Request URI. If they don’t match, the call will be rejected. If this option is enabled, the device will not be able to make direct IP calls.

'''Allow Incoming SIP Messages from SIP Proxy Only''' - Default is No. Check the incoming SIP messages. If they don’t come from the SIP
proxy, they will be rejected. If this option is enabled, the device will not be able to make direct IP calls.

== Auto Provisioning ==

Some newer models of the HT702 now have Auto Provisioning and will delete the changes you make in setting up the device to use our service. Please go to your Graphical user interface and go to 'Advanced Settings' tab and look for "Firmware Upgrade and Provisioning" and disable it.

== Star codes ==
Here are the star codes available for HT701 and HT702:
* *02 Forcing a Codec (per call) *027110 (PCMU), *027111 (PCMA), *02723 (G723), *02729 (G729),*0272616 (G726-r16), *0272624 (G724-r24), *0272632 (G726-r32), *0272640 (G726-r40), *027201 (iLBC)

* *03 Disable LEC (per call) Dial “*03” + ” number ”. No dial tone is played in the middle.
* *16 Enable SRTP
* *17 Disable SRTP
* *30 Block Caller ID (for all subsequent calls)
* *31 Send Caller ID (for all subsequent calls)
* *47 Direct IP Calling. Dial “*47” + “IP address”. No dial tone is played in the middle. Detail see Direct IP Calling section on page 12.
* *50 Disable Call Waiting (for all subsequent calls)
* *51 Enable Call Waiting (for all subsequent calls)
* *67 Block Caller ID (per call). Dial “*67” + ” number ”. No dial tone is played in the middle.
* *82 Send Caller ID (per call). Dial “*82” + ” number ”. No dial tone is played in the middle.
* *69 Call Return Service: Dial *69 and the phone will dial the last incoming phone number received.
* *70 Disable Call Waiting (per call). Dial “*70” + ” number ”. No dial tone is played in the middle.
* *71 Enable Call Waiting (per call). Dial “*71” + ” number ”. No dial tone is played in the middle.
* *72 Unconditional Call Forward: Dial “*72” and then the forwarding number followed by “#”. Wait for dial tone and hang up. (dial tone indicates successful forward)
* *73 Cancel Unconditional Call Forward. To cancel “Unconditional Call Forward”, dial “*73”, wait for dial tone, then hang up.
* *74 Enable Paging Call: Dial “*74” and then the destination phone number you want to page.
* *78 Enable Do Not Disturb (DND): When enabled all incoming calls are rejected.
* *79 Disable Do Not Disturb (DND): When disabled, incoming calls are accepted.
* *87 Blind Transfer
* *90 Busy Call Forward: Dial “*90” and then the forwarding number followed by “#”. Wait for dial tone then hang up.
* *91 Cancel Busy Call Forward. To cancel “Busy Call Forward”, dial “*91”, wait for dial tone, then hang up.
* *92 Delayed Call Forward. Dial “*92” and then the forwarding number followed by “#”. Wait for dial tone then hang up
* *93 Cancel Delayed Call Forward. To cancel Delayed Call Forward, dial “*93”, wait for dial tone, then hang up. Flash/Hook Toggles between active call and incoming call (call waiting tone). If not in conversation,

*flash/hook will switch to a new channel for a new call.
*# Pressing pound sign will serve as Re-Dial key

''Keywords: Handytone 701, HT701, Grandstream''

[[Category:Analog Telephone Adapters]]

Service Cost

2017-02-14T19:18:45Z

Rod: /* From Canadian provider */

Voip.ms is basically a worldwide termination ('''outgoing calls''') and origination ('''incoming calls''') provider. In VoIP.ms' system, the inbound and outbound calls are charged separately. The service can be used for outgoing calls only, it is not necessary for you to get a DID if you do not need to receive incoming calls.

:''VoIP.ms allows you to use one of your own [[Devices]] (hardware like a Linksys ATA or software like [[X-Lite|X-lite]]).''

VoIP.ms is a prepaid service provider and you need to deposit a minimum of $25 in your account to be able to use the service. All of the charges will be deducted from that balance. Your account will not be able to receive or make calls once your balance reaches zero.

Signing up is free of charge. You can navigate around, check features, configure your device and dial Echo Test and DTMF Test numbers even without adding funds.

'''Route''' (Value/Premium) can be switched to premium by the customer in the [[Account Settings]] section of the customer portal. By default, new accounts are setup to use the value route. The Value/Premium route applies separately for Canada and International. The Premium route is the only available route for the U.S..

=== Outgoing Calls ===

'''USA Rates'''

*Premium Route: $0.0100 (1¢) per minute
*Toll-Free Numbers Value Route: Free
*Toll-Free Numbers Premium Route: $0.0106 (1.06¢) per minute
*e411: $0.99 per call. Must be activated by customer in settings

'''Canada Rates'''

*Value Route: Most Canada at $0.0052 (1/2¢)
*Premium route: <s>$0.0100 (1¢)</s> $0.009 (0.9¢) per minute
*Rates for all Canada destinations are listed in the rate section

'''International Rates'''

*List available in the [http://www.voip.ms/rates.php rates] section.
*Rate update frequency: Rare. Sometimes we'll lower or raise the rate on some destinations if needed.

'''Billing Increments'''

*USA and Canada: 6 seconds increment
*International: 6 seconds increment
*Mexico: 60 seconds initial / 60 seconds increment

Refer to [[Calls Cost]] for further explanation.

=== Incoming Calls ===

'''USA / Canada DID numbers availability'''

*48 US Continental States
*4 Canadian Provinces

====USA / Canada DID Per Minute Pricing====

*Monthly Fee:
:USA: <s>($0.99 to $1.49)</s> $0.85 to $1.25
:Canada: <s>($0.99 to $1.99)</s> $0.85 to $1.70<br/>
*Per minute inbound:
:<s>($0.01 to $0.0149)</s> $0.009 to $0.0125
*One time setup Fee: $0.40 <s>($0.50)</s>
*Billing Increment: 6 seconds
*Channels: 25
*Intended Use: Any

====USA / Canada DID Flat Monthly Pricing====

*Monthly Fee:
:USA: <s>($4.95 to $6.95</s> $4.25 to 6.00
:Canada: <s>$4.95 to $5.95)</s> $4.25 to $5.15
*Per minute inbound: free, up to 3500 minutes monthly
*One time setup Fee: $0.85 <s>($1.00)</s>
*Billing Increment: Not applicable
*Channels: 2
*Intended Use: Residential, up to 3500 INCOMING minutes per month

====Extras Features====

*[[E911]]/911 $1.50 per month
*CNAM [[Caller ID]] name Lookup - 0.8 cent per query
*Unlimited [[Voicemail]] accounts

CNAM Lookup will be charged if the call matches a North American CallerID (NPANXXXXXX) whether the call is answered or not. You are not charged for a
CNAM Lookup if the number is in your Customer Portal [[Phone book]].

==== Toll-Free DID numbers pricing ====

=====From USA provider=====

It accepts USA calls by default and it can be unlocked on request to accept incoming calls from Canada as well.
*Monthly Fee: $0.99
*Per minute inbound:
*USA / $0.019 (1.9¢)
*CANADA (On demand only, unlocked on request) / $0.08 (8¢)
*ALASKA (On demand only, unlocked on request) / $0.17
*PUERTO RICO (On demand only, unlocked on request) / $0.095
*Billing Increment: 6 seconds
*One time setup Fee: Not applicable
*Channels: 25
*A custom [[toll-free number]] carries a $15 setup charge and has an approximate delivery time from 1-5 business days.

=====From Canadian provider=====

It accepts incoming calls from both USA and Canadian callers, all at the same incoming rate.
*Monthly Fee: <s>($1.49)</s> $1.25
*Per minute inbound:
:Canada: <s>$0.032 (3.2¢)</s> $0.027 (2.7¢)
:USA: <s>$0.032 (3.2¢)</s> $0.027 (2.7¢)
:Alaska: $0.17
:Puerto Rico: $0.095
*Billing Increment: 6 seconds
*Channels: 25
*A custom toll-free number carries a $30 setup charge.

====International DID pricing====

*Pricing varies per country, from your main portal please refer to DID Numbers >> Order DID >> International.

====International DID billing====

*Monthly Flat Rate
*Unlimited inbound calls
*Channels: 2
*Setup fee paid when DID is ordered
*Monthly fee billed on the 1st of each month

[[category:guides]]

Service Cost

2017-02-14T19:17:34Z

Rod: /* From USA provider */

Voip.ms is basically a worldwide termination ('''outgoing calls''') and origination ('''incoming calls''') provider. In VoIP.ms' system, the inbound and outbound calls are charged separately. The service can be used for outgoing calls only, it is not necessary for you to get a DID if you do not need to receive incoming calls.

:''VoIP.ms allows you to use one of your own [[Devices]] (hardware like a Linksys ATA or software like [[X-Lite|X-lite]]).''

VoIP.ms is a prepaid service provider and you need to deposit a minimum of $25 in your account to be able to use the service. All of the charges will be deducted from that balance. Your account will not be able to receive or make calls once your balance reaches zero.

Signing up is free of charge. You can navigate around, check features, configure your device and dial Echo Test and DTMF Test numbers even without adding funds.

'''Route''' (Value/Premium) can be switched to premium by the customer in the [[Account Settings]] section of the customer portal. By default, new accounts are setup to use the value route. The Value/Premium route applies separately for Canada and International. The Premium route is the only available route for the U.S..

=== Outgoing Calls ===

'''USA Rates'''

*Premium Route: $0.0100 (1¢) per minute
*Toll-Free Numbers Value Route: Free
*Toll-Free Numbers Premium Route: $0.0106 (1.06¢) per minute
*e411: $0.99 per call. Must be activated by customer in settings

'''Canada Rates'''

*Value Route: Most Canada at $0.0052 (1/2¢)
*Premium route: <s>$0.0100 (1¢)</s> $0.009 (0.9¢) per minute
*Rates for all Canada destinations are listed in the rate section

'''International Rates'''

*List available in the [http://www.voip.ms/rates.php rates] section.
*Rate update frequency: Rare. Sometimes we'll lower or raise the rate on some destinations if needed.

'''Billing Increments'''

*USA and Canada: 6 seconds increment
*International: 6 seconds increment
*Mexico: 60 seconds initial / 60 seconds increment

Refer to [[Calls Cost]] for further explanation.

=== Incoming Calls ===

'''USA / Canada DID numbers availability'''

*48 US Continental States
*4 Canadian Provinces

====USA / Canada DID Per Minute Pricing====

*Monthly Fee:
:USA: <s>($0.99 to $1.49)</s> $0.85 to $1.25
:Canada: <s>($0.99 to $1.99)</s> $0.85 to $1.70<br/>
*Per minute inbound:
:<s>($0.01 to $0.0149)</s> $0.009 to $0.0125
*One time setup Fee: $0.40 <s>($0.50)</s>
*Billing Increment: 6 seconds
*Channels: 25
*Intended Use: Any

====USA / Canada DID Flat Monthly Pricing====

*Monthly Fee:
:USA: <s>($4.95 to $6.95</s> $4.25 to 6.00
:Canada: <s>$4.95 to $5.95)</s> $4.25 to $5.15
*Per minute inbound: free, up to 3500 minutes monthly
*One time setup Fee: $0.85 <s>($1.00)</s>
*Billing Increment: Not applicable
*Channels: 2
*Intended Use: Residential, up to 3500 INCOMING minutes per month

====Extras Features====

*[[E911]]/911 $1.50 per month
*CNAM [[Caller ID]] name Lookup - 0.8 cent per query
*Unlimited [[Voicemail]] accounts

CNAM Lookup will be charged if the call matches a North American CallerID (NPANXXXXXX) whether the call is answered or not. You are not charged for a
CNAM Lookup if the number is in your Customer Portal [[Phone book]].

==== Toll-Free DID numbers pricing ====

=====From USA provider=====

It accepts USA calls by default and it can be unlocked on request to accept incoming calls from Canada as well.
*Monthly Fee: $0.99
*Per minute inbound:
*USA / $0.019 (1.9¢)
*CANADA (On demand only, unlocked on request) / $0.08 (8¢)
*ALASKA (On demand only, unlocked on request) / $0.17
*PUERTO RICO (On demand only, unlocked on request) / $0.095
*Billing Increment: 6 seconds
*One time setup Fee: Not applicable
*Channels: 25
*A custom [[toll-free number]] carries a $15 setup charge and has an approximate delivery time from 1-5 business days.

=====From Canadian provider=====

It accepts incoming calls from both USA and Canadian callers, all at the same incoming rate.
*Monthly Fee: <s>($1.49)</s> $1.25
*Per minute inbound:
:Canada: <s>$0.032 (3.2¢)</s> $0.027 (2.7¢)
:USA: <s>$0.032 (3.2¢)</s> $0.027 (2.7¢)
*Billing Increment: 6 seconds
*Channels: 25
*A custom toll-free number carries a $30 setup charge.

====International DID pricing====

*Pricing varies per country, from your main portal please refer to DID Numbers >> Order DID >> International.

====International DID billing====

*Monthly Flat Rate
*Unlimited inbound calls
*Channels: 2
*Setup fee paid when DID is ordered
*Monthly fee billed on the 1st of each month

[[category:guides]]

FreePBX / PBX in a Flash

2017-01-25T21:42:10Z

Rod: /* Outgoing Settings */

{{DISPLAYTITLE: FreePBX / PBX in a Flash}}

[https://www.freepbx.org/ '''FreePBX'''] is a web-based open source GUI (graphical user interface) that controls and manages Asterisk (PBX), an open source communication server. FreePBX is licensed under the GNU General Public License (GPL), an open source license. FreePBX can be installed manually or as part of the pre-configured FreePBX Distro that includes the system OS, Asterisk, FreePBX GUI and assorted dependencies.

==Important Security Information==

A '''critical vulnerability''' has been discovered that can affect FreePBX versions between 13.0.12 and 13.0.26. An unauthenticated remote attacker can run shell commands as the Asterisk user of any FreePBX machine with ‘Recordings’ This has been fixed in Recordings 13.0.27.

You can read more about this vulnerability including how to fix this here: http://wiki.freepbx.org/display/FOP/2016-08-09+CVE+Remote+Command+Execution+with+Privileged+Escalation

------------

We are also aware of an '''important''' and '''critical exploit''' related to all FreePBX versions prior to 12. This Zero-Day Remote Code Execution and Privilege Escalation exploit allows users to bypass authentication and gain ‘Full Administrator’ access to the FreePBX server when the ‘FreePBX ARI Framework module/Asterisk Recording Interface (ARI)’ is present on the system. This vulnerability may offer to any non authorized user full remote code execution access as the user running the Apache process. This exploit can be present also for users who have updated to version 12 from a prior version and did not remove the legacy FreePBX ARI Framework module.

Here are some recommendations for their product from the freepbx.org website for protection against this issue: http://www.freepbx.org/critical-freepbx-rce-vulnerability-all-versions/

== Creating a Trunk ==

To connect your FPBX server with ours, you need to create a '''trunk'''. To achieve this, once you are into your FBPX's GUI, follow this path: Connectivity >> Trunks >> Add SIP (chan_sip) Trunk.

[[File:FPBX_00.png|550px|thumb|left|Creating a trunk- click to enlarge]]
<br clear="all" />

Once you are there, you will see a list of options, create a '''SIP''' or '''IAX''' trunk (depending on your needs)

[[File:FPBX_01.png|550px|thumb|left|Creating a trunk- click to enlarge]]
<br clear="all" />

=== SIP Trunk ===

From here, use the following example to configure your SIP trunk:

==== General Settings ====

* Trunk name: Set your trunk name, a recommended one could be '''voipms''', remember that you can manage more than 1 DID number with the same trunk (using your inbound routes).

* Outbound CallerID: The 10 digit valid caller ID number that you will pass with this trunk for Outbound calls. This can be override from your extension's settings.

==== Dialed number Manipulation Rules ====

You can set here your outbound rules. These rules can manipulate the dialed number before sending it out this trunk. If no rule applies, the number is not changed. (Optional)

==== Outgoing Settings ====

This section is very important, below you'll find a sample, please replace the dummy information with yours and delete the comments. '''The information after the semicolon (;) is considered a comment and must be deleted for the trunk to work properly on some PBX versions.'''

canreinvite=nonat
nat=yes
context=from-trunk
host=atlanta.voip.ms ; (one of our multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closer to your location)
username=100000 ; (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
fromuser=100000 ; (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
secret=***** ; (password associated with the Main or Sub-account. '''Please avoid using the '#' character in the password as it will cause authentication issues''')
type=peer
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you purchased g.729 from Digium
trustrpid=yes
sendrpid=yes
insecure=invite
qualify=yes

==== Incoming Settings ====

Please '''delete''' the default settings you'll find here, this section must me in blank.

==== Registration ====

At this section you'll set your '''register string''', this is needed when you use "registration" as authentication method (If you use '''IP Authentication''' leave this in blank)

It is formed with your SIP username, password, server and registration port as below:

YourAccountNumber:Password@server.voip.ms:5060, for example:

100000:YourPassword@atlanta.voip.ms:5060

Finally, click on '''Submit changes''', after that you'll see a Red button in the top "'''Apply config'''", do not forget to click it to apply the changes.

[[File:FPBX_SIP_01.png|1000px|thumb|left|SIP trunk configuration- click to enlarge]]
<br clear="all" />

=== IAX2 Trunk ===

From here, use the following example to configure your IAX2 trunk:

==== General Settings ====

* Trunk name: Set your trunk name, this is different from SIP trunks and it '''must''' be '''voipms''', otherwise you could experience issues with your registration and calls.

* Outbound CallerID: The 10 digit valid caller ID number that you will pass with this trunk for Outbound calls. This can be override from your extension's settings.

==== Dialed number Manipulation Rules ====

You can set here your outbound rules. These rules can manipulate the dialed number before sending it out this trunk. If no rule applies, the number is not changed. (Optional)

==== Outgoing Settings ====

This section is very important, below you'll find a sample, please replace the dummy information with yours and delete the comments. '''The information after the semicolon (;) is considered a comment and must be deleted for the trunk to work properly on some PBX versions.'''

type=friend
username=100000 ; (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
secret=***** ; (password associated with the Main or Sub-account. '''Please avoid using the '#' character in the password as it will cause authentication issues''')
context=from-trunk
host=atlanta.voip.ms ; (one of our multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closer to your location)
disallow=all
allow=ulaw
insecure=port,invite
requirecalltoken=no
qualify=yes

==== Incoming Settings ====

Please '''delete''' the default settings you'll find here, this section must me in blank.

==== Registration ====

At this section you'll set your '''register string''', this is needed when you use "registration" as authentication method (If you use '''IP Authentication''' leave this in blank)

It is formed with your SIP username, password, server and registration port as below:

YourAccountNumber:Password@server.voip.ms:4569, for this example:

100000:YourPassword@atlanta.voip.ms:4569

Finally, click on '''Submit changes''', after that you'll see a Red button in the top "'''Apply config'''", do not forget to click it to apply the changes.

[[File:FPBX_IAX_01.png|1000px|thumb|left|IAX2 trunk configuration- click to enlarge]]
<br clear="all" />

== Outbound routes ==

Once you have your trunk configured, you will need an outbound route to make calls.

To create an outbound route go to "Connectivity" menu and then select "Outbound routes".

=== Route Settings ===

* Route Name: Name of this route. Should be used to describe what type of calls this route matches (for example, 'local' or 'longdistance').

* Route CID: If set, this will override all CIDS specified except:
**Extension/device EMERGENCY CIDs if this route is checked as an EMERGENCY Route
**Trunk CID if trunk is set to force it's CID
**Forwarded call CIDs (CF, Follow Me, Ring Groups, etc)
**Extension/User CIDs if checked

=== Dial Patterns ===

A Dial Pattern is a unique set of digits that will select this route and send the call to the designated trunks. If a dialed pattern matches this route, no subsequent routes will be tried. If Time Groups are enabled, subsequent routes will be checked for matches outside of the designated time(s).

''' Rules: '''
* X: matches any digit from 0-9
* Z: matches any digit from 2-9
* [1237-9]: matches any digit in the brackets (example: 1,2,3,7,8,9)
* '''.''' : wildcard, matches one or more dialed digits
* Prepend: Digits to prepend to a successful match. If the dialed number matches the patterns specified by the subsequent columns, then this will be prepended before sending to the trunks.
* Prefix: Prefix to remove on a successful match. The dialed number is compared to this and the subsequent columns for a match. Upon a match, this prefix is removed from the dialed number before sending it to the trunks.
* Match patterns: The dialed number will be compared against the prefix + this match pattern. Upon a match, the match pattern portion of the dialed number will be sent to the trunks.
* Caller ID: If CallerID is supplied, the dialed number will only match the prefix + match pattern if the CallerID being transmitted matches this. When extensions make outbound calls, the CallerID will be their extension number and NOT their Outbound CID. The above special matching sequences can be used for CallerID matching similar to other number matches.

Recommended Dial patterns are:
* 1NXXNXXXXXX
* NXXNXXXXXX
* 4XXX (This one to be able to test our '''echo test''' and '''DTMF test''')

=== Trunk Sequence for Matched routes ===

The Trunk Sequence controls the order of trunks that will be used when the above Dial Patterns are matched.

Select there your voip.ms' trunk.

[[File:FPBX_Outbound.png|1000px|thumb|left|Outbound route configuration - click to enlarge]]
<br clear="all" />

== Inbound routes ==

If you have DID numbers with us and route calls to your trunks, you need inbound routes to manage them. To create an inbound route, go to "Connectivity" menu, option "Inbound routes"

=== Add Incoming Route ===

*Description: Provide a meaningful description of what this incoming route is.

*DID number: Define the expected DID Number if your trunk passes DID on incoming calls. Set your '''voip.ms DID number''' with only 10 digits (Without dots, commas, spaces or the 1 in front of the number).

=== Set Destination ===

Set here the destination for your incoming calls received at the DID you configured as '''DID number''' (an extension, IVR, recording, voice mail, etc).

[[File:FPBX_Inbound.png|1000px|thumb|left|Inbound route configuration - click to enlarge]]
<br clear="all" />

Once you have finished the basic configuration of your PBX server, do not forget to click on the red button "Apply Config"

[[File:After_Changes.png|550px|thumb|left|Apply Setting's button - click to enlarge]]
<br clear="all" />

FreePBX / PBX in a Flash

2017-01-25T21:41:30Z

Rod: /* Outgoing Settings */

{{DISPLAYTITLE: FreePBX / PBX in a Flash}}

[https://www.freepbx.org/ '''FreePBX'''] is a web-based open source GUI (graphical user interface) that controls and manages Asterisk (PBX), an open source communication server. FreePBX is licensed under the GNU General Public License (GPL), an open source license. FreePBX can be installed manually or as part of the pre-configured FreePBX Distro that includes the system OS, Asterisk, FreePBX GUI and assorted dependencies.

==Important Security Information==

A '''critical vulnerability''' has been discovered that can affect FreePBX versions between 13.0.12 and 13.0.26. An unauthenticated remote attacker can run shell commands as the Asterisk user of any FreePBX machine with ‘Recordings’ This has been fixed in Recordings 13.0.27.

You can read more about this vulnerability including how to fix this here: http://wiki.freepbx.org/display/FOP/2016-08-09+CVE+Remote+Command+Execution+with+Privileged+Escalation

------------

We are also aware of an '''important''' and '''critical exploit''' related to all FreePBX versions prior to 12. This Zero-Day Remote Code Execution and Privilege Escalation exploit allows users to bypass authentication and gain ‘Full Administrator’ access to the FreePBX server when the ‘FreePBX ARI Framework module/Asterisk Recording Interface (ARI)’ is present on the system. This vulnerability may offer to any non authorized user full remote code execution access as the user running the Apache process. This exploit can be present also for users who have updated to version 12 from a prior version and did not remove the legacy FreePBX ARI Framework module.

Here are some recommendations for their product from the freepbx.org website for protection against this issue: http://www.freepbx.org/critical-freepbx-rce-vulnerability-all-versions/

== Creating a Trunk ==

To connect your FPBX server with ours, you need to create a '''trunk'''. To achieve this, once you are into your FBPX's GUI, follow this path: Connectivity >> Trunks >> Add SIP (chan_sip) Trunk.

[[File:FPBX_00.png|550px|thumb|left|Creating a trunk- click to enlarge]]
<br clear="all" />

Once you are there, you will see a list of options, create a '''SIP''' or '''IAX''' trunk (depending on your needs)

[[File:FPBX_01.png|550px|thumb|left|Creating a trunk- click to enlarge]]
<br clear="all" />

=== SIP Trunk ===

From here, use the following example to configure your SIP trunk:

==== General Settings ====

* Trunk name: Set your trunk name, a recommended one could be '''voipms''', remember that you can manage more than 1 DID number with the same trunk (using your inbound routes).

* Outbound CallerID: The 10 digit valid caller ID number that you will pass with this trunk for Outbound calls. This can be override from your extension's settings.

==== Dialed number Manipulation Rules ====

You can set here your outbound rules. These rules can manipulate the dialed number before sending it out this trunk. If no rule applies, the number is not changed. (Optional)

==== Outgoing Settings ====

This section is very important, below you'll find a sample, please replace the dummy information with yours and delete the comments. '''The information after the semicolon (;) is considered a comment and must be deleted for the trunk to work properly on some PBX versions.'''

canreinvite=nonat
nat=yes
context=from-trunk
host=atlanta.voip.ms ; (one of our multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closer to your location)
username=100000 ; (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
fromuser=100000 ; (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
secret=***** ; (password associated with the Main or Sub-account. '''Please avoid using the '#' character in the password as it will cause authentication issues''')
type=peer
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you purchased g.729 from Digium
trustrpid=yes
sendrpid=yes
insecure=invite
qualify=yes

==== Incoming Settings ====

Please '''delete''' the default settings you'll find here, this section must me in blank.

==== Registration ====

At this section you'll set your '''register string''', this is needed when you use "registration" as authentication method (If you use '''IP Authentication''' leave this in blank)

It is formed with your SIP username, password, server and registration port as below:

YourAccountNumber:Password@server.voip.ms:5060, for example:

100000:YourPassword@atlanta.voip.ms:5060

Finally, click on '''Submit changes''', after that you'll see a Red button in the top "'''Apply config'''", do not forget to click it to apply the changes.

[[File:FPBX_SIP_01.png|1000px|thumb|left|SIP trunk configuration- click to enlarge]]
<br clear="all" />

=== IAX2 Trunk ===

From here, use the following example to configure your IAX2 trunk:

==== General Settings ====

* Trunk name: Set your trunk name, this is different from SIP trunks and it '''must''' be '''voipms''', otherwise you could experience issues with your registration and calls.

* Outbound CallerID: The 10 digit valid caller ID number that you will pass with this trunk for Outbound calls. This can be override from your extension's settings.

==== Dialed number Manipulation Rules ====

You can set here your outbound rules. These rules can manipulate the dialed number before sending it out this trunk. If no rule applies, the number is not changed. (Optional)

==== Outgoing Settings ====

This section is very important, below you'll find a sample, please replace the dummy information with yours and delete the comments. '''The information after the semicolon (;) is considered a comment and must be deleted for the trunk to work properly on some PBX versions.'''

type=friend
username=100000 ; (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
secret=***** ; (password associated with the Main or Sub-account)
context=from-trunk
host=atlanta.voip.ms ; (one of our multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closer to your location)
disallow=all
allow=ulaw
insecure=port,invite
requirecalltoken=no
qualify=yes

==== Incoming Settings ====

Please '''delete''' the default settings you'll find here, this section must me in blank.

==== Registration ====

At this section you'll set your '''register string''', this is needed when you use "registration" as authentication method (If you use '''IP Authentication''' leave this in blank)

It is formed with your SIP username, password, server and registration port as below:

YourAccountNumber:Password@server.voip.ms:4569, for this example:

100000:YourPassword@atlanta.voip.ms:4569

Finally, click on '''Submit changes''', after that you'll see a Red button in the top "'''Apply config'''", do not forget to click it to apply the changes.

[[File:FPBX_IAX_01.png|1000px|thumb|left|IAX2 trunk configuration- click to enlarge]]
<br clear="all" />

== Outbound routes ==

Once you have your trunk configured, you will need an outbound route to make calls.

To create an outbound route go to "Connectivity" menu and then select "Outbound routes".

=== Route Settings ===

* Route Name: Name of this route. Should be used to describe what type of calls this route matches (for example, 'local' or 'longdistance').

* Route CID: If set, this will override all CIDS specified except:
**Extension/device EMERGENCY CIDs if this route is checked as an EMERGENCY Route
**Trunk CID if trunk is set to force it's CID
**Forwarded call CIDs (CF, Follow Me, Ring Groups, etc)
**Extension/User CIDs if checked

=== Dial Patterns ===

A Dial Pattern is a unique set of digits that will select this route and send the call to the designated trunks. If a dialed pattern matches this route, no subsequent routes will be tried. If Time Groups are enabled, subsequent routes will be checked for matches outside of the designated time(s).

''' Rules: '''
* X: matches any digit from 0-9
* Z: matches any digit from 2-9
* [1237-9]: matches any digit in the brackets (example: 1,2,3,7,8,9)
* '''.''' : wildcard, matches one or more dialed digits
* Prepend: Digits to prepend to a successful match. If the dialed number matches the patterns specified by the subsequent columns, then this will be prepended before sending to the trunks.
* Prefix: Prefix to remove on a successful match. The dialed number is compared to this and the subsequent columns for a match. Upon a match, this prefix is removed from the dialed number before sending it to the trunks.
* Match patterns: The dialed number will be compared against the prefix + this match pattern. Upon a match, the match pattern portion of the dialed number will be sent to the trunks.
* Caller ID: If CallerID is supplied, the dialed number will only match the prefix + match pattern if the CallerID being transmitted matches this. When extensions make outbound calls, the CallerID will be their extension number and NOT their Outbound CID. The above special matching sequences can be used for CallerID matching similar to other number matches.

Recommended Dial patterns are:
* 1NXXNXXXXXX
* NXXNXXXXXX
* 4XXX (This one to be able to test our '''echo test''' and '''DTMF test''')

=== Trunk Sequence for Matched routes ===

The Trunk Sequence controls the order of trunks that will be used when the above Dial Patterns are matched.

Select there your voip.ms' trunk.

[[File:FPBX_Outbound.png|1000px|thumb|left|Outbound route configuration - click to enlarge]]
<br clear="all" />

== Inbound routes ==

If you have DID numbers with us and route calls to your trunks, you need inbound routes to manage them. To create an inbound route, go to "Connectivity" menu, option "Inbound routes"

=== Add Incoming Route ===

*Description: Provide a meaningful description of what this incoming route is.

*DID number: Define the expected DID Number if your trunk passes DID on incoming calls. Set your '''voip.ms DID number''' with only 10 digits (Without dots, commas, spaces or the 1 in front of the number).

=== Set Destination ===

Set here the destination for your incoming calls received at the DID you configured as '''DID number''' (an extension, IVR, recording, voice mail, etc).

[[File:FPBX_Inbound.png|1000px|thumb|left|Inbound route configuration - click to enlarge]]
<br clear="all" />

Once you have finished the basic configuration of your PBX server, do not forget to click on the red button "Apply Config"

[[File:After_Changes.png|550px|thumb|left|Apply Setting's button - click to enlarge]]
<br clear="all" />

3CX Phone System

2016-12-20T19:37:46Z

Rod:

3CX Phone System for Windows
Version 9.

3CX Phone System is a software-based IP PBX that replaces a traditional PBX and delivers employees the ability to make, receive and transfer calls. The IP PBX supports all traditional PBX features. An IP PBX is also referred to as a VOIP Phone System, IP PBX or SIP server.

Calls are sent as data packets over the computer data network instead of via the traditional phone network. Phones share the network with computers and separate phone wiring can therefore be eliminated. With the use of a VOIP gateway, you can connect existing phone lines to the IP PBX and make and receive phone calls via a regular PSTN line. The 3CX phone system uses standard SIP software or hardware phones, and provides internal call switching, as well as outbound or inbound calling via the standard phone network or via a VOIP service.

'''NOTICE:''' The latest update of 3CX introduced a feature that might block Call Requests from Voip.ms. In order to solve this you need to go into 3CX, 3CX Management Console > Settings > Parameters > Customer Parameters tab ,
find "SEC_IGNORE_USER_AGENT" and remove "voip" from the Value Section. This only applies to 3CX V15 SP4.

==Adding VoIP.ms to 3CX==
Please make sure NAT is enabled on both your side and on the account you are using.

In the 3CX Phone System Portal, in the middle of the upper-part of the screen, we can see the '''Add Voip Provider Wizard''' , please click on the link. Once we are there we can put the '''Name of the Provider''', in this case we are going to enter''' Voip.ms''' and in the Provider list we can select '''Generic Voip Provider'''.

[[File:3CX trunk 1.jpg|1024px]]

Once we already click on continue, we will see the '''Voip Provider Details''' configure it according to the following instructions:

[[File:3CXtrunk2.jpg|1024px]]

*'''SIP server hostname or IP''': any of the 10+ VoIP.ms servers, all have the naming convention server.voip.ms (e.g: newyork.voip.ms).
*'''SIP Server port''': 5060
*'''Outbound proxy hostname or IP''': Leave in blank
*'''Outbound proxy port (default is 5060)''': 5060

For a full list of the servers and their IP's please check [http://wiki.voip.ms/article/Choosing_Server#IPs[Servers]]

Account details section, to complete this section please do it according to the following instructions:

[[File:3CXtrunk3.jpg|1024px]]

*'''External Number''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
*'''Authentication ID''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
*'''Authentication Password''': Your voip.ms password. (the password you set when you signup is the default SIP password too)
*'''Maximum simultaneous calls''': Specify how many concurrent calls your account supports.

'''Calls Routing'''

[[File:3CX trunk 4.jpg|1024px]]

Now specify how calls from this VOIP provider should be routed. You can specify a different route outside office hours.

End call
Connect to Extension
Connect to Queue
Connect to Digital Receptionist
Voicemail Box for extension
Forward to Outside Number

On the next page, you will be asked for a prefix so as to create an outbound rule for Voip.ms . Enter the dialing prefix in the “Calls to numbers starting with (prefix)” text box. To make calls via this provider, precede the number to be dialed with this prefix.

==Outbound Rules==

An outbound rule defines on which provider an outbound call should be placed,based on who is making the call, the number that is being dialed and the length of the number.

[[File:3CXOutboundRules.jpg|1024px]]

Specify for which calls to apply the outbound route. In the „Apply this rule to these calls‟ section, specify any of these options:

*'''Calls to Numbers starting with''': apply this rule to all calls starting with the number you specify. For example, specify 1 to specify that all calls starting with a 1 (usually a prefix) are outbound calls. Callers would dial „123456789‟ to reach the number „23456789‟

*'''Calls from extensions''': Select this option to define particular extensions or extension ranges for which this rule applies. Specify one or more extensions separated by commas, or specify a range using a -, for example 105-140.

*'''Calls with a Number length of''': Select this option to apply the rule to numbers with a particular digit length, for example 10 digits. This way you can capture calls to local area numbers or national numbers.

*Now specify how the outbound calls should be made. In the '''Make outbound calls on''' section, select up to 3 routes for the call. Each defined gateway or provider will be listed as a possible route. If the first route is not available or busy, 3CX Phone System will automatically try the second route.

==Inbound Rules==

With the Inbound rules we can configure calls made to that particular DID number to go to a particular extension, digital receptionist or other destination.
The latest update of 3CX introduced a feature that might block Call Requests from Voip.ms. In order to solve this you need to go into 3CX, 3CX Management Console > Settings > Parameters > Customer Parameters tab ,
find "SEC_IGNORE_USER_AGENT" and remove "voip" from the Value Section. This only applies to 3CX V15 SP4.

[[File:Create DID.jpg|1024px]]

*Click on the "Create DID‟ button in the 3CX Management Console in the toolbar.
*Enter a name for the DID (for example support).
*Now enter the DID number as it will appear in the SIP “to” header.

* Now select for which ports you wish to add this DID. If the DID number is associated with multiple ISDN ports, then you must select each. An inbound rule will be created for each port that you select.

*Now specify where you wish to direct calls made to this DID:
End Call
Connection to extension
Connect to Queue/Ring Group
Connect to Digital receptionist
Voicemail box for extension
Forward to outside number
Send fax to email of extension

You can specify that an incoming call is routed differently if it is received outside office hours. De-select the "Same as during office hours‟ option to specify a different route.

'''Click OK''' to create the DID / Inbound rule. The newly created DID‟s will be listed as inbound rules.

*There is record of an '''issue''' where you can't get incoming calls, and only outbound will work, like the issue mentioned [http://www.3cx.com/forums/cant-get-inbound-routes-from-voip-ms-did-to-work-30255.html Here]
The option '''"Source Identification by DID"''' can be found [http://www.3cx.com/blog/docs/did-voip-provider/ Here]

==Creating Extensions==

To add an extension, click on "Add Extension‟ from the toolbar.

[[File:Extensions 1.jpg|1024px]]

*'''Enter the extension number''': first and last name and the Email address (Optional) of the user. The email address will be used for voice mail notifications and as the default SIP ID. You can leave the field empty if you wish.

*'''Now specify an authentication ID and password''':

ID – The SIP "Username‟. e.g. 200.
Password –The SIP Password (password can be hidden from the user).

* '''Voicemail''': You can enable different features from your Voicemail

Enable voice mail
Play Caller ID – the voicemail system will play the number of the caller who left the voice message
Read out date/time of message – the voicemail system will play back the time of the voice message to be played
PIN number – this pin number is used to protect the voice mailbox and is used by the user to access the mailbox. The PIN number is also used
as a password to logon to 3CX Assistant or the MyPhone User portal.

* '''Click OK''' to create the extension.

After you have created the extension, a summary page will appear, which shows the information that the SIP phone will need:

* '''Proxy server IP or FQDN''': Host name of 3CX Phone System
* '''User ID''': Extension number created
* '''Authentication ID''': As specified in Authentication ID field
* '''Password''': As specified in Authentication password field

==Security Measures==

'''We strongly recommend you to change the password in your account, PBX system and extensions on it, periodically'''

'''As a preventive measure you also can disable International calls on your account'''.
'''From the customer portal >> Main Menu >> Account Settings >> Account Restrictions.
'''These settings define the restrictions the system will use when you place calls to either USA48, Canada or International Numbers.'''

''We strongly recommend to specifically select only the countries that you on your regular traffic (outgoing calls). You can do this by clicking in Currently Allowed: All Countries Allowed >> Click here to manage list of allowed countries''
[[Category:PBXes]]

3CX Phone System

2016-12-16T18:56:19Z

Rod:

3CX Phone System for Windows
Version 9.

3CX Phone System is a software-based IP PBX that replaces a traditional PBX and delivers employees the ability to make, receive and transfer calls. The IP PBX supports all traditional PBX features. An IP PBX is also referred to as a VOIP Phone System, IP PBX or SIP server.

Calls are sent as data packets over the computer data network instead of via the traditional phone network. Phones share the network with computers and separate phone wiring can therefore be eliminated. With the use of a VOIP gateway, you can connect existing phone lines to the IP PBX and make and receive phone calls via a regular PSTN line. The 3CX phone system uses standard SIP software or hardware phones, and provides internal call switching, as well as outbound or inbound calling via the standard phone network or via a VOIP service.

'''NOTICE:''' The latest update of 3CX, as of this writing, introduced a feature that might block Call Requests from Voip.ms. In order to solve this you need to go into 3CX, open system|parameters, find "SEC_IGNORE_USER_AGENT" and remove "voip"
from the Value Section. This only applies to 3CX V15 SP4.

==Adding VoIP.ms to 3CX==
Please make sure NAT is enabled on both your side and on the account you are using.

In the 3CX Phone System Portal, in the middle of the upper-part of the screen, we can see the '''Add Voip Provider Wizard''' , please click on the link. Once we are there we can put the '''Name of the Provider''', in this case we are going to enter''' Voip.ms''' and in the Provider list we can select '''Generic Voip Provider'''.

[[File:3CX trunk 1.jpg|1024px]]

Once we already click on continue, we will see the '''Voip Provider Details''' configure it according to the following instructions:

[[File:3CXtrunk2.jpg|1024px]]

*'''SIP server hostname or IP''': any of the 10+ VoIP.ms servers, all have the naming convention server.voip.ms (e.g: newyork.voip.ms).
*'''SIP Server port''': 5060
*'''Outbound proxy hostname or IP''': Leave in blank
*'''Outbound proxy port (default is 5060)''': 5060

For a full list of the servers and their IP's please check [http://wiki.voip.ms/article/Choosing_Server#IPs[Servers]]

Account details section, to complete this section please do it according to the following instructions:

[[File:3CXtrunk3.jpg|1024px]]

*'''External Number''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
*'''Authentication ID''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
*'''Authentication Password''': Your voip.ms password. (the password you set when you signup is the default SIP password too)
*'''Maximum simultaneous calls''': Specify how many concurrent calls your account supports.

'''Calls Routing'''

[[File:3CX trunk 4.jpg|1024px]]

Now specify how calls from this VOIP provider should be routed. You can specify a different route outside office hours.

End call
Connect to Extension
Connect to Queue
Connect to Digital Receptionist
Voicemail Box for extension
Forward to Outside Number

On the next page, you will be asked for a prefix so as to create an outbound rule for Voip.ms . Enter the dialing prefix in the “Calls to numbers starting with (prefix)” text box. To make calls via this provider, precede the number to be dialed with this prefix.

==Outbound Rules==

An outbound rule defines on which provider an outbound call should be placed,based on who is making the call, the number that is being dialed and the length of the number.

[[File:3CXOutboundRules.jpg|1024px]]

Specify for which calls to apply the outbound route. In the „Apply this rule to these calls‟ section, specify any of these options:

*'''Calls to Numbers starting with''': apply this rule to all calls starting with the number you specify. For example, specify 1 to specify that all calls starting with a 1 (usually a prefix) are outbound calls. Callers would dial „123456789‟ to reach the number „23456789‟

*'''Calls from extensions''': Select this option to define particular extensions or extension ranges for which this rule applies. Specify one or more extensions separated by commas, or specify a range using a -, for example 105-140.

*'''Calls with a Number length of''': Select this option to apply the rule to numbers with a particular digit length, for example 10 digits. This way you can capture calls to local area numbers or national numbers.

*Now specify how the outbound calls should be made. In the '''Make outbound calls on''' section, select up to 3 routes for the call. Each defined gateway or provider will be listed as a possible route. If the first route is not available or busy, 3CX Phone System will automatically try the second route.

==Inbound Rules==

With the Inbound rules we can configure calls made to that particular DID number to go to a particular extension, digital receptionist or other destination.
The latest update of 3CX introduced a feature that might block Call Requests from Voip.ms. In order to solve this you need to go into 3CX, open system|parameters, find "SEC_IGNORE_USER_AGENT" and remove "voip"
from the Value Section. This only applies to 3CX V15 SP4.

[[File:Create DID.jpg|1024px]]

*Click on the "Create DID‟ button in the 3CX Management Console in the toolbar.
*Enter a name for the DID (for example support).
*Now enter the DID number as it will appear in the SIP “to” header.

* Now select for which ports you wish to add this DID. If the DID number is associated with multiple ISDN ports, then you must select each. An inbound rule will be created for each port that you select.

*Now specify where you wish to direct calls made to this DID:
End Call
Connection to extension
Connect to Queue/Ring Group
Connect to Digital receptionist
Voicemail box for extension
Forward to outside number
Send fax to email of extension

You can specify that an incoming call is routed differently if it is received outside office hours. De-select the "Same as during office hours‟ option to specify a different route.

'''Click OK''' to create the DID / Inbound rule. The newly created DID‟s will be listed as inbound rules.

*There is record of an '''issue''' where you can't get incoming calls, and only outbound will work, like the issue mentioned [http://www.3cx.com/forums/cant-get-inbound-routes-from-voip-ms-did-to-work-30255.html Here]
The option '''"Source Identification by DID"''' can be found [http://www.3cx.com/blog/docs/did-voip-provider/ Here]

==Creating Extensions==

To add an extension, click on "Add Extension‟ from the toolbar.

[[File:Extensions 1.jpg|1024px]]

*'''Enter the extension number''': first and last name and the Email address (Optional) of the user. The email address will be used for voice mail notifications and as the default SIP ID. You can leave the field empty if you wish.

*'''Now specify an authentication ID and password''':

ID – The SIP "Username‟. e.g. 200.
Password –The SIP Password (password can be hidden from the user).

* '''Voicemail''': You can enable different features from your Voicemail

Enable voice mail
Play Caller ID – the voicemail system will play the number of the caller who left the voice message
Read out date/time of message – the voicemail system will play back the time of the voice message to be played
PIN number – this pin number is used to protect the voice mailbox and is used by the user to access the mailbox. The PIN number is also used
as a password to logon to 3CX Assistant or the MyPhone User portal.

* '''Click OK''' to create the extension.

After you have created the extension, a summary page will appear, which shows the information that the SIP phone will need:

* '''Proxy server IP or FQDN''': Host name of 3CX Phone System
* '''User ID''': Extension number created
* '''Authentication ID''': As specified in Authentication ID field
* '''Password''': As specified in Authentication password field

==Security Measures==

'''We strongly recommend you to change the password in your account, PBX system and extensions on it, periodically'''

'''As a preventive measure you also can disable International calls on your account'''.
'''From the customer portal >> Main Menu >> Account Settings >> Account Restrictions.
'''These settings define the restrictions the system will use when you place calls to either USA48, Canada or International Numbers.'''

''We strongly recommend to specifically select only the countries that you on your regular traffic (outgoing calls). You can do this by clicking in Currently Allowed: All Countries Allowed >> Click here to manage list of allowed countries''
[[Category:PBXes]]

SMS

2016-11-28T15:06:24Z

Rod: Updated SMS image. Unofficial apps ommited, as they are not supported by Voip.ms Tech Support.

[[File:Smslogo4.jpg|200px|left]]

Short Message Service (SMS). This feature will allow you to send and receive messages with your DID Number (US and Canada DID Numbers Only). Currently it is in beta and will remain free until further notice.

Please note that this feature is for regular customer usage. '''No automation, telemarketing, bulk sending or receiving will be allowed.'''

__TOC__

== Important information to know about the SMS Service ==

* '''The SMS Service is in BETA version, that means it is not fully deployed. It is important to us that you report any issues with this service by sending an email to support@voip.ms so that the developers can get involved if necessary.'''

* '''The SMS Service will be free until further notice.'''

* '''The SMS Service is only available for US and Canadian local DID Numbers marked with the distinctive SMS Icon.'''

* '''The SMS Service can only be used to send SMS Messages to Canadian and US 10 digit numbers at this time. We also cannot guarantee that International SMS will be properly received.'''

* '''VoIP.ms reserves the right, at its sole discretion, to limit or disable the SMS service from any account that may present marketing patterns / automation patterns / bulk sending or receiving while the service is free and in beta.'''

* '''At this time we cannot guarantee that Short Code SMS Messages, which are usually 6 digits or less (e.g.: Skype, Bank Codes, TV Commercials etc...), will work.'''

* '''We cannot guarantee that accents or special characters including non Latin letters will be properly delivered.'''

* '''For Ported In Numbers: The SMS Functionality will be available up to 48 hours after the porting process is marked as Completed.'''

* '''IMPORTANT: At this time, it is not possible to use a softphone to send or receive messages with our SMS Feature. This has to be done through the SMS Message Center in the Customer Portal at DID Numbers >> SMS Message Center in the Customer Portal or through https://sms.voip.ms

If you have further questions don't hesitate to contact the Support Staff on the Live Chat or Ticket System.

== Identifying a SMS DID Number ==

The first thing you need to know is that this feature is only available for '''local US and Canadian numbers''' at the moment. You will note that some numbers will have a little icon of a phone device, this indicates that the number supports SMS. Please note that not all locations support SMS at this time.

You can start ordering a DID Number from your portal >> DID Numbers >> Order DID >> Local Numbers >> (US or Canada) Numbers and you will be able to purchase your desired number, just make sure it supports the SMS feature.

[[File:SMS 1.png]]

== Configuring the SMS service ==

Once purchased the SMS Service needs to be activated on the DID settings, from your portal >> DID Numbers >> Manage DID >> Edit Selection >> SMS.

On that section you can activate and configure some forward options for the SMS service.

[[File:SMS_options.png]]

The first action to take is to mark the '''Short Message Service (SMS) service''', by enabling this field the SMS Service will be active and free until further notice.

* '''VoIP.ms SMS Portal''': This is enabled by default and it can't be modified, that means that you will be able to create and send messages from the SMS Message Center.

*'''SMS Email Address''': By activating this option all the SMS Messages will be sent to the email address you configure in this field. The advantage of this option is that you will receive your SMS Messages directly to your email and you will be able to Reply to these messages from your email too. You just need to click on '''reply''' to the email.

To reply the SMS Message via your email it is important to click on Reply and '''DON'T change or modify''' the destination mail "TO"
(sms@voip.ms)and the subject E.G [#USXXX] Message sent to 5555555555.

*'''SMS Forward''': You can have your SMS Messages forwarded to your cellphone or any other number that supports SMS, with this option you will receive the SMS Messages from our system to the configured phone number.

The CallerID sent to the Call Forwarded Cell Phone will be your DID Number that received the SMS Message.

*'''SMS URL Callback''': By enabling this option you will be able to send the messages to another destination e.g. your own server. If selected SMS Messages received by your DID will Send a GET request to the URL Callback provided. Available variables for your URL:

'''{ID}''' The ID of the SMS message.
'''{TIMESTAMP}''' The date and time the message was received.
'''{FROM}''' The phone number that sent you the message
'''{TO}''' The DID Number that received the message
'''{MESSAGE}''' The content of the message
'''Example''': http: //mysite.com/sms.php?to={TO}&from={FROM}&message={MESSAGE}&id={ID}&date={TIMESTAMP}

*'''URL Callback Retry''': When selected, we will be expecting an "ok" output (without quotes) from your URL callback page as an indicator that you have received the message correctly. If we don't received the "ok" (without quotes) from your callback page, we will keep sending you the same message every 30 minutes.

Once you configure the desired option, click on Apply Changes.

== Send and Receive Messages ==

===SMS Message Center===

To start using the service you will need to use the SMS Message Center, from your portal >> DID Numbers >> [https://www.voip.ms/m/sms.php SMS Message Center].

For Mobile Devices please use [https://sms.voip.ms/ sms.voip.ms] as a complete all in one solution.

From the SMS Message Center you can check your SMS History, send and receive SMS messages.

[[File:SMS.png]]

*'''Search Range''': Select a Date Range of your SMS History. You can select a time period by marking the "Show Details" field and the text messages will be displayed.

*'''Search Filter''': You can filter your Search by DID, Contact and Type (Sent, Received and Both).

*'''Send New SMS Messages''': Another window will open by clicking here in order to create a new SMS Message.

*'''Delete Selected Messages''': This action will delete any messages that you have selected, by clicking on the box next to them. '''This action cannot be undone.'''

*'''Delete All Messages''': Delete ALL your SMS Messages, both sent and received. '''This action cannot be undone.'''

*'''Checking a Message''': To check a received SMS Message just click it directly in order to display the information.

====Create a New SMS Message====

[[File:SMS41.png]]

You can create a new '''SMS Message''' we will explain you all the parameters you can find on this section:

* '''DID''': If you have more than one DID Number you can select any of the numbers available to send your message.
* '''Contact''': The destination phone number or if you have a [[Phone book|Phone book entry]] start typing the name and the system will display the information if available.
* '''Message''': You can create your message on that field, please note that you have up to 160 characters.

Then click on 'Send Message' to send the SMS.

Porting a Number

2016-11-28T14:59:40Z

Rod: Undo revision 8471 by Rod (talk)

== Port a number ==

Porting a number is the process of transfer a number from one provider to another. If you are a [http://voip.ms/ VoIP.ms] customer and have an existing number with a different provider you can optionally bring it (port it in) to [http://voip.ms/ VoIP.ms] network. Toll free numbers, local USA/Canada and even International numbers can be ported to [http://voip.ms/ VoIP.ms].

Our LNP (Local Number Portability) Department handles the porting process and guides the customer through it. They are also in charge of updating the customer via email with any relevant information about the port every time there is a change of the port request status. (Such as when the information has been submitted to the carrier, when a porting date -known as FOC- has been received, when the port has completed, etc)

== Numbers portable ==

As of today, we can port numbers to our network from the following countries:

For Geographical numbers:

Australia, Brazil, '''Canada ($8.75 USD/number)''', Chile, Croatia, Cyprus, Czech Republic, Denmark, Dominican Republic, Finland, France, Germany, Greece, Hungary, Ireland, Israel, Italy, Lithuania, Luxembourg, Mexico, Netherlands, New Zealand, Norway, Panama, Poland, Puerto Rico, Romania, Slovakia, Slovenia, South Africa, Spain, Sweden, Switzerland, '''United Kingdom ($30 USD/number)''', '''United States (Free for limited time)'''

For Toll Free numbers:

Australia, Austria, Belgium, Brazil, '''Canada ($8.75 USD/number)''', Croatia, Cyprus, Czech Republic, Denmark, Finland, France, Germany, Hungary, Ireland, Luxembourg, Netherlands, Norway, Panama, Portugal, Puerto Rico, Slovakia, Slovenia, Spain, Sweden, United Kingdom, '''United States (Free for limited time)'''

For Fax numbers:

Canada & United States. $15 USD/Number, Tollfree Numbers cannot be currently ported in as Fax Numbers.

For National numbers:

Austria, Belgium, Denmark, Estonia, Finland, France, Germany, Israel, Luxembourg, Netherlands, Norway, Panama, Portugal, Slovenia, Spain, Sweden, United Kingdom

Important: Numbers from these countries can be ported to our network even if we don't offer DID numbers from those countries. If you wish to verify
the portability of an International number please contact our LNP staff and provide the Country code, Area code and phone number, it is important
to also mention if the number is a National, Geographical or Toll-Free number.

For more information about prices you can contact [http://voip.ms/ VoIP.ms] staff, since the rate can vary depending on the country your number belongs to.

== Check Availability ==

Not all numbers from a specific country can be ported in. The most important factor to find out if a number is portable or not is the number´s city (also called ratecenter). In your customer portal, refer to DID Numbers section, you will find a link for DID Portability. In this link you can verify if your number is portable to our network, just click on the link that says "Check availability".

[[File:Port.png|850px]]

Important: For Canadian Numbers there is a one time fee of $8.75 USD per number in order to port to our network, the process can
take from 1-4 weeks, for US Numbers is Free (For limited time) and it takes from 2-4 weeks for Toll Free Canadian Carrier
the fee is $8.75 per number, Free for American Carrier (For limited time) and it takes from 2-4 weeks.

== Start Procedure ==

Once you have verified your number can be ported, on the same link (DID Portability) click on "Start procedure" you will be presented with a process of 6 steps to initiate the port of your number.

[[File:Steps.jpg]]

: Important: Make sure you have a scan of your latest invoice/statement in PDF, MS Word or as an Image file, SIGNED on a blank part.
: Do not sign it over any lines or graphics.

After following all the steps in the process and confirmed your order, you will receive an email notification from our LNP Department, that will guide you during the entire process and provide all the necessary updates. If any additional information is required they will contact you, so you can be informed all the time.

Here you can find a detailed explanation of each step you have to follow in order to start a port request:

* [[Porting a US Number]]
* [[Porting a Canadian Number]]
* [[Porting a Toll Free Number]]

== Port Rejection ==

If there is a problem with your porting process, the LNP department will notify you via email on the following steps to take to address this problem. You can read more about port rejections and the most common reasons here, [[Port Rejection]].

== Important Notes to Remember ==

Some important requirements you need to know before starting a port request:

*We are not responsible for any Port Out Fee your carrier may charge when leaving them, as it is beyond our control. Prior to starting a porting order with us, please contact the losing carrier and ask them about any port out fee they may charge you.

*Make sure to provide the exact information as it is on the losing carrier's '''Customer Service Record'''. This is usually your Service Address. Please note that the service address CAN be different from your billing address. Billing addresses will usually get rejected if it's not the same as your Service Address. If you are not sure of what is the correct service information, please contact your current service provider to obtain the Customer Service Record before proceeding.

*During a regular porting process your number must be working with the losing provider, which means you need to also make sure your number is active and not scheduled for cancellation, until the port gets completed. Disconnected numbers are not portable.

*If you decide to port in your number, you will have to make sure that your balance covers the porting fee. Otherwise, our system won't allow you to start the process until the balance is in a positive standing.

*It is the customer's responsibility to contact the losing provider '''after''' the number has been completely ported to VoIP.ms in order to have the number and/or other services disconnected from them.

*It is important to remember that, if the initial request comes back rejected due to needing additional information or some error, the process time starts again after it has been resubmitted.

*Once you receive a FOC, for a number being ported, that number is now treated as if it´s already on our network. The Number will show in your available DID list and you can configure it. You must configure the DID to be able to reach it via the VOIP.ms network.

''Canadian and USA numbers are activated on per minute plan once ported. If you would rather have it activated on flat rate, please mention it in the Notes/Comments field, on the step 5 of the port request.''

== What about Porting Out? ==

*If you wish to port out a number from the service you can do so at any time by starting a porting request with the new provider, and give your name/address as it is listed on your [http://voip.ms/ VoIP.ms] Contact Information section.

Canadian local and Toll Free port out numbers are authorized as long as the information matches with what is on your Contact Information section and you have positive balance.
Please avoid deleting the number since in most cases the number is not ported yet and if you delete it, you may lose it. Once the number is ported out, the LNP staff will remove it from your account.
We are only notified once the number has ported away so we can remove it. We have no insight to an order prior to that. Please let us know if there is a specific rejection that you have received and we may be able to help further.

This is the same for US port outs, since we are not a CLEC, we are not advised when a number is either rejected or ported out. Please open up a ticket to our LNP Department if there is a specific rejection that you have received. Also, make sure you delete the number after the new carrier confirms it has been properly ported and tested with them.

Please open up a ticket to our LNP Department if you get a rejection when porting out your number.
[[category:guides]]

Porting a Number

2016-11-28T14:59:06Z

Rod:

Choosing Server

2016-11-14T15:04:57Z

Rod: /* Server Realms */

[[File:Serverlocation3.png]]

= Choosing a Server =

[http://www.voip.ms VoIP.ms] offers many different servers, but which one should you choose? One misconception is that you should pick the closest to your location, however this is not needed most of the time. For example, if you are in the USA, any of the US servers will provide a really good latency and service quality. The newest server within a city is indicated with the highest number attached to the name, as they are classified in ascending order. Also worth noting is that there is a network tool that will help you when deciding which server you want to use, generally named a "ping", which will provide you the latency between you and the server. Therefore the server which provides you less latency should be used.

=== IPs ===

*Amsterdam, NL ('''amsterdam.voip.ms''') 37.58.88.242
*Atlanta 1, GA ('''atlanta.voip.ms''') 75.127.65.130
*Atlanta 2, GA ('''atlanta2.voip.ms''') 72.9.246.170
*Chicago 1, IL ('''chicago.voip.ms''') 208.100.39.52
*Chicago 2, IL ('''chicago2.voip.ms''') 208.100.39.53
*Chicago 3, IL ('''chicago3.voip.ms''') 208.100.39.54
*Chicago 4, IL ('''chicago4.voip.ms''') 208.100.39.55
*Dallas, TX ('''dallas.voip.ms''') 158.85.149.162
*Dallas 2, TX ('''dallas2.voip.ms''') 158.85.149.163
*Denver 1, CO ('''denver.voip.ms''') 173.248.161.90
*Denver 2, CO ('''denver2.voip.ms''') 173.248.159.210
*Houston, TX ('''houston.voip.ms''') 173.193.85.18
*Houston 2, TX ('''houston2.voip.ms''') 173.193.85.19
*London, UK ('''london.voip.ms''') 159.8.157.212
*Los Angeles 1, CA ('''losangeles.voip.ms''') 96.44.149.186
*Los Angeles 2, CA ('''losangeles2.voip.ms''') 96.44.149.202
*Melbourne, AU ('''melbourne.voip.ms''') 168.1.73.84
*Montreal 1, QC ('''montreal.voip.ms''') 67.205.74.184
*Montreal 2, QC ('''montreal2.voip.ms''') 67.205.74.187
*Montreal 3, QC ('''montreal3.voip.ms''') 192.175.96.68
*Montreal 4, QC ('''montreal4.voip.ms''') 67.205.74.179
*Montreal 5, QC ('''montreal5.voip.ms''') 192.175.96.69
*Montreal 6, QC ('''montreal6.voip.ms''') 192.175.96.70
*Montreal 7, QC ('''montreal7.voip.ms''') 192.175.96.71
*Montreal 8, QC ('''montreal8.voip.ms''') 192.175.96.72
*New York 1, NY ('''newyork.voip.ms''') 74.63.41.218
*New York 2, NY ('''newyork2.voip.ms''') 107.6.67.236
*New York 3, NY ('''newyork3.voip.ms''') 107.6.67.237
*New York 4, NY ('''newyork4.voip.ms''') 107.6.67.238
*New York 5, NY ('''newyork5.voip.ms''') 23.29.136.28
*New York 6, NY ('''newyork6.voip.ms''') 23.29.136.29
*New York 7, NY ('''newyork7.voip.ms''') 23.29.136.38
*New York 8, NY ('''newyork8.voip.ms''') 23.29.136.40
*Paris, FR ('''paris.voip.ms''') 159.8.85.180
*San Jose, CA ('''sanjose.voip.ms''') 23.246.247.146
*San Jose 2, CA ('''sanjose2.voip.ms''') 23.246.247.147
*Seattle 1, WA ('''seattle.voip.ms''') 50.23.160.50
*Seattle 2, WA ('''seattle2.voip.ms''') 50.23.160.51
*Seattle 3, WA ('''seattle3.voip.ms''') 50.23.160.52
*Tampa, FL ('''tampa.voip.ms''') 162.254.144.173
*Tampa 2, FL ('''tampa2.voip.ms''') 162.254.144.176
*Toronto 1, ON ('''toronto.voip.ms''') 158.85.70.148
*Toronto 2, ON ('''toronto2.voip.ms''') 158.85.70.149
*Toronto 3, ON ('''toronto3.voip.ms''') 158.85.70.150
*Toronto 4, ON ('''toronto4.voip.ms''') 158.85.70.151
*Toronto 5, ON ('''toronto5.voip.ms''') 184.75.215.106
*Toronto 6, ON ('''toronto6.voip.ms''') 184.75.215.114
*Toronto 7, ON ('''toronto7.voip.ms''') 184.75.215.146
*Toronto 8, ON ('''toronto8.voip.ms''') 184.75.213.210
*Vancouver 1, BC ('''vancouver.voip.ms''') 162.213.157.82
*Vancouver 2, BC ('''vancouver2.voip.ms''') 162.213.157.117
*Washington 1, DC ('''washington.voip.ms''') 208.43.234.226
*Washington 2, DC ('''washington2.voip.ms''') 208.43.234.227

===Server Realms===

For IOS, Please click here [http://wiki.voip.ms/article/Server_Realms Server Realms] to get the Realm Name for the server you plan on using, this can differ from the Domain Name being used.

Please note that the following servers will not be available to select as a DID Point of Presence for newer accounts:
Atlanta 1, Chicago 1, New York 1, New York 4, Seattle 1, Montreal 1, Montreal 2, Montreal 3, Montreal 4, Toronto, Toronto 2, Toronto 3, Toronto 4.

= What is a Ping? =

Ping is a standard tool used to test network connections. It is mostly used to determine if a server or device can be reached across the network and the latency of the response(the time it takes to send a packet to the destination and for it to return to your computer).

Ping tools are part of Windows, Mac OS X and Linux as well as some routers.

== How does the ping work? ==

It sends request messages to a target network address or DNS names at periodic intervals and measures the time it takes for a response message to arrive and return(better known as latency).

==How to ping on a PC==

Pinging is a command which tells you if the connection between your computer and a particular domain is working correctly.

In Windows, select Start > Programs > Accessories > Command Prompt. This will give you a window like the one below.

Enter the word ping, followed by a space, then the domain name.(montreal.voip.ms) in this case domain is our server name.

If the results show a series of replies, the connection is working. The time shows you how fast the connection is. If you see a "timed out" error instead of a reply, there is a breakdown somewhere between your computer and the domain.

[[File:Ping.gif]]

==How to ping on a Mac Computer==

1- Click on Finder in the dock.

2- Click on Applications.

3- Click on Utilities.

4- Double-click on Network Utility. *

* In OS X Mavericks (10.9.x) this utility app changed location. Launch it from spotlight instead, either press "command"+"space bar" or click on spotlight directly (magnifying glass icon at top right of screen), type "network utility" and hit "return"

5- In the Network Utility window, click on the Ping tab

6- In the field under "Please enter the network address to ping," like montreal.voip.ms

''If pings results are not consistent, you may have an issue with Jitter. You can work on this issue by adjusting the "Network Jitter Level" setting on your VoIP device. Usually a ping of under 150 ms is recommended in order to have good quality. The latency time to the server is important, however there are also other factors that could affect the quality of the calls such as packet loss (VoIP communications are very sensitive to this), and the Jitter level of your Internet connection.''

The following is the output of running ping with the target losangeles.voip.ms.

#ping losangeles.voip.ms
Ping to losangeles.voip.ms [67.215.241.250] with 32 bytes de datos:
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
Response from 67.215.241.250: bytes=32 time=69ms TTL=52
Response from 67.215.241.250: bytes=32 time=68ms TTL=52
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
ping statistics from 67.215.241.250:
4 packets transmitted, 4 received, 0% packet lost. rtt min/avg/max/mdev = 67ms, 69ms, 67ms

Sample ping output in windows:
C:\Windows\system32>ping montreal.voip.ms

Pinging montreal.voip.ms [67.205.74.184] with 32 bytes of data:
Reply from 67.205.74.184: bytes=32 time=85ms TTL=49
Reply from 67.205.74.184: bytes=32 time=86ms TTL=49
Reply from 67.205.74.184: bytes=32 time=86ms TTL=49
Reply from 67.205.74.184: bytes=32 time=85ms TTL=49

Ping statistics for 67.205.74.184:
Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 85ms, Maximum = 86ms, Average = 85ms

=== Sample Linux Shell Script ===
Pings several voip.ms servers

#!/bin/sh
# Ping several servers and display Latency, Jitter and Packet Loss
#
# First, create a text file with all servers you want to ping - one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
myHF="voip_ping_hosts.txt"
# Sample file:
# toronto.voip.ms
# montreal.voip.ms
# seattle.voip.ms
# chicago.voip.ms
# newyork.voip.ms
#
echo "============================================"
printf "%-20s %7s %8s %6s\n" "VoIP Server" "Latency" "Jitter" "Loss"
echo "============================================"
cat ${myHF} |\
while read myLn
do
ping -c 3 -i 5 -q $myLn |\
awk '/^PING / {myH=$2}
/packet loss/ {myPL=$6}
/min\/avg\/max/ {
split($4,myS,"/")
printf( "%-20s %3.1f %1.3f %4s\n", myH, myS[2], myS[4], myPL)
}'
done
echo "============================================"

Output:

============================================
VoIP Server Latency Jitter Loss
============================================
toronto.voip.ms 68.3 0.439 0%
montreal.voip.ms 89.6 0.197 0%
seattle.voip.ms 71.2 0.387 0%
chicago.voip.ms 71.6 0.084 0%
newyork.voip.ms 79.1 0.411 0%
============================================

===Bash Script To Handle The Mac Ping Output Format (User Submitted) ===
<p>pingVoipMS.sh script
</p>
<pre>
#!/bin/bash
# Ping several servers and display Latency, Jitter and Packet Loss
# Usage: [-c <count>] [-i <wait time>] [<server list file>]
#
# The optional text file should be formatted with one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
# If no args are supplied, this script will scrape a ping server list from voip.ms

DNLD_DIR="/tmp"; SERVER_LIST="voip_ping_hosts.txt"; PING_LIST="ping_result.txt"; USER_FILE=""
COUNT=3; INTERVAL=5

# Handle any passed in script arguments
while getopts c:i: parm
do
case $parm in
c)count_opt=$OPTARG;;
i)interval_opt=$OPTARG;;
*)echo -e "Invalid arg\nUsage:\t[-c <count of ECHO_REQUESTs to Tx, default 3> ] \
\n\t[-i <wait time (s) between datagrams, default 5> ] \
\n\t[FILE <ping server list> ]";exit 1;;
esac
done

# Test if an option was specified and whether it's a +ve integer
[[ -n $count_opt && ($count_opt =~ ^[[:digit:]]+$) ]] && COUNT=$count_opt
[[ -n $interval_opt && ($interval_opt =~ ^[[:digit:]]+$) ]] && INTERVAL=$interval_opt

shift $(($OPTIND -1))

# Move the last arg (server list) to $1 and validate
[[ -n $1 && !(-f $1 && -r $1) ]] && { echo "\"$1\" file does not exist or is not readable"; exit 1; }
[[ -n $1 && -f $1 && -r $1 ]] && USER_FILE="$1"

if [[ -n $USER_FILE ]]
then
grep -v '^\s*#' $USER_FILE | awk NF > $DNLD_DIR/$SERVER_LIST
else
# N.B. The script looks for the html boldface tags <b> </b> inside a bracket
# If the website alters and the parse fails, manually create the list and
# supply as a script arg (or perhaps update the parsing to work again :)
curl --silent http://wiki.voip.ms/article/Choosing_Server | \
grep '(<b>[[:alpha:]]*[[:alnum:]]\.voip\.ms</b>)' | \
tr "<>" " " | awk '{print $(NF-3)}' > $DNLD_DIR/$SERVER_LIST
fi

echo "PING will send $COUNT packet(s) with a wait of $INTERVAL sec(s) between each packet"
echo "Change the PING options by invoking this script with -c and/or -i, default \"-c 3 -i 5\""
echo "================================================================"
printf "%-20s %-18s %7s %8s %6s\n" "VoIP Server" "IP Address" "Latency" "Jitter" "Loss"
echo "================================================================"

while read myLn
do
ping -c $COUNT -i $INTERVAL -q $myLn | awk '\
/^PING / {myH=$2}
/^PING / {
IP = substr($3,2,15)
split(IP,myIP,")") }
/packet loss/ {myPL=$7}
/min\/avg\/max/ {
split($4,myS,"/")
printf("%-20s %-18s %7.3f %8.3f %6s\n",
myH, myIP[1], myS[2], myS[4], myPL) }
' | tee -a $DNLD_DIR/$PING_LIST
done < $DNLD_DIR/$SERVER_LIST

echo "================================================================"
echo -e "\nMost appropriate server listed in order of best latency\n"
echo "================================================================"
printf "%-20s %-18s %7s %8s %6s\n" "VoIP Server" "IP Address" "Latency" "Jitter" "Loss"
echo "================================================================"
sort -n -k 3,3 -k 5,5 -k 4,4 $DNLD_DIR/$PING_LIST | awk '{printf("%s $%2d$\n",$0, NR)}'
echo "================================================================"
rm $DNLD_DIR/$PING_LIST $DNLD_DIR/$SERVER_LIST
</pre>

=== Perl Script ===
Pings list of voip.ms servers round robin with optional output csv file.

# usage ping_voip.ms.pl <number of times> <seconds in between> <output.csv>
use Net::Ping;
use Time::HiRes;
use strict;

# input list
my @hosts = qw(
atlanta.voip.ms
atlanta2.voip.ms
chicago.voip.ms
chicago2.voip.ms
chicago3.voip.ms
chicago4.voip.ms
dallas.voip.ms
denver.voip.ms
denver2.voip.ms
houston.voip.ms
losangeles.voip.ms
losangeles2.voip.ms
newyork.voip.ms
newyork2.voip.ms
newyork3.voip.ms
newyork4.voip.ms
seattle.voip.ms
seattle2.voip.ms
seattle3.voip.ms
tampa.voip.ms
washington.voip.ms
washington2.voip.ms
montreal.voip.ms
montreal2.voip.ms
montreal3.voip.ms
montreal4.voip.ms
toronto2.voip.ms
toronto3.voip.ms
toronto4.voip.ms
toronto.voip.ms
london.voip.ms
);

$| = 1; #autoflush
# High precision syntax (requires Time::HiRes)
my $p = Net::Ping->new("icmp",1);
$p->hires();
my $max_name_length = (reverse sort { $a <=> $b } map { length($_) } @hosts)[0];
my $count = 4; # number of times to ping
my $interval = 5; # seconds between ping rounds
my $output_file = "";
my @data;

# check for arguments
my $num_args = @ARGV;
if ($num_args >= 1) {$count = $ARGV[0];}
if ($num_args >= 2) {$interval = $ARGV[1];}
if ($num_args >= 3) {$output_file = $ARGV[2];}

# check argument validity
$0 =~ /^.*\\(.*)$/;
my $script = $1;
if ($count !~ /^\d+$/ or $interval !~ /^\d+$/) {die "Usage: $script <number of rounds> <seconds between rounds> <output.csv>\n";}
if (length($output_file) > 0 and $output_file !~ /\.csv$/) {$output_file .= ".csv";}

# main loop
for my $i (1..$count)
{
sleep $interval unless $i == 1;
print "Round $i\n";
my $host_num=0;
foreach my $host (@hosts)
{
(my $ret, my $duration, my $ip) = $p->ping($host);
$ip =~ /(\d+)\.(\d+)\.(\d+)\.(\d+)/;
if ($ret)
{
printf("%*s [ip: %3s.%3s.%3s.%3s] is alive (%6.2f ms)\n", $max_name_length, $host, $1, $2, $3, $4, $duration*1000);
$data[$host_num][$i]=$duration*1000;
}
else
{
printf("%*s [ip: %3s.%3s.%3s.%3s] is dead\n", $max_name_length, $host, $1, $2, $3, $4);
}
$host_num++;
}
print "\n";
}
$p->close();

# if output file name given
if (length($output_file)>0)
{
# print output to file
open FILE, ">$output_file" or die "$!\n";

# print column headers
print FILE "Server\\Round";
for my $i (1..$count)
{
print FILE ", $i";
}
print FILE ", Average\n";

# print data
my $i = 0;
foreach my $host (@hosts)
{
print FILE "$host";
my $sum = 0;
for my $j (1..$count)
{
$sum += $data[$i][$j];
printf FILE ", %8.4f",$data[$i][$j];
}
printf FILE ", %8.4f\n",$sum/$count;
$i++;
}

close FILE;
print "Data written to $output_file\n";
}

# print summary to screen
my $i = 0;
printf("%-*s Average (ms)\n", $max_name_length, "Server");
foreach my $host (@hosts)
{
my $sum = 0;
for my $j (1..$count)
{
$sum += $data[$i][$j];
}
printf("%-*s %8.4f\n", $max_name_length+1, $host, $sum/$count);
$i++;
}

Output:
Round 1
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.97 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.99 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.70 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.76 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.53 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.73 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 94.99 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.05 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.13 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (102.87 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 64.92 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 63.41 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (131.75 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (120.64 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (120.49 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (111.43 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.25 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.86 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 90.85 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (123.29 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.71 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.19 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 81.82 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 86.13 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 77.09 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.18 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.70 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (131.27 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (125.13 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.26 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (152.77 ms)

Round 2
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.14 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.86 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 50.03 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.44 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.33 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.22 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 95.58 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.94 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.29 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (102.73 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.59 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 64.27 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (112.74 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (121.22 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (121.34 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.75 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.06 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.33 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 91.58 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (122.94 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.28 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.40 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 81.91 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 85.64 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 75.15 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.79 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.10 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (150.85 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.40 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.45 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (170.79 ms)

Round 3
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.76 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.86 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.65 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.01 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.05 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.53 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 95.82 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.02 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.60 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (103.35 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.79 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 64.05 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (113.01 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (121.41 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (122.23 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.62 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 93.65 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.19 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 90.75 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (125.12 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.19 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.98 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 80.16 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 87.16 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 76.54 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 97.51 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.18 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (142.81 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.95 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.78 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (153.14 ms)

Round 4
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 89.19 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.98 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.21 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.50 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.68 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.18 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 93.93 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.22 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.10 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (103.67 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.58 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 63.60 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (114.76 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (120.44 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (121.05 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.51 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.04 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 96.92 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 91.23 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (123.28 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.45 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (100.94 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 82.33 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 85.02 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 76.85 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.32 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.22 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (148.33 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (141.61 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (105.91 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (152.85 ms)

Server Average (ms)
atlanta.voip.ms 88.7630
atlanta2.voip.ms 92.9233
chicago.voip.ms 49.6477
chicago2.voip.ms 59.9305
chicago3.voip.ms 59.3972
chicago4.voip.ms 49.9152
dallas.voip.ms 95.0790
denver.voip.ms 94.8077
denver2.voip.ms 85.2797
houston.voip.ms 103.1562
losangeles.voip.ms 65.4693
losangeles2.voip.ms 63.8347
newyork.voip.ms 118.0643
newyork2.voip.ms 120.9265
newyork3.voip.ms 121.2778
newyork4.voip.ms 110.8275
seattle.voip.ms 93.9993
seattle2.voip.ms 95.8267
seattle3.voip.ms 91.1035
tampa.voip.ms 123.6570
washington.voip.ms 98.4065
washington2.voip.ms 101.3774
montreal.voip.ms 81.5525
montreal2.voip.ms 85.9863
montreal3.voip.ms 76.4058
montreal4.voip.ms 96.7013
toronto2.voip.ms 103.7986
toronto3.voip.ms 143.3156
toronto4.voip.ms 136.0254
toronto.voip.ms 104.1012
london.voip.ms 157.3885

For Windows (User Submitted):
# Usage: Copy and paste the following code into a powershell window
# To run it from a command prompt, save this file with extension ps1. Then run Powershell.exe -file "pathtothisscript.ps1"
#Get the list of servers into an array
$Servers =
@(“atlanta.voip.ms”,”atlanta2.voip.ms”,”chicago.voip.ms”,”chicago2.voip.ms”,”chicago3.voip.ms”,”chicago4.voip.ms”,”dallas.voip.ms”,
”denver.voip.ms”,”denver2.voip.ms”,”houston.voip.ms”,”london.voip.ms”,”losangeles.voip.ms”,”losangeles2.voip.ms”,”montreal.voip.ms”,
”montreal2.voip.ms”,”montreal3.voip.ms”,”montreal4.voip.ms”,”newyork.voip.ms”,” newyork2.voip.ms”,”newyork3.voip.ms”,”newyork4.voip.ms”,
”seattle.voip.ms”,”seattle2.voip.ms”,”seattle3.voip.ms”,”tampa.voip.ms”,”toronto.voip.ms”,”toronto2.voip.ms”,”toronto3.voip.ms”,
”toronto4.voip.ms”,”vancouver.voip.ms”,”vancouver2.voip.ms”,”washington.voip.ms”,”washington2.voip.ms”)
$k = 0 #Counting variable so we know what server number we are testing
#num of servers to test
$servercount = $servers.length
#Do the following code for each server in our array
For Each($server in $servers){
$k++ #Add one to the counting variable....we are on server #1...then server 2, then server 3 etc...
Write-Progress -Activity "Testing Server: ${server}" -status "Testing Server $k out of $servercount" -percentComplete ($k /
$servercount*100) #Update the progress bar
$i = 0 #Counting variable for number of times we tried to ping a given server
Do{
$pingsuccess = $false #assume a failure
$i++ #Add one to the counting variable.....1st try....2nd try....3rd try etc...
Try{
$currentping = (test-connection $server -count 1 -ErrorAction Stop).responsetime #Try to ping
$pingsuccess = $True #If success full, set success variable
}Catch { $pingsuccess = $false } #Catch the failure and set the success variable to false
}While($pingsuccess -eq $false -and $i -le 5) #Try everything between Do and While up to 5 times, or while $pingsuccess is not
true
If($pingsuccess -and ($currentping -lt $bestping -or (!($bestping)))){ #Compare the last ping test with the best known ping
test....if there is no known best ping test, assume this one is the best $bestping = $currentping #If this is the best ping...save it
$bestserver = $server #Save the best server
}
write-host "tested: $server at $currentping ms after $i attempts" #write the results of the test for this server
}
write-host "The server with the best ping is: $bestserver at $bestping ms" #write the end result

= Latency and it's importance =

Latency is very important for Voip, this will determine the time that will take for the data package transmission to reach the destination. A high latency will lead to a delay and echoes in the communication.

Latency is measured in milliseconds (ms) For example: a latency of 150ms is barely noticeable, thus acceptable. Higher than that, quality starts to suffer. When it gets higher than 300 ms, it becomes unacceptable.

[[category:guides]]

Choosing Server

2016-11-14T15:03:22Z

Rod: /* IPs */

[[File:Serverlocation3.png]]

= Choosing a Server =

[http://www.voip.ms VoIP.ms] offers many different servers, but which one should you choose? One misconception is that you should pick the closest to your location, however this is not needed most of the time. For example, if you are in the USA, any of the US servers will provide a really good latency and service quality. The newest server within a city is indicated with the highest number attached to the name, as they are classified in ascending order. Also worth noting is that there is a network tool that will help you when deciding which server you want to use, generally named a "ping", which will provide you the latency between you and the server. Therefore the server which provides you less latency should be used.

=== IPs ===

*Amsterdam, NL ('''amsterdam.voip.ms''') 37.58.88.242
*Atlanta 1, GA ('''atlanta.voip.ms''') 75.127.65.130
*Atlanta 2, GA ('''atlanta2.voip.ms''') 72.9.246.170
*Chicago 1, IL ('''chicago.voip.ms''') 208.100.39.52
*Chicago 2, IL ('''chicago2.voip.ms''') 208.100.39.53
*Chicago 3, IL ('''chicago3.voip.ms''') 208.100.39.54
*Chicago 4, IL ('''chicago4.voip.ms''') 208.100.39.55
*Dallas, TX ('''dallas.voip.ms''') 158.85.149.162
*Dallas 2, TX ('''dallas2.voip.ms''') 158.85.149.163
*Denver 1, CO ('''denver.voip.ms''') 173.248.161.90
*Denver 2, CO ('''denver2.voip.ms''') 173.248.159.210
*Houston, TX ('''houston.voip.ms''') 173.193.85.18
*Houston 2, TX ('''houston2.voip.ms''') 173.193.85.19
*London, UK ('''london.voip.ms''') 159.8.157.212
*Los Angeles 1, CA ('''losangeles.voip.ms''') 96.44.149.186
*Los Angeles 2, CA ('''losangeles2.voip.ms''') 96.44.149.202
*Melbourne, AU ('''melbourne.voip.ms''') 168.1.73.84
*Montreal 1, QC ('''montreal.voip.ms''') 67.205.74.184
*Montreal 2, QC ('''montreal2.voip.ms''') 67.205.74.187
*Montreal 3, QC ('''montreal3.voip.ms''') 192.175.96.68
*Montreal 4, QC ('''montreal4.voip.ms''') 67.205.74.179
*Montreal 5, QC ('''montreal5.voip.ms''') 192.175.96.69
*Montreal 6, QC ('''montreal6.voip.ms''') 192.175.96.70
*Montreal 7, QC ('''montreal7.voip.ms''') 192.175.96.71
*Montreal 8, QC ('''montreal8.voip.ms''') 192.175.96.72
*New York 1, NY ('''newyork.voip.ms''') 74.63.41.218
*New York 2, NY ('''newyork2.voip.ms''') 107.6.67.236
*New York 3, NY ('''newyork3.voip.ms''') 107.6.67.237
*New York 4, NY ('''newyork4.voip.ms''') 107.6.67.238
*New York 5, NY ('''newyork5.voip.ms''') 23.29.136.28
*New York 6, NY ('''newyork6.voip.ms''') 23.29.136.29
*New York 7, NY ('''newyork7.voip.ms''') 23.29.136.38
*New York 8, NY ('''newyork8.voip.ms''') 23.29.136.40
*Paris, FR ('''paris.voip.ms''') 159.8.85.180
*San Jose, CA ('''sanjose.voip.ms''') 23.246.247.146
*San Jose 2, CA ('''sanjose2.voip.ms''') 23.246.247.147
*Seattle 1, WA ('''seattle.voip.ms''') 50.23.160.50
*Seattle 2, WA ('''seattle2.voip.ms''') 50.23.160.51
*Seattle 3, WA ('''seattle3.voip.ms''') 50.23.160.52
*Tampa, FL ('''tampa.voip.ms''') 162.254.144.173
*Tampa 2, FL ('''tampa2.voip.ms''') 162.254.144.176
*Toronto 1, ON ('''toronto.voip.ms''') 158.85.70.148
*Toronto 2, ON ('''toronto2.voip.ms''') 158.85.70.149
*Toronto 3, ON ('''toronto3.voip.ms''') 158.85.70.150
*Toronto 4, ON ('''toronto4.voip.ms''') 158.85.70.151
*Toronto 5, ON ('''toronto5.voip.ms''') 184.75.215.106
*Toronto 6, ON ('''toronto6.voip.ms''') 184.75.215.114
*Toronto 7, ON ('''toronto7.voip.ms''') 184.75.215.146
*Toronto 8, ON ('''toronto8.voip.ms''') 184.75.213.210
*Vancouver 1, BC ('''vancouver.voip.ms''') 162.213.157.82
*Vancouver 2, BC ('''vancouver2.voip.ms''') 162.213.157.117
*Washington 1, DC ('''washington.voip.ms''') 208.43.234.226
*Washington 2, DC ('''washington2.voip.ms''') 208.43.234.227

===Server Realms===

For IOS, Please click here [http://wiki.voip.ms/article/Server_Realms Server Realms] to get the Realm Name for the server you plan on using, this can differ from the Domain Name being used.

Please note that the following servers will not be available to select as a DID Point of Presence for newer accounts:
Chicago 1, New York 1, Seattle 1, Montreal 1, Montreal 2 and Toronto 1.

= What is a Ping? =

Ping is a standard tool used to test network connections. It is mostly used to determine if a server or device can be reached across the network and the latency of the response(the time it takes to send a packet to the destination and for it to return to your computer).

Ping tools are part of Windows, Mac OS X and Linux as well as some routers.

== How does the ping work? ==

It sends request messages to a target network address or DNS names at periodic intervals and measures the time it takes for a response message to arrive and return(better known as latency).

==How to ping on a PC==

Pinging is a command which tells you if the connection between your computer and a particular domain is working correctly.

In Windows, select Start > Programs > Accessories > Command Prompt. This will give you a window like the one below.

Enter the word ping, followed by a space, then the domain name.(montreal.voip.ms) in this case domain is our server name.

If the results show a series of replies, the connection is working. The time shows you how fast the connection is. If you see a "timed out" error instead of a reply, there is a breakdown somewhere between your computer and the domain.

[[File:Ping.gif]]

==How to ping on a Mac Computer==

1- Click on Finder in the dock.

2- Click on Applications.

3- Click on Utilities.

4- Double-click on Network Utility. *

* In OS X Mavericks (10.9.x) this utility app changed location. Launch it from spotlight instead, either press "command"+"space bar" or click on spotlight directly (magnifying glass icon at top right of screen), type "network utility" and hit "return"

5- In the Network Utility window, click on the Ping tab

6- In the field under "Please enter the network address to ping," like montreal.voip.ms

''If pings results are not consistent, you may have an issue with Jitter. You can work on this issue by adjusting the "Network Jitter Level" setting on your VoIP device. Usually a ping of under 150 ms is recommended in order to have good quality. The latency time to the server is important, however there are also other factors that could affect the quality of the calls such as packet loss (VoIP communications are very sensitive to this), and the Jitter level of your Internet connection.''

The following is the output of running ping with the target losangeles.voip.ms.

#ping losangeles.voip.ms
Ping to losangeles.voip.ms [67.215.241.250] with 32 bytes de datos:
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
Response from 67.215.241.250: bytes=32 time=69ms TTL=52
Response from 67.215.241.250: bytes=32 time=68ms TTL=52
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
ping statistics from 67.215.241.250:
4 packets transmitted, 4 received, 0% packet lost. rtt min/avg/max/mdev = 67ms, 69ms, 67ms

Sample ping output in windows:
C:\Windows\system32>ping montreal.voip.ms

Pinging montreal.voip.ms [67.205.74.184] with 32 bytes of data:
Reply from 67.205.74.184: bytes=32 time=85ms TTL=49
Reply from 67.205.74.184: bytes=32 time=86ms TTL=49
Reply from 67.205.74.184: bytes=32 time=86ms TTL=49
Reply from 67.205.74.184: bytes=32 time=85ms TTL=49

Ping statistics for 67.205.74.184:
Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 85ms, Maximum = 86ms, Average = 85ms

=== Sample Linux Shell Script ===
Pings several voip.ms servers

#!/bin/sh
# Ping several servers and display Latency, Jitter and Packet Loss
#
# First, create a text file with all servers you want to ping - one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
myHF="voip_ping_hosts.txt"
# Sample file:
# toronto.voip.ms
# montreal.voip.ms
# seattle.voip.ms
# chicago.voip.ms
# newyork.voip.ms
#
echo "============================================"
printf "%-20s %7s %8s %6s\n" "VoIP Server" "Latency" "Jitter" "Loss"
echo "============================================"
cat ${myHF} |\
while read myLn
do
ping -c 3 -i 5 -q $myLn |\
awk '/^PING / {myH=$2}
/packet loss/ {myPL=$6}
/min\/avg\/max/ {
split($4,myS,"/")
printf( "%-20s %3.1f %1.3f %4s\n", myH, myS[2], myS[4], myPL)
}'
done
echo "============================================"

Output:

============================================
VoIP Server Latency Jitter Loss
============================================
toronto.voip.ms 68.3 0.439 0%
montreal.voip.ms 89.6 0.197 0%
seattle.voip.ms 71.2 0.387 0%
chicago.voip.ms 71.6 0.084 0%
newyork.voip.ms 79.1 0.411 0%
============================================

===Bash Script To Handle The Mac Ping Output Format (User Submitted) ===
<p>pingVoipMS.sh script
</p>
<pre>
#!/bin/bash
# Ping several servers and display Latency, Jitter and Packet Loss
# Usage: [-c <count>] [-i <wait time>] [<server list file>]
#
# The optional text file should be formatted with one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
# If no args are supplied, this script will scrape a ping server list from voip.ms

DNLD_DIR="/tmp"; SERVER_LIST="voip_ping_hosts.txt"; PING_LIST="ping_result.txt"; USER_FILE=""
COUNT=3; INTERVAL=5

# Handle any passed in script arguments
while getopts c:i: parm
do
case $parm in
c)count_opt=$OPTARG;;
i)interval_opt=$OPTARG;;
*)echo -e "Invalid arg\nUsage:\t[-c <count of ECHO_REQUESTs to Tx, default 3> ] \
\n\t[-i <wait time (s) between datagrams, default 5> ] \
\n\t[FILE <ping server list> ]";exit 1;;
esac
done

# Test if an option was specified and whether it's a +ve integer
[[ -n $count_opt && ($count_opt =~ ^[[:digit:]]+$) ]] && COUNT=$count_opt
[[ -n $interval_opt && ($interval_opt =~ ^[[:digit:]]+$) ]] && INTERVAL=$interval_opt

shift $(($OPTIND -1))

# Move the last arg (server list) to $1 and validate
[[ -n $1 && !(-f $1 && -r $1) ]] && { echo "\"$1\" file does not exist or is not readable"; exit 1; }
[[ -n $1 && -f $1 && -r $1 ]] && USER_FILE="$1"

if [[ -n $USER_FILE ]]
then
grep -v '^\s*#' $USER_FILE | awk NF > $DNLD_DIR/$SERVER_LIST
else
# N.B. The script looks for the html boldface tags <b> </b> inside a bracket
# If the website alters and the parse fails, manually create the list and
# supply as a script arg (or perhaps update the parsing to work again :)
curl --silent http://wiki.voip.ms/article/Choosing_Server | \
grep '(<b>[[:alpha:]]*[[:alnum:]]\.voip\.ms</b>)' | \
tr "<>" " " | awk '{print $(NF-3)}' > $DNLD_DIR/$SERVER_LIST
fi

echo "PING will send $COUNT packet(s) with a wait of $INTERVAL sec(s) between each packet"
echo "Change the PING options by invoking this script with -c and/or -i, default \"-c 3 -i 5\""
echo "================================================================"
printf "%-20s %-18s %7s %8s %6s\n" "VoIP Server" "IP Address" "Latency" "Jitter" "Loss"
echo "================================================================"

while read myLn
do
ping -c $COUNT -i $INTERVAL -q $myLn | awk '\
/^PING / {myH=$2}
/^PING / {
IP = substr($3,2,15)
split(IP,myIP,")") }
/packet loss/ {myPL=$7}
/min\/avg\/max/ {
split($4,myS,"/")
printf("%-20s %-18s %7.3f %8.3f %6s\n",
myH, myIP[1], myS[2], myS[4], myPL) }
' | tee -a $DNLD_DIR/$PING_LIST
done < $DNLD_DIR/$SERVER_LIST

echo "================================================================"
echo -e "\nMost appropriate server listed in order of best latency\n"
echo "================================================================"
printf "%-20s %-18s %7s %8s %6s\n" "VoIP Server" "IP Address" "Latency" "Jitter" "Loss"
echo "================================================================"
sort -n -k 3,3 -k 5,5 -k 4,4 $DNLD_DIR/$PING_LIST | awk '{printf("%s $%2d$\n",$0, NR)}'
echo "================================================================"
rm $DNLD_DIR/$PING_LIST $DNLD_DIR/$SERVER_LIST
</pre>

=== Perl Script ===
Pings list of voip.ms servers round robin with optional output csv file.

# usage ping_voip.ms.pl <number of times> <seconds in between> <output.csv>
use Net::Ping;
use Time::HiRes;
use strict;

# input list
my @hosts = qw(
atlanta.voip.ms
atlanta2.voip.ms
chicago.voip.ms
chicago2.voip.ms
chicago3.voip.ms
chicago4.voip.ms
dallas.voip.ms
denver.voip.ms
denver2.voip.ms
houston.voip.ms
losangeles.voip.ms
losangeles2.voip.ms
newyork.voip.ms
newyork2.voip.ms
newyork3.voip.ms
newyork4.voip.ms
seattle.voip.ms
seattle2.voip.ms
seattle3.voip.ms
tampa.voip.ms
washington.voip.ms
washington2.voip.ms
montreal.voip.ms
montreal2.voip.ms
montreal3.voip.ms
montreal4.voip.ms
toronto2.voip.ms
toronto3.voip.ms
toronto4.voip.ms
toronto.voip.ms
london.voip.ms
);

$| = 1; #autoflush
# High precision syntax (requires Time::HiRes)
my $p = Net::Ping->new("icmp",1);
$p->hires();
my $max_name_length = (reverse sort { $a <=> $b } map { length($_) } @hosts)[0];
my $count = 4; # number of times to ping
my $interval = 5; # seconds between ping rounds
my $output_file = "";
my @data;

# check for arguments
my $num_args = @ARGV;
if ($num_args >= 1) {$count = $ARGV[0];}
if ($num_args >= 2) {$interval = $ARGV[1];}
if ($num_args >= 3) {$output_file = $ARGV[2];}

# check argument validity
$0 =~ /^.*\\(.*)$/;
my $script = $1;
if ($count !~ /^\d+$/ or $interval !~ /^\d+$/) {die "Usage: $script <number of rounds> <seconds between rounds> <output.csv>\n";}
if (length($output_file) > 0 and $output_file !~ /\.csv$/) {$output_file .= ".csv";}

# main loop
for my $i (1..$count)
{
sleep $interval unless $i == 1;
print "Round $i\n";
my $host_num=0;
foreach my $host (@hosts)
{
(my $ret, my $duration, my $ip) = $p->ping($host);
$ip =~ /(\d+)\.(\d+)\.(\d+)\.(\d+)/;
if ($ret)
{
printf("%*s [ip: %3s.%3s.%3s.%3s] is alive (%6.2f ms)\n", $max_name_length, $host, $1, $2, $3, $4, $duration*1000);
$data[$host_num][$i]=$duration*1000;
}
else
{
printf("%*s [ip: %3s.%3s.%3s.%3s] is dead\n", $max_name_length, $host, $1, $2, $3, $4);
}
$host_num++;
}
print "\n";
}
$p->close();

# if output file name given
if (length($output_file)>0)
{
# print output to file
open FILE, ">$output_file" or die "$!\n";

# print column headers
print FILE "Server\\Round";
for my $i (1..$count)
{
print FILE ", $i";
}
print FILE ", Average\n";

# print data
my $i = 0;
foreach my $host (@hosts)
{
print FILE "$host";
my $sum = 0;
for my $j (1..$count)
{
$sum += $data[$i][$j];
printf FILE ", %8.4f",$data[$i][$j];
}
printf FILE ", %8.4f\n",$sum/$count;
$i++;
}

close FILE;
print "Data written to $output_file\n";
}

# print summary to screen
my $i = 0;
printf("%-*s Average (ms)\n", $max_name_length, "Server");
foreach my $host (@hosts)
{
my $sum = 0;
for my $j (1..$count)
{
$sum += $data[$i][$j];
}
printf("%-*s %8.4f\n", $max_name_length+1, $host, $sum/$count);
$i++;
}

Output:
Round 1
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.97 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.99 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.70 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.76 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.53 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.73 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 94.99 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.05 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.13 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (102.87 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 64.92 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 63.41 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (131.75 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (120.64 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (120.49 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (111.43 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.25 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.86 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 90.85 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (123.29 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.71 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.19 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 81.82 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 86.13 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 77.09 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.18 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.70 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (131.27 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (125.13 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.26 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (152.77 ms)

Round 2
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.14 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.86 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 50.03 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.44 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.33 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.22 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 95.58 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.94 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.29 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (102.73 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.59 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 64.27 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (112.74 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (121.22 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (121.34 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.75 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.06 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.33 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 91.58 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (122.94 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.28 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.40 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 81.91 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 85.64 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 75.15 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.79 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.10 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (150.85 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.40 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.45 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (170.79 ms)

Round 3
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.76 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.86 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.65 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.01 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.05 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.53 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 95.82 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.02 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.60 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (103.35 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.79 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 64.05 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (113.01 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (121.41 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (122.23 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.62 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 93.65 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 95.19 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 90.75 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (125.12 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.19 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (101.98 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 80.16 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 87.16 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 76.54 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 97.51 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.18 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (142.81 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.95 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (103.78 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (153.14 ms)

Round 4
atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 89.19 ms)
atlanta2.voip.ms [ip: 72. 9.246.170] is alive ( 92.98 ms)
chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.21 ms)
chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.50 ms)
chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.68 ms)
chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.18 ms)
dallas.voip.ms [ip: 74. 54. 54.178] is alive ( 93.93 ms)
denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.22 ms)
denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.10 ms)
houston.voip.ms [ip: 209. 62. 1. 2] is alive (103.67 ms)
losangeles.voip.ms [ip: 96. 44.149.186] is alive ( 65.58 ms)
losangeles2.voip.ms [ip: 96. 44.149.202] is alive ( 63.60 ms)
newyork.voip.ms [ip: 74. 63. 41.218] is alive (114.76 ms)
newyork2.voip.ms [ip: 107. 6. 67.236] is alive (120.44 ms)
newyork3.voip.ms [ip: 107. 6. 67.237] is alive (121.05 ms)
newyork4.voip.ms [ip: 107. 6. 67.238] is alive (110.51 ms)
seattle.voip.ms [ip: 50. 23.160. 50] is alive ( 94.04 ms)
seattle2.voip.ms [ip: 50. 23.160. 51] is alive ( 96.92 ms)
seattle3.voip.ms [ip: 50. 23.160. 52] is alive ( 91.23 ms)
tampa.voip.ms [ip: 68.233.226. 97] is alive (123.28 ms)
washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.45 ms)
washington2.voip.ms [ip: 208. 43.234.227] is alive (100.94 ms)
montreal.voip.ms [ip: 67.205. 74.184] is alive ( 82.33 ms)
montreal2.voip.ms [ip: 67.205. 74.187] is alive ( 85.02 ms)
montreal3.voip.ms [ip: 72. 55.168. 18] is alive ( 76.85 ms)
montreal4.voip.ms [ip: 67.205. 74.179] is alive ( 96.32 ms)
toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.22 ms)
toronto3.voip.ms [ip: 184. 75.215.146] is alive (148.33 ms)
toronto4.voip.ms [ip: 184. 75.213.210] is alive (141.61 ms)
toronto.voip.ms [ip: 184. 75.215.106] is alive (105.91 ms)
london.voip.ms [ip: 5. 77. 36.136] is alive (152.85 ms)

Server Average (ms)
atlanta.voip.ms 88.7630
atlanta2.voip.ms 92.9233
chicago.voip.ms 49.6477
chicago2.voip.ms 59.9305
chicago3.voip.ms 59.3972
chicago4.voip.ms 49.9152
dallas.voip.ms 95.0790
denver.voip.ms 94.8077
denver2.voip.ms 85.2797
houston.voip.ms 103.1562
losangeles.voip.ms 65.4693
losangeles2.voip.ms 63.8347
newyork.voip.ms 118.0643
newyork2.voip.ms 120.9265
newyork3.voip.ms 121.2778
newyork4.voip.ms 110.8275
seattle.voip.ms 93.9993
seattle2.voip.ms 95.8267
seattle3.voip.ms 91.1035
tampa.voip.ms 123.6570
washington.voip.ms 98.4065
washington2.voip.ms 101.3774
montreal.voip.ms 81.5525
montreal2.voip.ms 85.9863
montreal3.voip.ms 76.4058
montreal4.voip.ms 96.7013
toronto2.voip.ms 103.7986
toronto3.voip.ms 143.3156
toronto4.voip.ms 136.0254
toronto.voip.ms 104.1012
london.voip.ms 157.3885

For Windows (User Submitted):
# Usage: Copy and paste the following code into a powershell window
# To run it from a command prompt, save this file with extension ps1. Then run Powershell.exe -file "pathtothisscript.ps1"
#Get the list of servers into an array
$Servers =
@(“atlanta.voip.ms”,”atlanta2.voip.ms”,”chicago.voip.ms”,”chicago2.voip.ms”,”chicago3.voip.ms”,”chicago4.voip.ms”,”dallas.voip.ms”,
”denver.voip.ms”,”denver2.voip.ms”,”houston.voip.ms”,”london.voip.ms”,”losangeles.voip.ms”,”losangeles2.voip.ms”,”montreal.voip.ms”,
”montreal2.voip.ms”,”montreal3.voip.ms”,”montreal4.voip.ms”,”newyork.voip.ms”,” newyork2.voip.ms”,”newyork3.voip.ms”,”newyork4.voip.ms”,
”seattle.voip.ms”,”seattle2.voip.ms”,”seattle3.voip.ms”,”tampa.voip.ms”,”toronto.voip.ms”,”toronto2.voip.ms”,”toronto3.voip.ms”,
”toronto4.voip.ms”,”vancouver.voip.ms”,”vancouver2.voip.ms”,”washington.voip.ms”,”washington2.voip.ms”)
$k = 0 #Counting variable so we know what server number we are testing
#num of servers to test
$servercount = $servers.length
#Do the following code for each server in our array
For Each($server in $servers){
$k++ #Add one to the counting variable....we are on server #1...then server 2, then server 3 etc...
Write-Progress -Activity "Testing Server: ${server}" -status "Testing Server $k out of $servercount" -percentComplete ($k /
$servercount*100) #Update the progress bar
$i = 0 #Counting variable for number of times we tried to ping a given server
Do{
$pingsuccess = $false #assume a failure
$i++ #Add one to the counting variable.....1st try....2nd try....3rd try etc...
Try{
$currentping = (test-connection $server -count 1 -ErrorAction Stop).responsetime #Try to ping
$pingsuccess = $True #If success full, set success variable
}Catch { $pingsuccess = $false } #Catch the failure and set the success variable to false
}While($pingsuccess -eq $false -and $i -le 5) #Try everything between Do and While up to 5 times, or while $pingsuccess is not
true
If($pingsuccess -and ($currentping -lt $bestping -or (!($bestping)))){ #Compare the last ping test with the best known ping
test....if there is no known best ping test, assume this one is the best $bestping = $currentping #If this is the best ping...save it
$bestserver = $server #Save the best server
}
write-host "tested: $server at $currentping ms after $i attempts" #write the results of the test for this server
}
write-host "The server with the best ping is: $bestserver at $bestping ms" #write the end result

= Latency and it's importance =

Latency is very important for Voip, this will determine the time that will take for the data package transmission to reach the destination. A high latency will lead to a delay and echoes in the communication.

Latency is measured in milliseconds (ms) For example: a latency of 150ms is barely noticeable, thus acceptable. Higher than that, quality starts to suffer. When it gets higher than 300 ms, it becomes unacceptable.

[[category:guides]]

FAQ

2016-11-09T19:55:50Z

Rod:

[[File:Logofaq.jpg|250px|]]

<span style="font-size: 120%;">__TOC__</span>

Pour la page en Francais veillez cliquez ici: [[Questions_Les_Plus_Fréquentes|Questions Les Plus Fréquentes]]

Haga clic aquí para Español:[[Preguntas frecuentes]]

==Do you offer a Test Account or Number?==
We currently do not have a test account or test number program however opening an account is completely free and we provide free dialing codes to test the quality of the service.
Please check our many dialing codes here: [[Dialing Codes]]

==How do I start?==
You can open an account for free following this link : https://voip.ms/signup.php
Once your account is active you will be able to log in and find all of our features and options.
We suggest to also check our [[Getting Started]] guide.

==Is there a cost to open the account?==
Opening an account is completely free. The minimum amount of funds you can add is $25 USD which will become your account balance to use the services.

==Is there a Contract?==
There are no contracts with us, you can open an account for free at any time you wish, for as long as you want.

==What are the Payment Options?==
The current options available are Paypal (No Account Needed for Paypal Guest CC Payment), VISA, Master Card, American Express, Discover (Where supported) and via Bank Wire deposit to our bank account in Montreal.
Please contact Customer Support for information about Bank Wire.

==Can I deposit less than $25 into my account?==
Currently $25 is the minimum amount you can add to your account. This credit won’t expire and will last until depleted by using the services.
You can ask for a refund of the remaining balance at any time should you choose to cancel the account.

==Do you charge taxes?==
All customers with Canada as country will pay for the tax called GST. The customers who are also from Quebec in Canada will pay an additional province tax, PST. This applies for both Paypal and Credit Card payments.
The HST will get applied on Canadian provinces that require it.
In order to get further details about how these are applied, feel free to check this link: http://en.wikipedia.org/wiki/Sales_taxes_in_Canada

==How much does the service cost?==
The cost will vary depending on the services used.
For outbound calls you are subject to the per minute rate of the destination you call and the billing increment.
For inbound calls you are subject to the rate plan of the DID, the monthly fee and the billing increment it if applies.
There may be additional monthly fees or set up fees depending on the services you subscribe to.
Please check more details about the service cost here [[Service Cost]]

==What is the Billing Increment?==
The Billing Increment is a call duration measurement unit expressed in seconds.
For example, with a $0.0100 per minute rate, and a 60 seconds billing increment, any call less than 60 seconds is rounded up to 60 seconds or 1 full minute of usage. A 5 seconds call and a 50 seconds call are both billed as 1 minute or $0.0100.

With our 6 seconds billing increment, a 5 second call will be billed as 6 seconds or 1/10 of 1 minute or $0.0010.
This way, you don't get charged for a whole minute just for a 10 seconds call.

==What is a DID?==
A Direct Inward Dial number (DID), in simple terms, is a virtual number that, for all intents and purposes, can be considered a regular phone number, with the exception that it is not attached to any POTS line (Landline). Once your configuration is ready, your DID will be the phone number that everyone in the world will call to reach you, just like any other phone number.

==What type of DID numbers you offer?==
There are different types of DID numbers. They are mostly differentiated by their geographic presence.
Local DIDs from US or Canada.
International DIDs.
Toll Free numbers.

These are the most common used by our customers. For further information about getting a DID number and other types please check our related article [[Order a DID Number]]

==Can I dial out with my DID?==
The DID numbers are a type of service that are exclusively for receiving calls.
When you place outgoing calls with our service you are not calling from the DID number but from the account instead and the number you will pass on your calls as your Caller ID will be the one that is configured on the accounts settings from the portal.

==What is Caller ID?==
Caller ID is a telephone service that transmits the calling party’s number to the called party’s telephone . When available, the Caller ID number can be complemented with Caller ID name (e.g. John Smith).
If you are placing outgoing calls you will likely need to pass a Caller ID to ensure proper termination of your calls, particularly to reach toll free numbers.
There are two types of caller ID and it is important to differentiate them: Caller ID Number (CID) and Caller ID Name (CNAM).
Please contact Technical Support for assistance or more details about Caller ID or check our related article at [[Caller ID]]

==What is CNAM?==
CNAM stands for CallerID Name and it's the information that will be displayed in the phone of the receiving party, when you place an outgoing call.
If you will be making calls to Canadian numbers, you can simply pass the Caller ID name from your device or system as most of them support this.
The Caller ID name on US calls works differently, this is controlled by a national CNAM database with records of numbers and names matching each number.
We can update the CNAM database under request. For more information please check our related article at [[http://wiki.voip.ms/article/Caller_ID#Outgoing_Caller_ID_name Caller ID Name]]

==Can I port my existing number from another provider to VoIP.ms?==
VoIP.ms does offer Local Number Portability (LNP) service and your number may be available for porting. Please contact Customer Service to find out whether your current number is portable to VoIP.ms network or if you already have an open account, please refer to the [[Porting a Number]] guide.

==Do you provide any hardware for the service?==
We don’t provide any kind of hardware device, software or system to use the service. The service is a BYOD (Bring your own device). You should be able to get one from any Communications specialized store and all SIP- compatible devices are supported.

==Can I use my existing device with VoIP.ms?==
Basically any device or system that supports SIP or IAX2 protocol will work with our service . If you bring your device (ATA, IP phone) from a previous provider, make sure it is unlocked and you are able to make changes to its configuration. You can find a list of some devices that we support, with their configuration guides, at [[Devices]]

==Do you have Configuration Samples?==
You can find configuration samples for most of the common devices and phone systems used with VoIP.ms here: [[Devices]]
If your device or system is not listed there but supports the protocol SIP or IAX2 and at least one of the following codecs: G711u, G729a or GMS, you should be able to use it, and you can always contact our Technical Support staff for additional assistance.

==Do you offer Technical Support by phone?==
Due to the nature of the troubleshooting which can result time consuming we chose our main ways to offer support to be via Ticket System (Email) and Live chat.
At this point, Technical Support over the phone is not available.

==What is my Main Account SIP password?==
The main account SIP password is, by default, the same as your customer portal password. If you have not changed the SIP password, it is the same password you use to log into your portal. You can change your SIP password from the Customer Portal at any time from the Main Menu - Account Settings page. Please check our related article for more information: [[http://wiki.voip.ms/article/Account_Settings#Security Account Security]]

==What Server should I use?==
Usually, in order to receive better results, you should choose the server physically closest to your location. You can also send a ping to any of the servers to check the best response time.
Please check our related article for more information about this subject [[Choosing Server]]

==Does VoIP.ms encrypt the communication?==
The SIP communication is secure although not encrypted. However, the passwords are MD5 hashed and are not transmitted without encryption when establishing the call.

==Do you Support e911==
We offer e911 service only for US and Canadian DID numbers (including USA and Canadian Toll-free numbers). This feature can be activated in the Customer Portal directly under the e911 page under the DID numbers menu.
More information available in our related wiki article here [[E911]]

==Do you offer Calling Card?==
We do not offer any Calling Card feature at this moment, however you can set up a Calling Card solution with your own system and our DID numbers as long as this is used with DID numbers in our Per Minute plan.
For more information please contact Technical Support.

==Do you offer Fax service?==
We offer a Virtual Fax feature. This feature can be used with special Fax DID numbers which can be acquired and configured through our Fax portal.
Please check our related article for further details: [[Virtual Fax]]

==Do you offer SMS service?==
We do offer SMS service (currently on Beta). The SMS is available for a great number of DIDs and Cities from US and Canada and requires to be used through the SMS portal.
SMS is not available through SIP protocol for the moment.
You can read more details about this feature in our related article [[SMS]]

==Do you offer Conference Calls?==
We do not offer Conference calls as a feature in the customer portal, however if your system or device is capable of establishing a conference call or possess this feature, you can use it with our services.

==Do you support TCP for the SIP communication?==
TCP is fully supported in our servers.

==Do you transmit DSCP QoS priority bits?==
Yes we do, we transmit Code CS3 for SIP and EF for RTP.

==Do you offer alternative ports besides 5060?==
We offer alternative SIP ports, UDP/TCP 5080 and 42872 on all of our servers, You can try those ports in case your Internet Service Provider blocks the port 5060 UDP/TCP or if you need to use another one.

==Can I register 2 or more different devices with the same account username?==
It is strongly suggested not to do so, this can possibly cause conflicts while routing the calls to your device as well issues related with registration. If you need to register more than one device please create and use the Sub Accounts, you will get new credentials for any additional device.
For more information about Sub Accounts please check the related article on this subject [[Sub Accounts]]

==Do you accept Autodialers or Telemarketing traffic ?==
We do not accept the use of our termination (outbound) services for telemarketing purposes (Including but not limited to Automated Dialers, Call Centers and collection agencies).

==Do you offer service to Call Centers ?==
We currently provide service to Call Centers for inbound traffic, the use of this is restricted to DID numbers with a per minute plan.
Outbound traffic from call centers is not supported at this time. If you have any concern about this please contact Customer Support for more details.

==Can I resell your services ?==
Yes, our customers will be able to find a "Reseller Section" included in the Customer Portal.
We give you all the tools you need plus a White label Reseller Interface that has been developed to help our clients upsell our services under their own brand. We also recommend that you must familiarize yourself with the Voip.ms interface before reselling the service.
For more information about reseller please contact Technical Support or visit our related article here: [[Reseller Basic Guide]]

==Do you offer Wholesale Rates ?==
If you are interested on a discount based on traffic usage or volume, please send an email to sales@voip.ms providing all details about your traffic like destinations you need to call and average of minutes used per month, in order to receive additional information and quotes.

==Can I port out my number ?==
If you wish to port out a number from our service you can do so at any time by starting the porting request with the new provider. We authorize all Port Out Request matching the correct information from your account.
Remember to delete the DID number from your VoIP.ms account once the port out process is completed.
For any question regarding Port Out please send an email to our LNP Department at ports@voip.ms

==How do I delete my DID ?==
From the customer portal, go to the “DID Numbers” menu tab and then click “DID Billing”. From this page, you can cancel number(s). Remember this process cannot be reversed, please be sure that you will no longer need the number.

==Is there any Cancellation Fee ?==
We do not charge any cancellation fee of any kind, you are free to cancel your account or number at any time you wish to.

==Where are you located ?==
VoIP.ms is a Canadian Company founded back in 2007 in Montreal.
The main office is located in Montreal, QC, Canada and there is also a Technical Site in Merida, Yucatan, Mexico for the South America market.

==Do you offer a Referral Program ?==
For the moment we don't have any kind of referral program available.

[[Category: Guides]]

Fax Virtual

2016-11-08T21:00:22Z

Rod:

[[File:Faxhomelogo.png|center]]

El Fax Virtual es una aplicación cuyo uso es mandar y recibir faxes con el servicio de Voip.ms utilizando un número DID dedicado específicamente para Fax. Usted puede obtener un número para esta aplicación desde el Portal del usuario, en la sección de Fax Numbers bajo la opción de ''Order DID(s)'' del menu de ''DID Numbers''. Los Números DIDs regulares utilizados para llamadas de telefono ''NO'' son compatibles con la aplicación de Fax Virtual.

__TOC__

== Información importante acerca del servicio de Fax Virtual ==

* '''El Servicio de Fax Virtual solo esta disponible para números DID de US y Canada, específicamente adquiridos através de la sección de Fax Numbers en la página de ''Order DID''. El porteo de su propio número de Fax no es posible por el momento, estamos trabajando en eso pero aún no hay un tiempo determinado para su disponibilidad.'''
* '''Este servicio solo puede ser utilizado para mandar mensajes a números Canadienses y de Estados Unidos por el momento. No garantizamos que mensajes Internacionales sean recibidos apropiadamente.'''

Features

2016-11-08T20:59:48Z

Rod:

[[File:Logo.png|400px]]

One of the benefits of having a VoIP.ms account is that not only do you get both origination and termination services, but VoIP.ms also offers a complete solution for your business or home by providing you with many options and features to make your life easier than ever. Most of the Features listed below are completely free with the service and you will not need to pay extra to have these awesome benefits. To fully understand each of these features, you can click on the the feature name to access the specific guide for each topic.

{|border="1"
|+
!<span style="font-size: 150%;">Feature</span> !! <span style="font-size: 150%;">Summary</span>
|-
! [[Call Detail Records|Call Details Report and Invoice Generation]]
|You will be able to examine detailed information of all the calls coming and going from your Main account and Sub accounts simply from the portal.
Additionally you will find the '''Finances''' Menu, where you will get access to the tool to Generate Invoices according to your needs and also be able to add funds to your account.
|-
! [[Callback]]
|With this feature, you can define a number to be called by our system, in order to receive a dial tone and place outgoing calls through VoIP.ms. This could be useful if you want to place a call and you are not at home or don't have access to your voip device.
|-
! [[Caller ID]]
|Caller ID is a telephone service that transmits the calling party's number to the called party's telephone. When available, the Caller ID number can be complemented with the caller ID name (description e.g. John Smith)
|-
! [[Call Forwarding]]
|A Call Forwarding allows an incoming call to be redirected to a mobile telephone or another telephone number where the desired called party is able to answer. You can set any number, even international numbers.
|-
! [[CallerID Filtering]]
|This feature allows you to filter the incoming calls to your DID numbers given a specific Caller ID number, area code or even anonymous numbers. For example, if you receive annoying incoming calls from a telemarketing company you can create a filter to route all the calls to a recording that plays the message "That number is no longer in service, please hang-up and try again", amongst several other options.
|-
! [[Calling Queues]]
|If you want a solution to manage your incoming calls and have your customer(s) wait on the line while an agent picks up the call, you need to create a Calling Queue entry. This will permit you to have many calls on hold, queued calls in First In-First Out order until agents become available.
|-
! [[Porting a Number|DID Portability]]
|Don't lose that number that you like so much, if you are a VoIP.ms customer and have an existing number with a different provider you can optionally bring it (port it in) to VoIP.ms network. Toll free numbers, local USA/Canada and even International numbers can be ported to VoIP.ms.
The LNP (Local Number Portability) Department handles all porting processes and guides the customer through it. They are also in charge of updating the customer via email with any relevant information about the port every time there is a change on the port request status.
|-
! [[DigitalReceptionist_IVR|Digital Receptionist (IVR)]]
| Interactive Voice Response (IVR), (also known as Auto-attendant) this option can be used to present a recording to the people calling to your DID number and also give them the chance to enter the extension of the person or department they want to reach. For example, you could create an IVR and point one of your DID numbers to it, and when the IVR answers, the caller will hear "Thank you for calling XYZ Inc, for Sales press 1, for Service press 2", etc.
|-
! [[DISA]]
|Direct Inward System Access ( DISA ) allows you to use our system for placing outgoing calls, even if you are not close to any device where you are registering your account or sub account. In this case you just would need to dial to your DID number and to provide a 4 digit PIN number, you can then dial out to any number in the world under our termination rates.
|-
! [[E911]]
|VoIP 911 Service differs from traditional 911 services due to limitations brought on by VoIP technology. It is nearly impossible to detect where a call originates from when placed over the internet. e911 allows users to associate physical addresses with their DIDs, allowing them to have service similar to traditional 911.
'''Important Note: Use of our 911 Service costs a recovery setup fee of $ 1.50 on activation and a regulatory recovery fee of $ 1.50 per DID number activated per month.'''
|-
! Failover for Incoming calls
|Whether your system is unavailable or you are experiencing an Internet or power outage, VoIP.ms offers you the option to redirect the incoming calls to your DID numbers to any location where you can take the call, for example like your cell phone.
Since you are virtually able to select any regular Routing option for the failovers, you can customize your setup in different ways by selecting one of three states, No Answer, Unreachable or Busy.
|-
! Free Calls Between VoIP.ms Customers
|Save even more money with this feature, with VoIP.ms you get free calls from your account to any VoIP.ms Voice DID from US and Canada. So if your friends, family or customers have VoIP.ms DIDs from US/Can, you can be sure that the call will be FREE.
|-
! [[Sub_Accounts#Optional_Settings|Internal Extensions]]
|This feature will allow you to define a quick dial access number to contact the subaccount directly and internally, this way you will be able to reach each a sub account by dialing an extension number E.G "102". This is an amazing feature because calls between extensions are free, so you can have one extension at your office or home in Canada and another one in the US or another country in the world and reach that subaccount dialing only the extension number. With this feature your business can be more efficient and also your friends and family will be in touch easily.
|-
! IP Authentication and IAX support
|Besides the regular method of authentication (Username and password), VoIP.ms also offers the customer the option to authenticate their system via IP address, this way the customer can configure their system to fit their needs.
Even though the native and regular protocol is SIP, VoIP.ms also supports IAX protocol for Customers wanting to communicate this way.
|-
! [[Account_Settings#Notifications|Low Balance Email Notifications]]
|Since it could be time consuming to constantly keep track of the Account balance, VoIP.ms offers you the option to receive an Email notification to your email address, when your account reaches a Low balance limit, below your defined Balance Threshold.
|-
! Music on Hold
|With our services you can easily chose if you want to play "Music on Hold" when the call is on hold state, the caller will hear music from a variety of choices while they wait on line. Also you can deactivate this option if you don't want to play Music to the caller, this can be done from your '''customer portal >> Main Menu >> Account Settings >> General. '''[http://wiki.voip.ms/article/Account_Settings#General]
|-
! [[Phone book]]
|The Phone Book feature allows you to configure Speed-Dial entries and Caller ID name (CNAM) overrides. For example, let's say you have a Customer, Provider or Relative that you call frequently, you can create a phone book entry in order to make a call using a speed-dial entry of 4 digits long. You can also have a Caller ID name (CNAM) override to identify the incoming calls of an important customer if his number doesn't have a proper Caller ID name (CNAM) linked to it, for free.
|-
! [[Recordings]]
|VoIP.ms allows you to upload an audio file and use it in the different options we have under the DID numbers menu. It can be used with several of the other features of our system such as Digital receptionist, Calling queues, and others.
|-
! [[Ring Groups]]
|The Ring Group feature allows you to have incoming calls to be redirected to different destinations that are included in your Ring Group, where any member of the group is able to answer. When you receive a call to a DID routed to a Ring Group, all members of that group will ring at the same time until one of them answers the call. You can add various types of members to a ring group:
Main Account,[[Sub Accounts]],[[SIP URI]]'s,[[Call Forwarding]].
|-
! [[Reseller Basic Guide| Reseller]]
|The Reseller feature is intended for users who want to initiate a business in the VoIP world by reselling VoIP.ms services, this useful tool can help you create and manage accounts for your own clients, you will be able to assign specific rates and DIDs to them and also provide credentials to register a device or call system. You will be able to handle the billing for your customers depending on your needs and how much you want to earn with your business, setting rates and adding credits for them. Additionally you will find options to customize your reseller portal to have a more personalized site.
|-
! [[SIP_URI|SIP URI]]
|A SIP URI is the SIP addressing schema to call another person via SIP. In other words, a SIP URI is a user's SIP phone number or address. The SIP URI resembles an e-mail address.
|-
! [[SMS]]
|
This feature allows you to send and receive messages with your DID Number (US and Canada DID Numbers Only). Currently it is in beta and will remain free until further notice.
|-
! [[Sub Accounts]]
|Having a Sub Account allows you to register more than one device to make or receive calls simultaneously, you can also use it as internal extensions for your office or even your house. Many of the features within VoIP.ms make use of the sub accounts. With this guide we are going to learn how to create and use this feature properly.
|-
! [[Time Conditions]]
|Time Conditions is a feature that allows you to route your incoming calls to different destinations depending on the time of the day of the call. For example, you can have a Time condition for your Non-Business hours and all the incoming calls you receive will be routed to a recording or the voicemail directly (you can also route to other options, like IVR, Call Forwarding entry, etc.).
|-
! Vanity Toll Free Numbers
|Are you looking to have that special contact number for your business with specific digits? VoIP.ms offers the option to order a customized Toll Free number with any combination of digits you like.
Either for business or personal use, you will be able to use the tool from the portal and find that number.
|-
! [[Virtual Fax]]
|The Virtual Fax feature is used for sending and receiving a Fax (facsimile) with the VoIP.ms service using a DID number specifically dedicated to Faxing. You may obtain such a number from your Customer Portal in the Fax Numbers section under the Order DID(s) of the DID Numbers menu. Regular voice DID numbers are not compatible with the Virtual Fax feature.
|-
! [[Voicemail]] ___________________________________________________
|VoIP.ms has an advanced Voicemail feature that is free to use, which also provides you the option to forward your messages to your email address as an audio attachment.
In order to use the Voicemail feature with Voip.ms you will have to create a Voicemail entry and then assign your entry to one of your DIDs or Accounts.
|-
|}

'''If you have more inquires regarding the Features, feel free to contact the support team via the Live Chat or the Ticket system.

==Contact Us==
[[Image:Download.jpg|left|50x75px|link=https://livechat.boldchat.com/aid/2947277729005480016/bc.chat?cwdid=60236691424546376&url=https%3A//www.voip.ms/]][https://livechat.boldchat.com/aid/2947277729005480016/bc.chat?cwdid=60236691424546376&url=https%3A//www.voip.ms/ '''Live Chat Support'''] [[Image:Download.png|50x75px|link=https://www.voip.ms/m/hp/tickets.php]] '''support@voip.ms'''

FAQ

2016-11-08T20:59:18Z

Rod: /* Do you offer Fax service? */

[[File:Logofaq.jpg|250px|]]

<span style="font-size: 120%;">__TOC__</span>

Pour la page en Francais veillez cliquez ici: [[Questions_Les_Plus_Fréquentes|Questions Les Plus Fréquentes]]

Haga clic aquí para Español:[[Preguntas frecuentes]]

==Do you offer a Test Account or Number?==
We currently do not have a test account or test number program however opening an account is completely free and we provide free dialing codes to test the quality of the service.
Please check our many dialing codes here: [[Dialing Codes]]

==How do I start?==
You can open an account for free following this link : https://voip.ms/signup.php
Once your account is active you will be able to log in and find all of our features and options.
We suggest to also check our [[Getting Started]] guide.

==Is there a cost to open the account?==
Opening an account is completely free. The minimum amount of funds you can add is $25 USD which will become your account balance to use the services.

==Is there a Contract?==
There are no contracts with us, you can open an account for free at any time you wish, for as long as you want.

==What are the Payment Options?==
The current options available are Paypal (No Account Needed for Paypal Guest CC Payment), VISA, Master Card, American Express, Discover (Where supported) and via Bank Wire deposit to our bank account in Montreal.
Please contact Customer Support for information about Bank Wire.

==Can I deposit less than $25 into my account?==
Currently $25 is the minimum amount you can add to your account. This credit won’t expire and will last until depleted by using the services.
You can ask for a refund of the remaining balance at any time should you choose to cancel the account.

==Do you charge taxes?==
All customers with Canada as country will pay for the tax called GST. The customers who are also from Quebec in Canada will pay an additional province tax, PST. This applies for both Paypal and Credit Card payments.
The HST will get applied on Canadian provinces that require it.
In order to get further details about how these are applied, feel free to check this link: http://en.wikipedia.org/wiki/Sales_taxes_in_Canada

==How much does the service cost?==
The cost will vary depending on the services used.
For outbound calls you are subject to the per minute rate of the destination you call and the billing increment.
For inbound calls you are subject to the rate plan of the DID, the monthly fee and the billing increment it if applies.
There may be additional monthly fees or set up fees depending on the services you subscribe to.
Please check more details about the service cost here [[Service Cost]]

==What is the Billing Increment?==
The Billing Increment is a call duration measurement unit expressed in seconds.
For example, with a $0.0100 per minute rate, and a 60 seconds billing increment, any call less than 60 seconds is rounded up to 60 seconds or 1 full minute of usage. A 5 seconds call and a 50 seconds call are both billed as 1 minute or $0.0100.

With our 6 seconds billing increment, a 5 second call will be billed as 6 seconds or 1/10 of 1 minute or $0.0010.
This way, you don't get charged for a whole minute just for a 10 seconds call.

==What is a DID?==
A Direct Inward Dial number (DID), in simple terms, is a virtual number that, for all intents and purposes, can be considered a regular phone number, with the exception that it is not attached to any POTS line (Landline). Once your configuration is ready, your DID will be the phone number that everyone in the world will call to reach you, just like any other phone number.

==What type of DID numbers you offer?==
There are different types of DID numbers. They are mostly differentiated by their geographic presence.
Local DIDs from US or Canada.
International DIDs.
Toll Free numbers.

These are the most common used by our customers. For further information about getting a DID number and other types please check our related article [[Order a DID Number]]

==Can I dial out with my DID?==
The DID numbers are a type of service that are exclusively for receiving calls.
When you place outgoing calls with our service you are not calling from the DID number but from the account instead and the number you will pass on your calls as your Caller ID will be the one that is configured on the accounts settings from the portal.

==What is Caller ID?==
Caller ID is a telephone service that transmits the calling party’s number to the called party’s telephone . When available, the Caller ID number can be complemented with Caller ID name (e.g. John Smith).
If you are placing outgoing calls you will likely need to pass a Caller ID to ensure proper termination of your calls, particularly to reach toll free numbers.
There are two types of caller ID and it is important to differentiate them: Caller ID Number (CID) and Caller ID Name (CNAM).
Please contact Technical Support for assistance or more details about Caller ID or check our related article at [[Caller ID]]

==What is CNAM?==
CNAM stands for CallerID Name and it's the information that will be displayed in the phone of the receiving party, when you place an outgoing call.
If you will be making calls to Canadian numbers, you can simply pass the Caller ID name from your device or system as most of them support this.
The Caller ID name on US calls works differently, this is controlled by a national CNAM database with records of numbers and names matching each number.
We can update the CNAM database under request. For more information please check our related article at [[http://wiki.voip.ms/article/Caller_ID#Outgoing_Caller_ID_name Caller ID Name]]

==Can I port my existing number from another provider to VoIP.ms?==
VoIP.ms does offer Local Number Portability (LNP) service and your number may be available for porting. Please contact Customer Service to find out whether your current number is portable to VoIP.ms network or if you already have an open account, please refer to the [[Porting a Number]] guide.

==Do you provide any hardware for the service?==
We don’t provide any kind of hardware device, software or system to use the service. The service is a BYOD (Bring your own device). You should be able to get one from any Communications specialized store and all SIP- compatible devices are supported.

==Can I use my existing device with VoIP.ms?==
Basically any device or system that supports SIP or IAX2 protocol will work with our service . If you bring your device (ATA, IP phone) from a previous provider, make sure it is unlocked and you are able to make changes to its configuration. You can find a list of some devices that we support, with their configuration guides, at [[Devices]]

==Do you have Configuration Samples?==
You can find configuration samples for most of the common devices and phone systems used with VoIP.ms here: [[Devices]]
If your device or system is not listed there but supports the protocol SIP or IAX2 and at least one of the following codecs: G711u, G729a or GMS, you should be able to use it, and you can always contact our Technical Support staff for additional assistance.

==Do you offer Technical Support by phone?==
Due to the nature of the troubleshooting which can result time consuming we chose our main ways to offer support to be via Ticket System (Email) and Live chat.
At this point, Technical Support over the phone is not available.

==What is my Main Account SIP password?==
The main account SIP password is, by default, the same as your customer portal password. If you have not changed the SIP password, it is the same password you use to log into your portal. You can change your SIP password from the Customer Portal at any time from the Main Menu - Account Settings page. Please check our related article for more information: [[http://wiki.voip.ms/article/Account_Settings#Security Account Security]]

==What Server should I use?==
Usually, in order to receive better results, you should choose the server physically closest to your location. You can also send a ping to any of the servers to check the best response time.
Please check our related article for more information about this subject [[Choosing Server]]

==Does VoIP.ms encrypt the communication?==
The SIP communication is secure although not encrypted. However, the passwords are MD5 hashed and are not transmitted without encryption when establishing the call.

==Do you Support e911==
We offer e911 service only for US and Canadian DID numbers (including USA and Canadian Toll-free numbers). This feature can be activated in the Customer Portal directly under the e911 page under the DID numbers menu.
More information available in our related wiki article here [[E911]]

==Do you offer Calling Card?==
We do not offer any Calling Card feature at this moment, however you can set up a Calling Card solution with your own system and our DID numbers as long as this is used with DID numbers in our Per Minute plan.
For more information please contact Technical Support.

==Do you offer Fax service?==
We offer a Virtual Fax feature. This feature can be used with special Fax DID numbers which can be acquired and configured through our Fax portal.
Please check our related article for further details: [[Virtual Fax]]

==Do you offer SMS service?==
We do offer SMS service (currently on Beta). The SMS is available for a great number of DIDs and Cities from US and Canada and requires to be used through the SMS portal.
SMS is not available through SIP protocol for the moment.
You can read more details about this feature in our related article [[SMS]]

==Do you offer Conference Calls?==
We do not offer Conference calls as a feature in the customer portal, however if your system or device is capable of establishing a conference call or possess this feature, you can use it with our services.

==Do you support TCP for the SIP communication?==
TCP is fully supported in our servers.

==Do you offer alternative ports besides 5060?==
We offer alternative SIP ports, UDP/TCP 5080 and 42872 on all of our servers, You can try those ports in case your Internet Service Provider blocks the port 5060 UDP/TCP or if you need to use another one.

==Can I register 2 or more different devices with the same account username?==
It is strongly suggested not to do so, this can possibly cause conflicts while routing the calls to your device as well issues related with registration. If you need to register more than one device please create and use the Sub Accounts, you will get new credentials for any additional device.
For more information about Sub Accounts please check the related article on this subject [[Sub Accounts]]

==Do you accept Autodialers or Telemarketing traffic ?==
We do not accept the use of our termination (outbound) services for telemarketing purposes (Including but not limited to Automated Dialers, Call Centers and collection agencies).

==Do you offer service to Call Centers ?==
We currently provide service to Call Centers for inbound traffic, the use of this is restricted to DID numbers with a per minute plan.
Outbound traffic from call centers is not supported at this time. If you have any concern about this please contact Customer Support for more details.

==Can I resell your services ?==
Yes, our customers will be able to find a "Reseller Section" included in the Customer Portal.
We give you all the tools you need plus a White label Reseller Interface that has been developed to help our clients upsell our services under their own brand. We also recommend that you must familiarize yourself with the Voip.ms interface before reselling the service.
For more information about reseller please contact Technical Support or visit our related article here: [[Reseller Basic Guide]]

==Do you offer Wholesale Rates ?==
If you are interested on a discount based on traffic usage or volume, please send an email to sales@voip.ms providing all details about your traffic like destinations you need to call and average of minutes used per month, in order to receive additional information and quotes.

==Can I port out my number ?==
If you wish to port out a number from our service you can do so at any time by starting the porting request with the new provider. We authorize all Port Out Request matching the correct information from your account.
Remember to delete the DID number from your VoIP.ms account once the port out process is completed.
For any question regarding Port Out please send an email to our LNP Department at ports@voip.ms

==How do I delete my DID ?==
From the customer portal, go to the “DID Numbers” menu tab and then click “DID Billing”. From this page, you can cancel number(s). Remember this process cannot be reversed, please be sure that you will no longer need the number.

==Is there any Cancellation Fee ?==
We do not charge any cancellation fee of any kind, you are free to cancel your account or number at any time you wish to.

==Where are you located ?==
VoIP.ms is a Canadian Company founded back in 2007 in Montreal.
The main office is located in Montreal, QC, Canada and there is also a Technical Site in Merida, Yucatan, Mexico for the South America market.

==Do you offer a Referral Program ?==
For the moment we don't have any kind of referral program available.

[[Category: Guides]]

Virtual Fax

2016-11-08T20:58:33Z

Rod:

[[File:Faxhomelogo.png|center]]

The Virtual Fax feature is used for sending and receiving a Fax (facsimile) with the VoIP.ms service using a DID number specifically dedicated to Faxing. You may obtain such a number from your Customer Portal in the Fax Numbers section under the ''Order DID(s)'' of the ''DID Numbers'' menu.
Regular voice DID numbers are not compatible with the Virtual Fax feature.

__TOC__

== Important information about the Virtual Fax Service ==

* '''The Virtual Fax Service is only available for U.S. and Canadian DID Numbers specifically acquired from the Fax Numbers ''Order DID'' section'''
* '''It is also possible to port your Voip.ms Voice DID Numbers and Numbers from other Providers into our Virtual Fax service. For numbers from other providers, you can find this option under the ''DID Portability'' section of the Customer Portal. For Voip.ms numbers, you can request an internal port by sending an email to our LNP department at ports@voip.ms. The porting fee is $15 per number for both options.'''
* '''The Service can only be used to send Faxes to Canadian and U.S. Numbers at this time. We also cannot guarantee that International will be properly received.'''
* '''Virtual Fax DID Numbers cannot receive regular voice calls nor SMS, only faxes.'''

== Cost and Rates ==

Setup Fee: $0.00 (Currently Free)

Monthly Fee: $1.99

Per Minute Fee: $0.0290 (2.9 Cents)

[[Calls Cost|Billing Increment]] : 6 seconds

If you wish to calculate how much this service could cost, per page faxed, converted into the time of per minute charges you will want to take into consideration that this depends a lot on the destination fax speed and the content (How much content is on the page compared to the blank space) being faxed. Unfortunately there are no definite ways in calculating costs. If you are sending just text, a page can go from 30 secs up to 1 min. If you are faxing multiple page documents this will go faster since you are not having to handshake or negotiate with the far side's fax machine for each page. This can result in 1.5 cents to 3 cents on a 1-2 page fax. Since we also increment the charge to every 6 seconds you are saving even more money since we do not charge you a full min for partial minute usages. You can usually get around 1-2 pages faxed per minute at 2.9 cents and $2 a month which is much less than most other Electronic Fax services being offered at this time.

== Current Limitations ==

*Each DID Number can only send 100 messages per day. This limit can be raised upon request and verification.

*The files sent per message cannot exceed 25mb

== Virtual Fax DID Number ==

Virtual Fax works specifically with Fax Numbers only acquired from the VoIP.ms Customer Portal or numbers ported in specifically as Fax enabled. There are '''local US and Canadian numbers''' available for order. You can order a Fax DID Number from your portal >> DID Numbers >> Order DID >> Fax Numbers. You can select the desired region and a random number from the chosen area code will be assigned to you.

[[File:FaxorderDID2.jpg|700px|]]

You can also port in a number you currently own with another provider. This process can be started from the Customer Portal at DID Numbers >> DID Portability >> Porting Fax Numbers

[[File:FaxPortability.png|700px]]

== Send a Fax ==

To start using the service you will need to head to your Customer Portal >> DID Numbers >> Virtual Fax. From the Home Page you can select ‘Send Fax’. There you will see:
*Fax Number or Contact Name: This is where you will put the destination number. You can start typing a name or a number from your Phone book and it will become available.
*From Name: Here you will put the name to send in the Fax header.
*From Number: Select the Fax DID number from which you will send your Fax.
*File: Choose a file to send as a Fax. The file must be in pdf, txt, jpg, gif, png or tif
-'''IMPORTANT: The maximum file size is 25MB'''
*Station ID: This will be the station ID you set for the header of the Fax message. It could be a specified post if your location has several stations, such as Reception, Main Office, Accounting PC, etc.
*Send Email: If selected, an email will be sent to the specified address to confirm the Fax has been sent successfully or to advise of a failed attempt.

[[File:SendFax.png|900px]]

== My Faxes ==
In this section of the Virtual Fax menu you will be able to view your Inbound and Outbound Faxes. You may select a date range and choose the folder you would like to view. Click 'Get My Faxes' to view your selection. You can view the Status of each Fax and select from several Actions. You can select to View the Fax directly, Download the Fax, Email the Fax to an address of your choosing or alter the location of the Fax by moving it to another folder.

Only the Status gets updated automatically. You will have to refresh the page to get the costs after a fax completes.

[[File:MyFaxes.png|900px]]

== My Folders ==

In the 'My Folders' section you can create folders by typing in the folder name of your choosing under 'New Folder' and clicking 'Create'.
You will have an overview of your Folders, see the date they were Created, the amount of Faxes in each Folder and be able to Edit the Folder or Delete it.
Any Faxes contained in a created folder will revert back to either the INBOX or SENT folder if the created folder is deleted.

[[File:MyFolders.png|900px]]

== My Fax Numbers ==
In the 'My Fax Numbers' section you will see your Fax DID numbers and Description, the Options that have been enabled for each number, the Email address if one has been configured along with the URL if configured in the URL Callback section. You can Edit the number from the 'Actions' section or choose to Delete it. When editing you will have the option to set an Email Address to receive a notification when a new Fax is received (you can also select to have the PDF file attached in the email) and set a URL Callback (you can also enable URL Callback Retry).

[[File:MyFaxNumbers.png|900px]]

== Email to Fax==

This feature allows you to send a Fax message using your email account.

How to send a Fax message using your email account:
*Use the email account you provided when enabling the Email to Fax service.
*Send the email to fax@voip.ms
*In the subject field type the Destination Fax Number (Example: 5148000000).
*Attach the document you wish to send to the email message. VoIP.ms supports the following formats: pdf, txt, jpg, gif, png, tif.
*Send the email.
**'''NOTE: Only the file attached will be faxed, the body of the e-mail won't be transmitted'''

'''Security Code and From Number:'''

If Security Code is enabled, you need to add a dot (.) and the Security Code after the Destination Fax Number (Example: 5148000000.Az09).

If you have more than one Fax number, you could change the From Number by adding a dot (.) and the From Number you'd like to use after the Destination Fax Number and the Security Code (Example: 5148000000.Az09.2268280000).

If you have not enabled the Security Code and want to change the From Number, you can add a dot (.) and the From Number after the Destination Fax Number (Example: 5148000000.2268280000).

[[File:EmailToFax.png|900px|]]

FAQ

2016-10-25T14:26:10Z

Rod: /* Do you support TCP for the SIP communication? */

[[File:Logofaq.jpg|250px|]]

<span style="font-size: 120%;">__TOC__</span>

Pour la page en Francais veillez cliquez ici: [[Questions_Les_Plus_Fréquentes|Questions Les Plus Fréquentes]]

Haga clic aquí para Español:[[Preguntas frecuentes]]

==Do you offer a Test Account or Number?==
We currently do not have a test account or test number program however opening an account is completely free and we provide free dialing codes to test the quality of the service.
Please check our many dialing codes here: [[Dialing Codes]]

==How do I start?==
You can open an account for free following this link : https://voip.ms/signup.php
Once your account is active you will be able to log in and find all of our features and options.
We suggest to also check our [[Getting Started]] guide.

==Is there a cost to open the account?==
Opening an account is completely free. The minimum amount of funds you can add is $25 USD which will become your account balance to use the services.

==Is there a Contract?==
There are no contracts with us, you can open an account for free at any time you wish, for as long as you want.

==What are the Payment Options?==
The current options available are Paypal (No Account Needed for Paypal Guest CC Payment), VISA, Master Card, American Express, Discover (Where supported) and via Bank Wire deposit to our bank account in Montreal.
Please contact Customer Support for information about Bank Wire.

==Can I deposit less than $25 into my account?==
Currently $25 is the minimum amount you can add to your account. This credit won’t expire and will last until depleted by using the services.
You can ask for a refund of the remaining balance at any time should you choose to cancel the account.

==Do you charge taxes?==
All customers with Canada as country will pay for the tax called GST. The customers who are also from Quebec in Canada will pay an additional province tax, PST. This applies for both Paypal and Credit Card payments.
The HST will get applied on Canadian provinces that require it.
In order to get further details about how these are applied, feel free to check this link: http://en.wikipedia.org/wiki/Sales_taxes_in_Canada

==How much does the service cost?==
The cost will vary depending on the services used.
For outbound calls you are subject to the per minute rate of the destination you call and the billing increment.
For inbound calls you are subject to the rate plan of the DID, the monthly fee and the billing increment it if applies.
There may be additional monthly fees or set up fees depending on the services you subscribe to.
Please check more details about the service cost here [[Service Cost]]

==What is the Billing Increment?==
The Billing Increment is a call duration measurement unit expressed in seconds.
For example, with a $0.0100 per minute rate, and a 60 seconds billing increment, any call less than 60 seconds is rounded up to 60 seconds or 1 full minute of usage. A 5 seconds call and a 50 seconds call are both billed as 1 minute or $0.0100.

With our 6 seconds billing increment, a 5 second call will be billed as 6 seconds or 1/10 of 1 minute or $0.0010.
This way, you don't get charged for a whole minute just for a 10 seconds call.

==What is a DID?==
A Direct Inward Dial number (DID), in simple terms, is a virtual number that, for all intents and purposes, can be considered a regular phone number, with the exception that it is not attached to any POTS line (Landline). Once your configuration is ready, your DID will be the phone number that everyone in the world will call to reach you, just like any other phone number.

==What type of DID numbers you offer?==
There are different types of DID numbers. They are mostly differentiated by their geographic presence.
Local DIDs from US or Canada.
International DIDs.
Toll Free numbers.

These are the most common used by our customers. For further information about getting a DID number and other types please check our related article [[Order a DID Number]]

==Can I dial out with my DID?==
The DID numbers are a type of service that are exclusively for receiving calls.
When you place outgoing calls with our service you are not calling from the DID number but from the account instead and the number you will pass on your calls as your Caller ID will be the one that is configured on the accounts settings from the portal.

==What is Caller ID?==
Caller ID is a telephone service that transmits the calling party’s number to the called party’s telephone . When available, the Caller ID number can be complemented with Caller ID name (e.g. John Smith).
If you are placing outgoing calls you will likely need to pass a Caller ID to ensure proper termination of your calls, particularly to reach toll free numbers.
There are two types of caller ID and it is important to differentiate them: Caller ID Number (CID) and Caller ID Name (CNAM).
Please contact Technical Support for assistance or more details about Caller ID or check our related article at [[Caller ID]]

==What is CNAM?==
CNAM stands for CallerID Name and it's the information that will be displayed in the phone of the receiving party, when you place an outgoing call.
If you will be making calls to Canadian numbers, you can simply pass the Caller ID name from your device or system as most of them support this.
The Caller ID name on US calls works differently, this is controlled by a national CNAM database with records of numbers and names matching each number.
We can update the CNAM database under request. For more information please check our related article at [[http://wiki.voip.ms/article/Caller_ID#Outgoing_Caller_ID_name Caller ID Name]]

==Can I port my existing number from another provider to VoIP.ms?==
VoIP.ms does offer Local Number Portability (LNP) service and your number may be available for porting. Please contact Customer Service to find out whether your current number is portable to VoIP.ms network or if you already have an open account, please refer to the [[Porting a Number]] guide.

==Do you provide any hardware for the service?==
We don’t provide any kind of hardware device, software or system to use the service. The service is a BYOD (Bring your own device). You should be able to get one from any Communications specialized store and all SIP- compatible devices are supported.

==Can I use my existing device with VoIP.ms?==
Basically any device or system that supports SIP or IAX2 protocol will work with our service . If you bring your device (ATA, IP phone) from a previous provider, make sure it is unlocked and you are able to make changes to its configuration. You can find a list of some devices that we support, with their configuration guides, at [[Devices]]

==Do you have Configuration Samples?==
You can find configuration samples for most of the common devices and phone systems used with VoIP.ms here: [[Devices]]
If your device or system is not listed there but supports the protocol SIP or IAX2 and at least one of the following codecs: G711u, G729a or GMS, you should be able to use it, and you can always contact our Technical Support staff for additional assistance.

==Do you offer Technical Support by phone?==
Due to the nature of the troubleshooting which can result time consuming we chose our main ways to offer support to be via Ticket System (Email) and Live chat.
At this point, Technical Support over the phone is not available.

==What is my Main Account SIP password?==
The main account SIP password is, by default, the same as your customer portal password. If you have not changed the SIP password, it is the same password you use to log into your portal. You can change your SIP password from the Customer Portal at any time from the Main Menu - Account Settings page. Please check our related article for more information: [[http://wiki.voip.ms/article/Account_Settings#Security Account Security]]

==What Server should I use?==
Usually, in order to receive better results, you should choose the server physically closest to your location. You can also send a ping to any of the servers to check the best response time.
Please check our related article for more information about this subject [[Choosing Server]]

==Does VoIP.ms encrypt the communication?==
The SIP communication is secure although not encrypted. However, the passwords are MD5 hashed and are not transmitted without encryption when establishing the call.

==Do you Support e911==
We offer e911 service only for US and Canadian DID numbers (including USA and Canadian Toll-free numbers). This feature can be activated in the Customer Portal directly under the e911 page under the DID numbers menu.
More information available in our related wiki article here [[E911]]

==Do you offer Calling Card?==
We do not offer any Calling Card feature at this moment, however you can set up a Calling Card solution with your own system and our DID numbers as long as this is used with DID numbers in our Per Minute plan.
For more information please contact Technical Support.

==Do you offer Fax service?==
We offer a Virtual Fax feature (currently on Beta). This feature can be used with special Fax DID numbers which can be acquired and configured through our Fax portal.
Please check our related article for further details: [[Virtual Fax]]

==Do you offer SMS service?==
We do offer SMS service (currently on Beta). The SMS is available for a great number of DIDs and Cities from US and Canada and requires to be used through the SMS portal.
SMS is not available through SIP protocol for the moment.
You can read more details about this feature in our related article [[SMS]]

==Do you offer Conference Calls?==
We do not offer Conference calls as a feature in the customer portal, however if your system or device is capable of establishing a conference call or possess this feature, you can use it with our services.

==Do you support TCP for the SIP communication?==
TCP is fully supported in our servers.

==Do you offer alternative ports besides 5060?==
We offer alternative SIP ports, UDP/TCP 5080 and 42872 on all of our servers, You can try those ports in case your Internet Service Provider blocks the port 5060 UDP/TCP or if you need to use another one.

==Can I register 2 or more different devices with the same account username?==
It is strongly suggested not to do so, this can possibly cause conflicts while routing the calls to your device as well issues related with registration. If you need to register more than one device please create and use the Sub Accounts, you will get new credentials for any additional device.
For more information about Sub Accounts please check the related article on this subject [[Sub Accounts]]

==Do you accept Autodialers or Telemarketing traffic ?==
We do not accept the use of our termination (outbound) services for telemarketing purposes (Including but not limited to Automated Dialers, Call Centers and collection agencies).

==Do you offer service to Call Centers ?==
We currently provide service to Call Centers for inbound traffic, the use of this is restricted to DID numbers with a per minute plan.
Outbound traffic from call centers is not supported at this time. If you have any concern about this please contact Customer Support for more details.

==Can I resell your services ?==
Yes, our customers will be able to find a "Reseller Section" included in the Customer Portal.
We give you all the tools you need plus a White label Reseller Interface that has been developed to help our clients upsell our services under their own brand. We also recommend that you must familiarize yourself with the Voip.ms interface before reselling the service.
For more information about reseller please contact Technical Support or visit our related article here: [[Reseller Basic Guide]]

==Do you offer Wholesale Rates ?==
If you are interested on a discount based on traffic usage or volume, please send an email to sales@voip.ms providing all details about your traffic like destinations you need to call and average of minutes used per month, in order to receive additional information and quotes.

==Can I port out my number ?==
If you wish to port out a number from our service you can do so at any time by starting the porting request with the new provider. We authorize all Port Out Request matching the correct information from your account.
Remember to delete the DID number from your VoIP.ms account once the port out process is completed.
For any question regarding Port Out please send an email to our LNP Department at ports@voip.ms

==How do I delete my DID ?==
From the customer portal, go to the “DID Numbers” menu tab and then click “DID Billing”. From this page, you can cancel number(s). Remember this process cannot be reversed, please be sure that you will no longer need the number.

==Is there any Cancellation Fee ?==
We do not charge any cancellation fee of any kind, you are free to cancel your account or number at any time you wish to.

==Where are you located ?==
VoIP.ms is a Canadian Company founded back in 2007 in Montreal.
The main office is located in Montreal, QC, Canada and there is also a Technical Site in Merida, Yucatan, Mexico for the South America market.

==Do you offer a Referral Program ?==
For the moment we don't have any kind of referral program available.

[[Category: Guides]]

Porting FAQ

2016-10-24T17:41:16Z

Rod: /* What are all of these acronyms: LOA, FOC, BTN, LSP,CSR, LSR, PON? */

Haga clic aquí para la versión en Español:[[Preguntas_frecuentes_LNP|Preguntas frecuentes LNP]]
__TOC__

'''Please also check out our other Wiki Guides in Regards to Porting:'''
*[http://wiki.voip.ms/article/Porting_a_Number Porting a Number]
*[http://wiki.voip.ms/article/Porting_a_US_Number Porting a US Number]
*[http://wiki.voip.ms/article/Porting_a_Canadian_Number Porting a Canadian Number]
*[http://wiki.voip.ms/article/Porting_a_Toll_Free_Number Porting a Toll Free Number]
*[http://wiki.voip.ms/article/Port_Rejection Port Rejections]

== How much does it cost to Port a Number to VoIP.ms? ==

There is a $8.50* USD one time fee for porting each number into VoIP.ms service. This is what VoIP.ms is charged and there is no profit made.
Sometimes the underlying Carrier will offer free ports and you will be advised, if this is so, when you port your number.

(*Holiday Exchange Rate Special will be available for a Limited Time.)

== How to port my Google Voice Number to VoIP.ms ==

You will first want to make sure that your number is portable to our service by checking here [https://www.voip.ms/portability.php Number Portability] then please follow this guide to [https://support.google.com/voice/answer/1316844?hl=en Unlock your Google Voice Number]

== How do I start the process to port my number(s)? ==
To start the porting request you can log in on your portal -> DID Numbers -> DID Portability and click on Start Procedure to start your porting order.

== What type of information will you need in order to port my number(s)? ==
The first thing you will need to provide to us is the Service Address of the number(s) you wish to port. This is the address where you will be using the number(s) and where you would like your E911 services to be listed. Even virtual numbers have a service address listed on the losing carrier’s side.

The service address will help us to fill up the Letter of Authorization (LOA). This form will be sent to our carrier to process your order.

In order to fill out this form, we will need the following information from you:
Company Name (if applicable), Service Address (as it is listed with your current carrier), Service Provider Name, Number to port, Billing Telephone Number (BTN):
This is the main billing number on your account that all of your numbers may be listed under. Please note that it is possible to have several different BTN's depending on how your account is set up.

Also, you will need to submit a hand-signed copy of your most recent billing invoice from your current phone provider.

All of the information needed for filling out the LOA can be supplied by your current phone provider by requesting a Customer Service Record (CSR).

== My current carrier does not provide any invoices, what else can I submit? ==

If your carrier does not provide billing invoices, you can submit a screen shot from the losing carrier’s online portal that shows the customer name, number to port and account information. The screenshot must also be printed, hand-signed and sent to us.

== What type of numbers can I port? ==
You can port any local, virtual and even a wireless number as long as we have coverage for that particular area. You can also port your Toll Free numbers and Fax numbers, although Fax is only available for US and Canada numbers.
Please note that Pager numbers are non-portable.

== My number is not portable to your network, even though I am close to a city which is portable to you, why is that? ==

The portability of a number depends on our carriers to cover that city (or rate center which is where the number actually belongs) and some towns/cities are not available no matter how close they are to covered cities. In some cases, you can purchase a number from a certain city but it is not available for porting. This again depends on our carriers’ coverage and it is out of our hands.

Some numbers are simply not portable. These include anything from pagers to numbers from a few small rural independent telephone companies to "shared use" toll-free numbers where the customer only has use of the number in one city or one area code. A few loopholes exist for a provider to claim they, not you, "own" a number

No ICA. The provider that you want to port to does not have an Interconnect Agreement in place with the telephone company that holds the number.

Our managers are always looking for new carriers to cover more rate centers and also our current carriers are always expanding their coverage; we suggest that you come back in the future and verify if your rate center is available for porting. You may do so by logging in on your portal -> DID Numbers -> DID Portability -> Check Availability

== How long does the port process take? ==

On average porting requests take from 2 to 4 business weeks to complete for US numbers and on average 1 business week for Canadian numbers; however, this can change depending always on the losing carrier’s cooperation level and also if there is any holiday in between. Please keep in mind any type of change or rejection can delay this process. This is why it is very important to verify the porting information prior to submitting it.

== What happens behind the scenes while my number is being ported? ==

Once your order has been submitted and received, you will be notified via email. After verifying your order, we will submit your port request with the information you provided to our carrier. You will be notified of its progress/status via email. If the request is rejected, we will provide you with instructions on how to resolve the issue.
Once we have received a FOC date, you will again be notified via email. Please keep in mind any changes made to a port request can push the FOC date further out.

== What happens if my port request gets rejected? ==

In the event that we receive a rejection from your carrier, we will provide you with instructions on how to resolve the issue. You can check the different and most common rejections by clicking on our Wiki Site [[Port Rejection]].

== How can I expedite my port request? ==

Unfortunately it is very hard for us to expedite a port request. The reason for this is because we have no control over the FOC date, each carrier has its own proper time to process an order. This date is given by the losing end and the FCC (for USA) and the CRTC (for Canada) mandate that we can’t take a number unless the current provider gives us permission.

However, if you provide us the correct information as it is listed on the CSR, we will do everything we can to expedite the process from our side.

== Will I experience any downtime while my numbers are porting? ==

On the day of your port, at some point, your number(s) will be triggered in our system. You should not experience any downtime as long as you have set up everything as you need it and properly routed the phone number(s) in your VoIP.ms portal. You will be able to view your number(s) in your VoIP.ms portal prior to the day of port in order to give you time to set everything up. Please contact Support via Live Chat or Ticket if you need help setting up everything before your number ports. However, it is important to also note that some downtime may occur during the porting day as the number(s) is(are) changing provides and routes.

== When/How will I know my number(s) has/have ported? ==

After your number(s) has/have completed porting, we will make test calls to make sure your number(s) is/are working properly. Once we have verified calls are completing on our network, you will be notified via email that the porting was a success. If there are any problems while test calls are being made, we will work with you until the issue has been resolved.

== My previous carrier is charging me for porting out and/or asks for a 30 day notice, how can I deal with this? Is it legal? ==

Unfortunately we have no control over the fees the losing carrier charges for porting out their numbers. Prior to submitting an order with us, please verify with them if there is any fee involved.

Regarding the 30 day notice, this is usually for Pay-as-you-go services. Since you were not charged for 30 days from the day you started your phone service with your carrier, you are responsible for payment of the 30 days following the cancellation of your account (which, in most cases, occurs due to number porting). This has always been stated in the terms of service of almost every carrier.
Some carrier contracts mention that the customer will “Pay for 30 days after the transfer is requested", and not after notice is given (by customer). The one who requests the transfer is the new carrier.
Unfortunately the CRTC (or FCC) do not regulate phone company policies.

== What is a RespOrg? ==

The port for Toll Free numbers is a bit different from porting local numbers. RespOrg is the communication carrier responsible for managing your Toll free number(s) and they use the SMS800 system to do so.

== What is a NASC procedure? ==

A NASC is a forced change of carriers done through the Number Administration Service Center. For a per-number NASC fee and the most recent invoice that is hand signed by the customer, the number is moved without the cooperation of the current carrier. The timeline for NASCing is 24 to 72 hours. Customers must also be aware that NASCing may only be a temporary fix; their previous carrier could NASC the number back at any time; although, this rarely happens as there is a NASC fee involved.

Sometimes it becomes necessary to utilize the NASC to port your Toll Free number to us. This can be caused when a previous RespOrg will not transfer a toll-free number to us. Often, a current RespOrg can reject the transfer for legitimate reasons (such as billing issues) or simply to stop the process.
However, if the business or customer that uses the number is in jeopardy of losing its 800 number or sales because of the number potentially being disconnected, NASCing is a way of forcing the process through.

''Please note that this process is only for USA and Canada Toll Free numbers''

== What are all of these acronyms: LOA, FOC, BTN, LSP, CSR, LSR, PON? ==

LOA - Letter of Authorization. This is the form we will need to fill out when porting your number(s). This form gives us permission to request, on your behalf, to port your number(s) to a different phone provider.

FOC - Firm Order Commitment. This is the date that is issued by the losing carrier as to when a phone provider will complete the port of a number(s). This date can only be issued by the losing carrier. The gaining phone provider has no control over what date is issued.

BTN - Billing Telephone Number. This is the main telephone number listed on your account. Depending on what type of account you have, you could have just one BTN or you could have several. When porting numbers to VoIP.ms you want to be sure to ask your current phone provider what the BTN(s) is/are for the number(s) you wish to port.

LSP - Losing Service Provider. This is the phone provider that your number(s) is/are leaving from.

CSR - Customer Service Record. Every carrier has a record that has your number(s), address, and Billing Telephone Number (BTN)(s). This is the record that is used to verify your information during the porting process. Please note that the information listed on the CSR may not be the same information listed on the billing portion of your account.

LSR - Local Service Request. This is the actual order request that the losing carrier will receive from the gaining provider. All of the information on this request must match the losing carrier’s (CSR) in order for them to issue a FOC date.

PON – Porting Order Number. This is the confirmation number that is created once the order has been sent to the losing carrier.

== I am being asked for my PIN, SSN and Password, IMEI why is that? ==

When porting a wireless / cell number, some extra information is needed. For US numbers you must provide the last 4 digits of the SSN (Social Security Number) and the PIN/Password on the current account. If your wireless number is prepaid you might have either one; however, if you are using a pay-as-you go service, you have to have both.
For Canadian Wireless numbers, we will need only your account number; unless the account number and the number porting are the same (meaning it’s a prepaid account); then we will need the IMEI, PIN and SIMM card (this one is not always available but the first 2 are).
Some carriers don't ask for this information when subscribing with them; however, our carrier does need this information to process your order.
The PIN stands for Personal Identification Number (PIN) which it's a 4-digit number combination known only by you. PIN numbers are often used for cell phones to lock the phone. This lock keeps your private information secure and makes your phone useless to anyone who steals it or finds it if you misplace it.

== What is the IMEI? ==

The IMEI is the International Mobile Equipment Identity is a number, usually unique, to identify mobile phones, as well as some satellite phones. It is usually found printed inside the battery compartment of the phone.
To retrieve the IMEI, enter *#06# and the IMEI will be displayed on your phone.

== How do I Port Out my number ? ==

If you have a DID number with VoIP.ms and wish to port it out to another provider, you need to always start this process on the new provider's side. They should advise you about what information to provide to start this process.
Remember to submit always the information as listed on your Contact Information section.

Canadian local and Toll Free port out numbers are authorized as long as the information matches with what is on your Contact Information section and you have positive balance. Please avoid deleting the number since in most cases the number is not ported yet and if you delete it, you may lose it; after the number is ported out from us, our LNP Staff will email you about it and will also delete the number from your account.

For US numbers, please make sure you delete the number after it's ported and tested with the new carrier and mark the 'Ported Out' box. This is to avoid any extra charges as, in some cases, we are not advised when a number is ported out.

Please open a ticket to our LNP Department if you get a rejection when porting out your number.

== When Porting Out my number the new carrier requests the PIN, what is my PIN? ==

Please note that numbers that are ported into us, or purchased from our portal, are not assigned any passwords/PINs. This could be a system rejection that the gaining carrier provides before sending the LSR to our underlying carrier.
If this number is being ported to a wireless provider, they may just require the customer to make up a PIN to port the number.

== My current carrier says the number has already been released or the order has not been received on their side yet. ==

The porting process is dealt between CLECs and (sometimes) resellers. However, some resellers are not aware of the port, not even when the number has left their side. We have no reason why we should not send an order as long as the customer provides the proper information. We will submit the order, most likely on the same day; however, since our carrier works as well with some other upstream carriers the order might take some time to be processed.
It is not odd that, if your order was rejected, your current carrier won’t see it since most carriers don’t save records of rejected orders.
Some carriers also say that the number has already been released/ported when in fact it has not. Our system is always monitoring when a number has been released without prior notice, if your number was in fact released, we will be notified about it and we will email you immediately after adding the number to your account to avoid any missed calls.

== I need my number to be ported on weekend or on a specific date, can I request this? ==

LNP Departments are closed on weekends; therefore no numbers can be ported during Saturdays or Sundays, nor holidays. If you need a specific date, you can leave a note when filling your porting request and we will ask our carrier if such date can be met. Please remember that this is not always possible; however, we will email you on any updates.

== My port is now processing or has a FOC date, can I now cancel my number with the losing carrier to avoid any extra charges? ==

It is important to note that you must wait until the number is fully ported to our system (we will email you regarding so) before you can cancel the account with the old carrier. If the order gets cancelled before the number is fully ported, you might get a rejection or even lose the number. Please be cautious.

== Why are purchased numbers cheaper than ported numbers? ==

This is because porting a number involves different carriers that must be paid for the transfer. Also, it is easier to purchase a number and configure it than to transfer an existing number from another carrier.

== My number has been disconnected by my current carrier and I am not receiving any calls, has it ported? ==

Please note that the number will always belong to the current provider until we email you regarding its completion. The number must be active at all times during the porting process. Disconnected numbers are not portable. If the losing carrier disconnected the number claiming that it has ported, you must verify if you have received a completion notice from us. If not, you must contact the current carrier and ask them to reactivate the number as soon as possible. Remember that the number is still on their end, since it has not ported.
Some carriers may disconnect the number just to stop the port and then activate it again on their side.

== My porting process has been completed and I have enabled SMS on my number, why can't I receive messages yet? ==

The SMS feature can take up to 48 hours (24 hours in average) after the porting process has been completed to become fully active. Please allow this much time before fully testing the SMS functionality on your recently ported in number.

== I am porting many numbers, can I get a special price for it? ==

Yes, prior to submit any porting request for many numbers, you should open up a ticket to our Sales Department ( Sales@voip.ms ) requesting the special price for the batch of numbers. Please note that the special price will only apply as long as all the numbers on the same batch have the same carrier, same customer information and same account information.
If the information is somehow different, you will need another quote.

== Now that the number has been ported to VoIP.ms, who cancels the old account? ==

Once we confirm the number has been successfully ported to us, check to see if your account is in fact closed and you are not continuing to be charged by the old carrier. If the account wasn't closed, please contact the losing carrier and ask them to cancel your old account if it's no longer useful to you anymore. This is to avoid any extra charges, as some carriers still show the ported number on their end, even when the number was ported out.
Please note that the cancellation can only be done by the customer since he/she is the one paying for the old carrier's services. We can't contact them nor cancel any services/lines with them.
Some carriers might say otherwise just to keep on billing you. Please demand the cancellation of the old account if it's no longer useful to you.
Note that neither your account with us nor the number will be affected by this cancellation, since they are now with us.

== I currently have a CNAM on my number, when porting will I keep it? ==

If you have a CNAM with the current carrier, when porting your number, it is most likely that the current CNAM is deleted when the transfer is completed, showing some random word such as the number itself, customer's first or last name or simply “unknown”. In some special cases the CNAM is kept as it was.

To avoid this, please verify if your number can have CNAM updated before starting the porting process. If so, after the number is ported, please test your number to verify your CNAM values and if they are not what you wish, contact our Support Staff ( Support@voip.ms ) to have the CNAM updated.

== At what point and how can I configure my porting number? ==

When the FOC is received, your number will be added to our system. From that point you can configure it and prepare it for the day the port completes. Your number will be reachable from anyone on our network at this time. Remember that if you need further assistance with your configuration, you can always open a ticket to our Support Staff who will gladly help you.

[[Category: Guides]]

Virtual Fax

2016-10-21T14:37:46Z

Rod:

[[File:Faxhomelogo.png|center]]

The Virtual Fax feature is used for sending and receiving a Fax (facsimile) with the VoIP.ms service using a DID number specifically dedicated to Faxing. You may obtain such a number from your Customer Portal in the Fax Numbers section under the ''Order DID(s)'' of the ''DID Numbers'' menu.
Regular voice DID numbers are not compatible with the Virtual Fax feature.

__TOC__

== Important information about the Virtual Fax Service ==

* '''The Virtual Fax Service is in BETA version, that means it is not fully deployed. It is important to us that you report any issues with this service by sending an email to support@voip.ms so that the developers can get involved if necessary.'''
* '''The Virtual Fax Service is only available for U.S. and Canadian DID Numbers specifically acquired from the Fax Numbers ''Order DID'' section'''
* '''It is also possible to port your Voip.ms Voice DID Numbers and Numbers from other Providers into our Virtual Fax service. For numbers from other providers, you can find this option under the ''DID Portability'' section of the Customer Portal. For Voip.ms numbers, you can request an internal port by sending an email to our LNP department at ports@voip.ms. The porting fee is $15 per number for both options.'''
* '''The Service can only be used to send Faxes to Canadian and U.S. Numbers at this time. We also cannot guarantee that International will be properly received.'''
* '''Virtual Fax DID Numbers cannot receive regular voice calls nor SMS, only faxes.'''

== Cost and Rates ==

Setup Fee: $0.00 (Currently Free)

Monthly Fee: $1.99

Per Minute Fee: $0.0290 (2.9 Cents)

[[Calls Cost|Billing Increment]] : 6 seconds

If you wish to calculate how much this service could cost, per page faxed, converted into the time of per minute charges you will want to take into consideration that this depends a lot on the destination fax speed and the content (How much content is on the page compared to the blank space) being faxed. Unfortunately there are no definite ways in calculating costs. If you are sending just text, a page can go from 30 secs up to 1 min. If you are faxing multiple page documents this will go faster since you are not having to handshake or negotiate with the far side's fax machine for each page. This can result in 1.5 cents to 3 cents on a 1-2 page fax. Since we also increment the charge to every 6 seconds you are saving even more money since we do not charge you a full min for partial minute usages. You can usually get around 1-2 pages faxed per minute at 2.9 cents and $2 a month which is much less than most other Electronic Fax services being offered at this time.

== Current Limitations ==

*During the Beta test, each DID Number can only send 100 messages per day. This limit can be raised upon request and verification.

*The files sent per message cannot exceed 25mb

== Virtual Fax DID Number ==

Virtual Fax works specifically with Fax Numbers only acquired from the VoIP.ms Customer Portal or numbers ported in specifically as Fax enabled. There are '''local US and Canadian numbers''' available for order. You can order a Fax DID Number from your portal >> DID Numbers >> Order DID >> Fax Numbers. You can select the desired region and a random number from the chosen area code will be assigned to you.

[[File:FaxorderDID2.jpg|700px|]]

You can also port in a number you currently own with another provider. This process can be started from the Customer Portal at DID Numbers >> DID Portability >> Porting Fax Numbers

[[File:FaxPortability.png|700px]]

== Send a Fax ==

To start using the service you will need to head to your Customer Portal >> DID Numbers >> Virtual Fax. From the Home Page you can select ‘Send Fax’. There you will see:
*Fax Number or Contact Name: This is where you will put the destination number. You can start typing a name or a number from your Phone book and it will become available.
*From Name: Here you will put the name to send in the Fax header.
*From Number: Select the Fax DID number from which you will send your Fax.
*File: Choose a file to send as a Fax. The file must be in pdf, txt, jpg, gif, png or tif
-'''IMPORTANT: The maximum file size is 25MB'''
*Station ID: This will be the station ID you set for the header of the Fax message. It could be a specified post if your location has several stations, such as Reception, Main Office, Accounting PC, etc.
*Send Email: If selected, an email will be sent to the specified address to confirm the Fax has been sent successfully or to advise of a failed attempt.

[[File:SendFax.png|900px]]

== My Faxes ==
In this section of the Virtual Fax menu you will be able to view your Inbound and Outbound Faxes. You may select a date range and choose the folder you would like to view. Click 'Get My Faxes' to view your selection. You can view the Status of each Fax and select from several Actions. You can select to View the Fax directly, Download the Fax, Email the Fax to an address of your choosing or alter the location of the Fax by moving it to another folder.

Only the Status gets updated automatically. You will have to refresh the page to get the costs after a fax completes.

[[File:MyFaxes.png|900px]]

== My Folders ==

In the 'My Folders' section you can create folders by typing in the folder name of your choosing under 'New Folder' and clicking 'Create'.
You will have an overview of your Folders, see the date they were Created, the amount of Faxes in each Folder and be able to Edit the Folder or Delete it.
Any Faxes contained in a created folder will revert back to either the INBOX or SENT folder if the created folder is deleted.

[[File:MyFolders.png|900px]]

== My Fax Numbers ==
In the 'My Fax Numbers' section you will see your Fax DID numbers and Description, the Options that have been enabled for each number, the Email address if one has been configured along with the URL if configured in the URL Callback section. You can Edit the number from the 'Actions' section or choose to Delete it. When editing you will have the option to set an Email Address to receive a notification when a new Fax is received (you can also select to have the PDF file attached in the email) and set a URL Callback (you can also enable URL Callback Retry).

[[File:MyFaxNumbers.png|900px]]

== Email to Fax==

This feature allows you to send a Fax message using your email account.

How to send a Fax message using your email account:
*Use the email account you provided when enabling the Email to Fax service.
*Send the email to fax@voip.ms
*In the subject field type the Destination Fax Number (Example: 5148000000).
*Attach the document you wish to send to the email message. VoIP.ms supports the following formats: pdf, txt, jpg, gif, png, tif.
*Send the email.
**'''NOTE: Only the file attached will be faxed, the body of the e-mail won't be transmitted'''

'''Security Code and From Number:'''

If Security Code is enabled, you need to add a dot (.) and the Security Code after the Destination Fax Number (Example: 5148000000.Az09).

If you have more than one Fax number, you could change the From Number by adding a dot (.) and the From Number you'd like to use after the Destination Fax Number and the Security Code (Example: 5148000000.Az09.2268280000).

If you have not enabled the Security Code and want to change the From Number, you can add a dot (.) and the From Number after the Destination Fax Number (Example: 5148000000.2268280000).

[[File:EmailToFax.png|900px|]]

Extra Services Costs

2016-10-21T14:08:14Z

Rod:

Although most of the services at VoIP.ms are free there are some services that cost extra, beyond the Basic [[Service Cost]]s, either by a one time only, per call or monthly fee.

On this page you will see these services, what to do to enable or use them and how much they cost.

===Porting a Number / DID into VoIP.ms Service===
There is a one time fee of '''$8.50* USD''' per Canadian number in order to port it to our network (this may be lower if there is a promotion running), the process can take from 1-4 weeks for Canadian
Numbers and 2-4 weeks for US and Toll Free Numbers.

You can start the Porting Procedure from the [https://www.voip.ms/m/didporting.php DID Portability] page.

Please see the [[Porting a Number]] Page to get full details on Porting a Number.

(*Holiday Exchange Rate Special will be available for a Limited Time.)

===Porting Out a Number / DID from VoIP.ms Service===
At VoIP.ms sole discretion, you may incur a port away fee for any DID number(s) leaving our network as this is a pass-through charge from VoIP.ms carrier(s) or if you are caught using VoIP.ms as a DID Store to purchase DIDs and port them out immediately VoIP.ms reserves the right to charge you a porting out fee.

===CNAM Initiation and Updates===
VoIP.ms cannot provide this service to all numbers so please contact support@voip.ms to make sure they can for your number.

This process can take up to 15 business days and you should always test the CNAM with calling a landline.

CallerID Name (CNAM) is an string of 15 alphanumeric characters that are associated with a given CallerID number.

The sample below is a Caller ID, that includes both Caller ID name and Caller ID number, commonly abbreviated as CID and CNAM among other variations.

It is not possible to set any Caller ID Name from the VoIP.ms portal.

"John Smith" <9145551234>

'''In USA:'''
:There is a '''$10 fee''' to update or start the CNAM process, to get the name or update sent to the LIDB/CNAM Database.

:All CNAM queries originate from this database or an up to date copy of it. Most of VoIP.ms carriers support this process.

:Inbound CNAM [[Caller ID]] Name Lookup - '''0.8 cent''' (8/10 of a cent) per query.

:CNAM Lookup will be charged if the call matches a North American CallerID (NPANXXXXXX) and whether the call is answered or not. You are not charged for a CNAM Lookup if the number is in your Customer Portal's [[Phone book]]

'''In Canada:'''
:CallerID Name will pass on Outbound / Premium rate. VoIP.ms can also request the number to be submitted to the white pages (411), this will usually lead to the number getting a CNAM Entry to the LIDB/CNAM Database. However, the easiest way is to simply use the premium route and configure a name in the ATA Device, PBX or Softphone.

VoIP.ms does not make any profit from this, they are just passing on the charges as they are charged for this service.

On all requests, whether to start or to update it, VoIP.ms will need the following information in an email sent to support@voip.ms:

:DID Number(s):
:CNAM:
:(Up to 15 alphanumeric characters, a-z and 0-9 allowed)
:Name:
:Address: (No P.O. Boxes Please)
:City:
:State/Province:
:Zip/Postal Code:
:Contact Number:

===E911 / 911===
Use of the E911 Service costs a one time fee of '''$1.50''' on activation and a regulatory monthly fee of '''$1.50''' per DID number activated per month. VoIP.MS does not make any profit on this charge, it is simply what must be paid to provide this service.

Please note you can enable this service only for Canadian or US numbers (including USA or Canadian toll free numbers). You just have to click on the Apply button ([https://www.voip.ms/m/me911.php Click Here]) in order to start the process to enable the service for this number.

If at any time you need to change your address, you just need to select the "Modify" option, then click on Apply button. After this you will be able to change your e911 information and after the information has been approved, you will be confirmed by email and the e911 service will be updated.

Please read the [[E911|E911/911]] complete information page.

===411 / Directory Assistance===
4-1-1: '''$0.99''' per call.
:This must be activated by the customer in [https://www.voip.ms/m/settings.php Account Settings] then under Account Restrictions Tab.
:Please note that XXX-555-1212 has been disabled in our system and you will not be able to call it.

===Directory Listing / White Pages / 411 Listing===
VoIP.ms cannot provide this service to all numbers so please contact support@voip.ms to make sure they can for your number.

''For Canadians:''
:If your DID number can be listed in White Page Directory please also note that this process can take up to 15 business days.
:There is a '''$4.50 USD''' setup fee (one time only). You will be charged the setup fee again if you have to update your entry in the future.

''For USA Customers:''
:If your DID number can be listed in the White Page Directory please also note that this process can take up to 15 business days.
:There is a '''$5.00 USD''' setup fee (one time only) and an additional '''$1.49 USD''' charge per month. You will be charged the setup fee again if you have to update your entry in the future.

VoIP.ms does not make any profit from this, they are just passing on the charges as they are charged for this service.

On ALL Requests, whether to start or to update, VoIP.ms will need the following information in an email to support@voip.ms:

: Business Name: (if applicable)
: First Name:
: Last Name:
: Address:
: City:
: State:
: Zip Code:
: Email:
: Contact Number:

===SMS Messages===
Short Message Service ([[SMS]]) allows you to send and receive messages with your DID Number (Canada and US DID numbers only) for '''1 cent per message''' receiving and sending (once this feature is no longer in BETA). SMS is not currently available on [[toll-free number]] or [[Order a DID Number#Ordering iNums|iNum]] DIDs. Inbound SMS may be forwarded to an e-mail address or mobile phone. A [https://www.voip.ms/images/telephone.png handset icon] is displayed on the [https://www.voip.ms/m/managedid.php manage DID number] page to indicate each SMS-capable DID.

This feature is for regular customer usage. No automation, telemarketing, bulk sending or receiving will be allowed. For further information, please see [[SMS]].

Porting FAQ

2016-10-21T14:07:22Z

Rod:

Haga clic aquí para la versión en Español:[[Preguntas_frecuentes_LNP|Preguntas frecuentes LNP]]
__TOC__

'''Please also check out our other Wiki Guides in Regards to Porting:'''
*[http://wiki.voip.ms/article/Porting_a_Number Porting a Number]
*[http://wiki.voip.ms/article/Porting_a_US_Number Porting a US Number]
*[http://wiki.voip.ms/article/Porting_a_Canadian_Number Porting a Canadian Number]
*[http://wiki.voip.ms/article/Porting_a_Toll_Free_Number Porting a Toll Free Number]
*[http://wiki.voip.ms/article/Port_Rejection Port Rejections]

== How much does it cost to Port a Number to VoIP.ms? ==

There is a $8.50* USD one time fee for porting each number into VoIP.ms service. This is what VoIP.ms is charged and there is no profit made.
Sometimes the underlying Carrier will offer free ports and you will be advised, if this is so, when you port your number.

(*Holiday Exchange Rate Special will be available for a Limited Time.)

== How to port my Google Voice Number to VoIP.ms ==

You will first want to make sure that your number is portable to our service by checking here [https://www.voip.ms/portability.php Number Portability] then please follow this guide to [https://support.google.com/voice/answer/1316844?hl=en Unlock your Google Voice Number]

== How do I start the process to port my number(s)? ==
To start the porting request you can log in on your portal -> DID Numbers -> DID Portability and click on Start Procedure to start your porting order.

== What type of information will you need in order to port my number(s)? ==
The first thing you will need to provide to us is the Service Address of the number(s) you wish to port. This is the address where you will be using the number(s) and where you would like your E911 services to be listed. Even virtual numbers have a service address listed on the losing carrier’s side.

The service address will help us to fill up the Letter of Authorization (LOA). This form will be sent to our carrier to process your order.

In order to fill out this form, we will need the following information from you:
Company Name (if applicable), Service Address (as it is listed with your current carrier), Service Provider Name, Number to port, Billing Telephone Number (BTN):
This is the main billing number on your account that all of your numbers may be listed under. Please note that it is possible to have several different BTN's depending on how your account is set up.

Also, you will need to submit a hand-signed copy of your most recent billing invoice from your current phone provider.

All of the information needed for filling out the LOA can be supplied by your current phone provider by requesting a Customer Service Record (CSR).

== My current carrier does not provide any invoices, what else can I submit? ==

If your carrier does not provide billing invoices, you can submit a screen shot from the losing carrier’s online portal that shows the customer name, number to port and account information. The screenshot must also be printed, hand-signed and sent to us.

== What type of numbers can I port? ==
You can port any local, virtual and even a wireless number as long as we have coverage for that particular area. You can also port your Toll Free numbers and Fax numbers, although Fax is only available for US and Canada numbers.
Please note that Pager numbers are non-portable.

== My number is not portable to your network, even though I am close to a city which is portable to you, why is that? ==

The portability of a number depends on our carriers to cover that city (or rate center which is where the number actually belongs) and some towns/cities are not available no matter how close they are to covered cities. In some cases, you can purchase a number from a certain city but it is not available for porting. This again depends on our carriers’ coverage and it is out of our hands.

Some numbers are simply not portable. These include anything from pagers to numbers from a few small rural independent telephone companies to "shared use" toll-free numbers where the customer only has use of the number in one city or one area code. A few loopholes exist for a provider to claim they, not you, "own" a number

No ICA. The provider that you want to port to does not have an Interconnect Agreement in place with the telephone company that holds the number.

Our managers are always looking for new carriers to cover more rate centers and also our current carriers are always expanding their coverage; we suggest that you come back in the future and verify if your rate center is available for porting. You may do so by logging in on your portal -> DID Numbers -> DID Portability -> Check Availability

== How long does the port process take? ==

On average porting requests take from 2 to 4 business weeks to complete for US numbers and on average 1 business week for Canadian numbers; however, this can change depending always on the losing carrier’s cooperation level and also if there is any holiday in between. Please keep in mind any type of change or rejection can delay this process. This is why it is very important to verify the porting information prior to submitting it.

== What happens behind the scenes while my number is being ported? ==

Once your order has been submitted and received, you will be notified via email. After verifying your order, we will submit your port request with the information you provided to our carrier. You will be notified of its progress/status via email. If the request is rejected, we will provide you with instructions on how to resolve the issue.
Once we have received a FOC date, you will again be notified via email. Please keep in mind any changes made to a port request can push the FOC date further out.

== What happens if my port request gets rejected? ==

In the event that we receive a rejection from your carrier, we will provide you with instructions on how to resolve the issue. You can check the different and most common rejections by clicking on our Wiki Site [[Port Rejection]].

== How can I expedite my port request? ==

Unfortunately it is very hard for us to expedite a port request. The reason for this is because we have no control over the FOC date, each carrier has its own proper time to process an order. This date is given by the losing end and the FCC (for USA) and the CRTC (for Canada) mandate that we can’t take a number unless the current provider gives us permission.

However, if you provide us the correct information as it is listed on the CSR, we will do everything we can to expedite the process from our side.

== Will I experience any downtime while my numbers are porting? ==

On the day of your port, at some point, your number(s) will be triggered in our system. You should not experience any downtime as long as you have set up everything as you need it and properly routed the phone number(s) in your VoIP.ms portal. You will be able to view your number(s) in your VoIP.ms portal prior to the day of port in order to give you time to set everything up. Please contact Support via Live Chat or Ticket if you need help setting up everything before your number ports. However, it is important to also note that some downtime may occur during the porting day as the number(s) is(are) changing provides and routes.

== When/How will I know my number(s) has/have ported? ==

After your number(s) has/have completed porting, we will make test calls to make sure your number(s) is/are working properly. Once we have verified calls are completing on our network, you will be notified via email that the porting was a success. If there are any problems while test calls are being made, we will work with you until the issue has been resolved.

== My previous carrier is charging me for porting out and/or asks for a 30 day notice, how can I deal with this? Is it legal? ==

Unfortunately we have no control over the fees the losing carrier charges for porting out their numbers. Prior to submitting an order with us, please verify with them if there is any fee involved.

Regarding the 30 day notice, this is usually for Pay-as-you-go services. Since you were not charged for 30 days from the day you started your phone service with your carrier, you are responsible for payment of the 30 days following the cancellation of your account (which, in most cases, occurs due to number porting). This has always been stated in the terms of service of almost every carrier.
Some carrier contracts mention that the customer will “Pay for 30 days after the transfer is requested", and not after notice is given (by customer). The one who requests the transfer is the new carrier.
Unfortunately the CRTC (or FCC) do not regulate phone company policies.

== What is a RespOrg? ==

The port for Toll Free numbers is a bit different from porting local numbers. RespOrg is the communication carrier responsible for managing your Toll free number(s) and they use the SMS800 system to do so.

== What is a NASC procedure? ==

A NASC is a forced change of carriers done through the Number Administration Service Center. For a per-number NASC fee and the most recent invoice that is hand signed by the customer, the number is moved without the cooperation of the current carrier. The timeline for NASCing is 24 to 72 hours. Customers must also be aware that NASCing may only be a temporary fix; their previous carrier could NASC the number back at any time; although, this rarely happens as there is a NASC fee involved.

Sometimes it becomes necessary to utilize the NASC to port your Toll Free number to us. This can be caused when a previous RespOrg will not transfer a toll-free number to us. Often, a current RespOrg can reject the transfer for legitimate reasons (such as billing issues) or simply to stop the process.
However, if the business or customer that uses the number is in jeopardy of losing its 800 number or sales because of the number potentially being disconnected, NASCing is a way of forcing the process through.

''Please note that this process is only for USA and Canada Toll Free numbers''

== What are all of these acronyms: LOA, FOC, BTN, LSP,CSR, LSR, PON? ==

LOA - Letter of Authorization. This is the form we will need to fill out when porting your number(s). This form gives us permission to request, on your behalf, to port your number(s) to a different phone provider.

FOC - Firm Order Commitment. This is the date that is issued by the losing carrier as to when a phone provider will complete the port of a number(s). This date can only be issued by the losing carrier. The gaining phone provider has no control over what date is issued.

BTN - Billing Telephone Number. This is the main telephone number listed on your account. Depending on what type of account you have, you could have just one BTN or you could have several. When porting numbers to VoIP.ms you want to be sure to ask your current phone provider what the BTN(s) is/are for the number(s) you wish to port.

LSP - Losing Service Provider. This is the phone provider that your number(s) is/are leaving from.

CSR - Customer Service Record. Every carrier has a record that has your number(s), address, and Billing Telephone Number (BTN)(s). This is the record that is used to verify your information during the porting process. Please note that the information listed on the CSR may not be the same information listed on the billing portion of your account.

LSR - Local Service Request. This is the actual order request that the losing carrier will receive from the gaining provider. All of the information on this request must match the losing carrier’s (CSR) in order for them to issue a FOC date.

PON – Porting Order Number. This is the confirmation number that is created once the order has been sent to the losing carrier.

== I am being asked for my PIN, SSN and Password, IMEI why is that? ==

When porting a wireless / cell number, some extra information is needed. For US numbers you must provide the last 4 digits of the SSN (Social Security Number) and the PIN/Password on the current account. If your wireless number is prepaid you might have either one; however, if you are using a pay-as-you go service, you have to have both.
For Canadian Wireless numbers, we will need only your account number; unless the account number and the number porting are the same (meaning it’s a prepaid account); then we will need the IMEI, PIN and SIMM card (this one is not always available but the first 2 are).
Some carriers don't ask for this information when subscribing with them; however, our carrier does need this information to process your order.
The PIN stands for Personal Identification Number (PIN) which it's a 4-digit number combination known only by you. PIN numbers are often used for cell phones to lock the phone. This lock keeps your private information secure and makes your phone useless to anyone who steals it or finds it if you misplace it.

== What is the IMEI? ==

The IMEI is the International Mobile Equipment Identity is a number, usually unique, to identify mobile phones, as well as some satellite phones. It is usually found printed inside the battery compartment of the phone.
To retrieve the IMEI, enter *#06# and the IMEI will be displayed on your phone.

== How do I Port Out my number ? ==

If you have a DID number with VoIP.ms and wish to port it out to another provider, you need to always start this process on the new provider's side. They should advise you about what information to provide to start this process.
Remember to submit always the information as listed on your Contact Information section.

Canadian local and Toll Free port out numbers are authorized as long as the information matches with what is on your Contact Information section and you have positive balance. Please avoid deleting the number since in most cases the number is not ported yet and if you delete it, you may lose it; after the number is ported out from us, our LNP Staff will email you about it and will also delete the number from your account.

For US numbers, please make sure you delete the number after it's ported and tested with the new carrier and mark the 'Ported Out' box. This is to avoid any extra charges as, in some cases, we are not advised when a number is ported out.

Please open a ticket to our LNP Department if you get a rejection when porting out your number.

== When Porting Out my number the new carrier requests the PIN, what is my PIN? ==

Please note that numbers that are ported into us, or purchased from our portal, are not assigned any passwords/PINs. This could be a system rejection that the gaining carrier provides before sending the LSR to our underlying carrier.
If this number is being ported to a wireless provider, they may just require the customer to make up a PIN to port the number.

== My current carrier says the number has already been released or the order has not been received on their side yet. ==

The porting process is dealt between CLECs and (sometimes) resellers. However, some resellers are not aware of the port, not even when the number has left their side. We have no reason why we should not send an order as long as the customer provides the proper information. We will submit the order, most likely on the same day; however, since our carrier works as well with some other upstream carriers the order might take some time to be processed.
It is not odd that, if your order was rejected, your current carrier won’t see it since most carriers don’t save records of rejected orders.
Some carriers also say that the number has already been released/ported when in fact it has not. Our system is always monitoring when a number has been released without prior notice, if your number was in fact released, we will be notified about it and we will email you immediately after adding the number to your account to avoid any missed calls.

== I need my number to be ported on weekend or on a specific date, can I request this? ==

LNP Departments are closed on weekends; therefore no numbers can be ported during Saturdays or Sundays, nor holidays. If you need a specific date, you can leave a note when filling your porting request and we will ask our carrier if such date can be met. Please remember that this is not always possible; however, we will email you on any updates.

== My port is now processing or has a FOC date, can I now cancel my number with the losing carrier to avoid any extra charges? ==

It is important to note that you must wait until the number is fully ported to our system (we will email you regarding so) before you can cancel the account with the old carrier. If the order gets cancelled before the number is fully ported, you might get a rejection or even lose the number. Please be cautious.

== Why are purchased numbers cheaper than ported numbers? ==

This is because porting a number involves different carriers that must be paid for the transfer. Also, it is easier to purchase a number and configure it than to transfer an existing number from another carrier.

== My number has been disconnected by my current carrier and I am not receiving any calls, has it ported? ==

Please note that the number will always belong to the current provider until we email you regarding its completion. The number must be active at all times during the porting process. Disconnected numbers are not portable. If the losing carrier disconnected the number claiming that it has ported, you must verify if you have received a completion notice from us. If not, you must contact the current carrier and ask them to reactivate the number as soon as possible. Remember that the number is still on their end, since it has not ported.
Some carriers may disconnect the number just to stop the port and then activate it again on their side.

== My porting process has been completed and I have enabled SMS on my number, why can't I receive messages yet? ==

The SMS feature can take up to 48 hours (24 hours in average) after the porting process has been completed to become fully active. Please allow this much time before fully testing the SMS functionality on your recently ported in number.

== I am porting many numbers, can I get a special price for it? ==

Yes, prior to submit any porting request for many numbers, you should open up a ticket to our Sales Department ( Sales@voip.ms ) requesting the special price for the batch of numbers. Please note that the special price will only apply as long as all the numbers on the same batch have the same carrier, same customer information and same account information.
If the information is somehow different, you will need another quote.

== Now that the number has been ported to VoIP.ms, who cancels the old account? ==

Once we confirm the number has been successfully ported to us, check to see if your account is in fact closed and you are not continuing to be charged by the old carrier. If the account wasn't closed, please contact the losing carrier and ask them to cancel your old account if it's no longer useful to you anymore. This is to avoid any extra charges, as some carriers still show the ported number on their end, even when the number was ported out.
Please note that the cancellation can only be done by the customer since he/she is the one paying for the old carrier's services. We can't contact them nor cancel any services/lines with them.
Some carriers might say otherwise just to keep on billing you. Please demand the cancellation of the old account if it's no longer useful to you.
Note that neither your account with us nor the number will be affected by this cancellation, since they are now with us.

== I currently have a CNAM on my number, when porting will I keep it? ==

If you have a CNAM with the current carrier, when porting your number, it is most likely that the current CNAM is deleted when the transfer is completed, showing some random word such as the number itself, customer's first or last name or simply “unknown”. In some special cases the CNAM is kept as it was.

To avoid this, please verify if your number can have CNAM updated before starting the porting process. If so, after the number is ported, please test your number to verify your CNAM values and if they are not what you wish, contact our Support Staff ( Support@voip.ms ) to have the CNAM updated.

== At what point and how can I configure my porting number? ==

When the FOC is received, your number will be added to our system. From that point you can configure it and prepare it for the day the port completes. Your number will be reachable from anyone on our network at this time. Remember that if you need further assistance with your configuration, you can always open a ticket to our Support Staff who will gladly help you.

[[Category: Guides]]

Reseller Basic Guide

2016-10-19T20:37:06Z

Rod:

The Reseller feature is intended for users who want to initiate a business in the VoIP world. This useful tool can help you create and manage accounts for your own clients, you will be able to assign specific rates for them and also provide credentials to register a DID number. You will be able to handle the billing for your customers depending on your needs and how much you want to earn with your business, setting rates and adding credits for them. Additionally we will provide you with information about how to customize your Reseller portal to have a more personalized site.

The Reports section will allow you to check the Call detail records from the customers and review the Financial Reports related to their accounts. To activate your Reseller section, you need first to login to the Customer Portal then go to Reseller >> Reseller Main >> Activate

This guide will focus on the basic information to run the Reseller section properly. Here you can learn how to:

'''1. Manage Rates and Packages'''

'''2. Manage your clients'''

'''3. Use the client branding and access'''

'''4. Generate Call and Profit Reports'''

== Manage Rates and Packages. ==

This section will focus on how to create packages for the reseller feature, how to edit the rates depending on the destination and how to set a benefit for the calls your clients make. It is important to note that all the calls your customers make are charged to you as per our regulars fees, however on the reseller portal, they will see the charges according to the rates you set here.

=== Create a new package ===

From the Customer Portal go to Reseller >> Manage Rates and Packages >> Create a new Package.

[[File:CreatePackage2.png|right|650px]]

'''Package name''': Set a package name.

'''Outbound Markup''': Here you set the benefit you want to have on the calls that your clients make, you can set a fixed amount or a percentage, for this example if you set a 20%, your package will be set as follows ([cost rate] + $ fixed amount) * 20 %.

Example : for 1 cent per minute, the sell rate will (be 0.01 + 0.0000) + 20 % = 0.012

'''Pulse''': This is the charge increments that you will use for this package. For example, a pulse of 60 will make calls charged by the minute. You can read more about this in our [[Calls Cost]] article.

'''Canada Route & International Route''': These are the routes that the package will use on outgoing calls. For more information between the differences between Value and Premium, please read our article [[Value vs Premium]]

'''Voip.ms to Voip.ms Calls''': This option will make it so calls between VoIP.ms DID numbers are free. Applies to calls to USA and Canada VoIP.ms Numbers.

Under the '''Fees''' Tab you will be able to enter a value that you would charge as monthly base rate and a setup rate when connecting the package to a client.

The '''Free Minutes''' Tab will let you set a total of free calls, inbound & outbound for the package. If a vlue is entered in this field, the client will have free minutes (inbound & outbound calculated) up to the number of minutes entered here. Free minutes will be calculated from the invoice date for the package and will be calculated in total rounded minutes( for example, 3 minutes 11 seconds will be calculated as 4 minutes). The destinations added to the ''Free Zones'' at Reseller > Manage Rates & Packages > Edit Rates > Manage free zone will be able to spend these Free Minutes.

The '''Reseller System Configuration''' tab offers several advanced options for the package:

* '''User can edit Sub Account''': If enabled, the end user will be able to modify details of the sub account such as Password, Device Type, Authentication Type, CallerID Override, Music On Hold, Allowed Codecs, DTMF Mode and NAT.

* '''Type of Configuration''': This option determines the configuration of the services that will be shown in the Reseller Customer Portal. If the Default Configuration is set, the Services displayed will be the ones configured at Reseller > Client Branding and Access > Services tab. If the 'Package Configuration' is set, you will be able to set them in the Services section below.

* '''View Setting''': You can set if you want to display the information in the Client Interface on ''Simple View'' or ''Standard View'' here. Simple View is a feature that allows you to show your voipinterface.voip.ms client inferface in a simpler way to your clients. This hides some features like :
**Call Types (busy, answered, failed & noanswer) filters in the CDR
**This month and today`s amount spent on the Balance page
**Services specific information (Register type, Allowed codecs, etc) on the services page
**And some other information that would not be useful for "vonage style" clients

* '''Services''': In this section you can select which services will be available and which services will not be available for your customers, if you are using the "Package Configuration" option in the Type of Configuration section. (These Options are only available in the New Version of the reseller system)

=== Edit Rates ===

If you want to set different rates depending on the destination of the call, you can do this on the Edit Rates section from the package, all the custom rates you enter there will override the global markup. You can set a custom rate for all the calls to USA or Canada, by country or Calling code. You can also set your '''Free Zones''' here, which will be the destinations that will be included in the Free Minutes offered in the package.

[[File:EditRatesPackage.png|center|thumb|600px]]
<br clear="all" />

== Manage your clients ==

Now that you have a package, you need to add a client, this section will focus on how to create a client and associate it with a [http://wiki.voip.ms/article/Sub_Accounts sub account], as well as adding payments and charges for each client, assigning a DID number and also the proper way to delete a client account that is not longer needed.

[[File:ManageClientsAccounts.png|center|thumb|800px]]
<br clear="all" />

=== Adding a client ===

From the customer portal go to Reseller menu >> Manage Clients acct >> Create a new client account.

[[File:CreateClient.png|center|600px|thumb]]
<br clear="all" />

Fill the form with the client information and choose between soft and hard balance, when finished click on create account. A customer in HARD balance will be suspended if he is on zero or negative balance, while a customer in SOFT balance will be able to keep making use of the service even if their balance is zero or negative.

On the Manage section, you can add a new package or a DID number. To add numbers to your clients you need to order the number from your Customer Portal first. Don't forget to set a Per Minute rate and/or Monthly fee for your number, as incoming calls are charged independently from the outgoing calls.

Once the client account has been created, you will need to create a [http://wiki.voip.ms/article/Sub_Accounts sub account] for it as well.

Go to Sub accounts menu >> Create Sub account. During the Sub Account creation process, you will reach the Reseller configuration section, here you can choose:

[[File:ResellerConfiguration.png|right]]

*Reseller client

*Reseller client Package

*Next Billing date

*And if you decide to charge the set up fees now, make sure to check the appropriate box.

If you already have a [http://wiki.voip.ms/article/Sub_Accounts sub account] created, you can modify it at Sub Accounts >> Edit Sub Account >> Manage and add a package there.

=== Delete a client ===

You can delete a client on the Edit section from the Client's account, however note that before doing this you need to be sure to unconnect any package or DID number related to the client, you can do this from the Manage section on Client´s account.

=== Adding Payments manually ===

You can add payments manually from the customer portal >> Reseller menu >> Manage Clients acct >> Manage Client, then select the Payment button under the Balance panel.

=== Add Charges manually ===

You can add charges manually from the customer portal >> Reseller menu >> Manage Clients acct >> Manage Client, then select the Charge button under the Balance panel.

=== Add phone numbers to your clients (DIDs) ===

You can add phone numbers to your clients from the Customer portal >> Reseller menu >> Manage Clients acct >> Manage Client >> Package for this clients >> Add Phone number. Please note that you need to order a DID and have it on your account before you can assign it to a client. Don't forget to set a Per Minute rate and/or Monthly fee for your number, as incoming calls are charged independently from the outgoing calls.

[[File:AddPhone.png|center|thumb|600px]]
<br clear="all" />

== Use the client branding and access ==

It is always important to show a good brand image to your customer, for that reason we offer the option to customize our reseller interface to have the best presentation possible. There is also a useful feature on the site that can allow us to integrate your client's payments with a paypal account, so they can be added automatically as funds to the client.

=== Branded Interface ===

At the top of the screen you will see two available options, these are:

'''Version of the reseller system:''' This is the version of the Reseller System that will be displayed to your clients. The New Version of the Reseller System offers several customization options as well access to most of the VoIP.ms services, while the Old Version offers limited customization and access to basic services such as direct SIP routing and [[Call Forwarding]]

'''Configure Reseller Version:''' Determines which version of the Reseller System you will be configuring on this section. Enabling the New Version will offer you the option to modify the color of the panels, buttons and background of your Client Portal, as well as previewing it before making any changes.

=== Interface and Access ===

'''Clients Interface Access:''' This section will list the URL that you can give your clients to access their Account.

'''Auto Connect Form for Clients:''' This is an HTML form that you can use to automatically send your clients to their interface without needing authorization.

'''DID Access:''' You can set the pricing here for any DIDs that your clients purchase using the option at Services > Order DIDs on their Client Portal, if you have enabled this service for them.

=== Branding Management ===

In this section you will be able to change the Title that will appear in your Reseller Customer Portal page, as well as any of the colors of the interface, including the buttons* and the panels*, as well as selecting a Header, logo* and login background* for the interface.

<nowiki>*Options only available in the New Version of the reseller system.</nowiki>

=== Display Elements ===

Here you can choose whether or not you'd like to display the SIP and IAX password(s) in the Reseller Interface, as well as allowing your customers the option to reset their passwords in the Login Reseller Interface.

Lastly, you will be presented with an option to integrate an automated signup form into your website. If so, you can link your signup menu option to the URL presented there. You can also select which fields will be shown in order to sign up with a new account with you.

=== Services ===

View Setting: You can set if you want to display the information in the Client Interface on 'Simple View' here. Simple view is a feature that allows you to show your voipinterface.voip.ms client inferface in a simpler way to your clients. This hides some features like :

*Call Types (busy, answered, failed & noanswer) filters in the CDR
*This month and today`s amount spent on the Balance page
*Services specific information (Register type, Allowed codecs, etc) on the services page
*And some other information that would not be useful for "vonage style" clients

Services*: In this section you can select which services will be available and which services will not be available for your customers, if you are using the "Default Configuration" option in the Type of Configuration section of the Package settings.

<nowiki>*Options only available in the New Version of the reseller system</nowiki>

=== Text and Messages ===

In this tab you will be able to personalize the Welcome Text that will appear in the Reseller Customer Portal. You are also able to customize the Emails that will be sent when a Customer's Balance is running low, as well as when they request and complete a Password Reset.

== Payment ==

If you want your clients to deposit directly in your paypal account, you can activate the paypal integration. All payments will be sent to your paypal account and automatically credited in your client interface.

[[File:PaymentIntegration.png|center|thumb|600px]]
<br clear="all" />

Instructions :
#Enter your paypal email address in the "paypal email account" field
#Enter the amounts that you want to offer, without decimal and separated by a coma
#In your paypal account, go to "Profile" , and under the "Selling preference" section, activate the Instant payment notification(IPN). Enter the url : http://www.voipinterface.net/paynotify.php .
#We strongly recommend to make a live test with an amount of $1.

== Host Name ==

In order to personalize the interface with a hostname

#Configure a host with an A RECORD in your dns that points to 174.120.50.58
#Enter in the field bellow the fully qualified host + domain name (e.g: interface.mydomain.com)
#Click "submit"

There is a one time fee of $10.00 for adding your hostname to our system.

== Reports ==

Here you can see the Call Detail Records for the calls your customers have made and received, with the rates we have established on the packages created, showing a net total of the benefit earned as well.

Reports are divided in the following categories:

*'''Global CDR''': Display cdr report for all your reseller clients calls. You can see all the calls (incoming and outgoing) from all your clients.

*'''Per User CDR''': Display cdr report for a specific client. This filter help you to search calls only for one specific client.

*'''Financial report''': Month to month revenues ,deposits, costs related to the reseller clients. You can see the financial information by month.

*'''Sub accounts report''': List of all sub accounts connected to the reseller clients ,with related information. Here you will find a list of your clients, [http://wiki.voip.ms/article/Sub_Accounts sub account], package and DID number associated with each one.

*'''DID(s) Report''': List of all DID's connected to the reseller clients, with related information.

[[category:guides]]

New Reseller Guide

2016-10-19T19:33:34Z

Rod:

The Reseller feature is intended for users who want to initiate a business in the VoIP world. This useful tool can help you create and manage accounts for your own clients, you will be able to assign specific rates for them and also provide credentials to register a DID number. You will be able to handle the billing for your customers depending on your needs and how much you want to earn with your business, setting rates and adding credits for them. Additionally we will provide you with information about how to customize your Reseller portal to have a more personalized site.

The Reports section will allow you to check the Call detail records from the customers and review the Financial Reports related to their accounts. To activate your Reseller section, you need first to login to the Customer Portal then go to Reseller >> Reseller Main >> Activate

This guide will focus on the basic information to run the Reseller section properly. Here you can learn how to:

'''1. Manage Rates and Packages'''

'''2. Manage your clients'''

'''3. Use the client branding and access'''

'''4. Generate Call and Profit Reports'''

== Manage Rates and Packages. ==

This section will focus on how to create packages for the reseller feature, how to edit the rates depending on the destination and how to set a benefit for the calls your clients make. It is important to note that all the calls your customers make are charged to you as per our regulars fees, however on the reseller portal, they will see the charges according to the rates you set here.

=== Create a new package ===

From the Customer Portal go to Reseller >> Manage Rates and Packages >> Create a new Package.

[[File:CreatePackage2.png|right|650px]]

'''Package name''': Set a package name.

'''Outbound Markup''': Here you set the benefit you want to have on the calls that your clients make, you can set a fixed amount or a percentage, for this example if you set a 20%, your package will be set as follows ([cost rate] + $ fixed amount) * 20 %.

Example : for 1 cent per minute, the sell rate will (be 0.01 + 0.0000) + 20 % = 0.012

'''Pulse''': This is the charge increments that you will use for this package. For example, a pulse of 60 will make calls charged by the minute. You can read more about this in our [[Calls Cost]] article.

'''Canada Route & International Route''': These are the routes that the package will use on outgoing calls. For more information between the differences between Value and Premium, please read our article [[Value vs Premium]]

'''Voip.ms to Voip.ms Calls''': This option will make it so calls between VoIP.ms DID numbers are free. Applies to calls to USA and Canada VoIP.ms Numbers.

Under the '''Fees''' Tab you will be able to enter a value that you would charge as monthly base rate and a setup rate when connecting the package to a client.

The '''Free Minutes''' Tab will let you set a total of free calls, inbound & outbound for the package. If a vlue is entered in this field, the client will have free minutes (inbound & outbound calculated) up to the number of minutes entered here. Free minutes will be calculated from the invoice date for the package and will be calculated in total rounded minutes( for example, 3 minutes 11 seconds will be calculated as 4 minutes). The destinations added to the ''Free Zones'' at Reseller > Manage Rates & Packages > Edit Rates > Manage free zone will be able to spend these Free Minutes.

The '''Reseller System Configuration''' tab offers several advanced options for the package:

* '''User can edit Sub Account''': If enabled, the end user will be able to modify details of the sub account such as Password, Device Type, Authentication Type, CallerID Override, Music On Hold, Allowed Codecs, DTMF Mode and NAT.

* '''Type of Configuration''': This option determines the configuration of the services that will be shown in the Reseller Customer Portal. If the Default Configuration is set, the Services displayed will be the ones configured at Reseller > Client Branding and Access > Services tab. If the 'Package Configuration' is set, you will be able to set them in the Services section below.

* '''View Setting''': You can set if you want to display the information in the Client Interface on ''Simple View'' or ''Standard View'' here. Simple View is a feature that allows you to show your voipinterface.voip.ms client inferface in a simpler way to your clients. This hides some features like :
**Call Types (busy, answered, failed & noanswer) filters in the CDR
**This month and today`s amount spent on the Balance page
**Services specific information (Register type, Allowed codecs, etc) on the services page
**And some other information that would not be useful for "vonage style" clients

* '''Services''': In this section you can select which services will be available and which services will not be available for your customers, if you are using the "Package Configuration" option in the Type of Configuration section. (These Options are only available in the New Version of the reseller system)

=== Edit Rates ===

If you want to set different rates depending on the destination of the call, you can do this on the Edit Rates section from the package, all the custom rates you enter there will override the global markup. You can set a custom rate for all the calls to USA or Canada, by country or Calling code. You can also set your '''Free Zones''' here, which will be the destinations that will be included in the Free Minutes offered in the package.

[[File:EditRatesPackage.png|center|thumb|600px]]
<br clear="all" />

== Manage your clients ==

Now that you have a package, you need to add a client, this section will focus on how to create a client and associate it with a [http://wiki.voip.ms/article/Sub_Accounts sub account], as well as adding payments and charges for each client, assigning a DID number and also the proper way to delete a client account that is not longer needed.

[[File:ManageClientsAccounts.png|center|thumb|800px]]
<br clear="all" />

=== Adding a client ===

From the customer portal go to Reseller menu >> Manage Clients acct >> Create a new client account.

[[File:CreateClient.png|center|600px|thumb]]
<br clear="all" />

Fill the form with the client information and choose between soft and hard balance, when finished click on create account. A customer in HARD balance will be suspended if he is on zero or negative balance, while a customer in SOFT balance will be able to keep making use of the service even if their balance is zero or negative.

On the Manage section, you can add a new package or a DID number. To add numbers to your clients you need to order the number from your Customer Portal first. Don't forget to set a Per Minute rate and/or Monthly fee for your number, as incoming calls are charged independently from the outgoing calls.

Once the client account has been created, you will need to create a [http://wiki.voip.ms/article/Sub_Accounts sub account] for it as well.

Go to Sub accounts menu >> Create Sub account. During the Sub Account creation process, you will reach the Reseller configuration section, here you can choose:

[[File:ResellerConfiguration.png|right]]

*Reseller client

*Reseller client Package

*Next Billing date

*And if you decide to charge the set up fees now, make sure to check the appropriate box.

If you already have a [http://wiki.voip.ms/article/Sub_Accounts sub account] created, you can modify it at Sub Accounts >> Edit Sub Account >> Manage and add a package there.

=== Delete a client ===

You can delete a client on the Edit section from the Client's account, however note that before doing this you need to be sure to unconnect any package or DID number related to the client, you can do this from the Manage section on Client´s account.

=== Adding Payments manually ===

You can add payments manually from the customer portal >> Reseller menu >> Manage Clients acct >> Manage Client, then select the Payment button under the Balance panel.

=== Add Charges manually ===

You can add charges manually from the customer portal >> Reseller menu >> Manage Clients acct >> Manage Client, then select the Charge button under the Balance panel.

=== Add phone numbers to your clients (DIDs) ===

You can add phone numbers to your clients from the Customer portal >> Reseller menu >> Manage Clients acct >> Manage Client >> Package for this clients >> Add Phone number. Please note that you need to order a DID and have it on your account before you can assign it to a client. Don't forget to set a Per Minute rate and/or Monthly fee for your number, as incoming calls are charged independently from the outgoing calls.

[[File:AddPhone.png|center|thumb|600px]]
<br clear="all" />

== Use the client branding and access ==

It is always important to show a good brand image to your customer, for that reason we offer the option to customize our reseller interface to have the best presentation possible. There is also a useful feature on the site that can allow us to integrate your client's payments with a paypal account, so they can be added automatically as funds to the client.

=== Branded Interface ===

At the top of the screen you will see two available options, these are:

'''Version of the reseller system:''' This is the version of the Reseller System that will be displayed to your clients. The New Version of the Reseller System offers several customization options as well access to most of the VoIP.ms services, while the Old Version offers limited customization and access to basic services such as direct SIP routing and [[Call Forwarding]]

'''Configure Reseller Version:''' Determines which version of the Reseller System you will be configuring on this section. Enabling the New Version will offer you the option to modify the color of the panels, buttons and background of your Client Portal, as well as previewing it before making any changes.

=== Interface and Access ===

'''Clients Interface Access:''' This section will list the URL that you can give your clients to access their Account.

'''Auto Connect Form for Clients:''' This is an HTML form that you can use to automatically send your clients to their interface without needing authorization.

'''DID Access:''' You can set the pricing here for any DIDs that your clients purchase using the option at Services > Order DIDs on their Client Portal, if you have enabled this service for them.

=== Branding Management ===

In this section you will be able to change the Title that will appear in your Reseller Customer Portal page, as well as any of the colors of the interface, including the buttons* and the panels*, as well as selecting a Header, logo* and login background* for the interface.

<nowiki>*Options only available in the New Version of the reseller system.</nowiki>

=== Display Elements ===

Here you can choose whether or not you'd like to display the SIP and IAX password(s) in the Reseller Interface, as well as allowing your customers the option to reset their passwords in the Login Reseller Interface.

Lastly, you will be presented with an option to integrate an automated signup form into your website. If so, you can link your signup menu option to the URL presented there. You can also select which fields will be shown in order to sign up with a new account with you.

=== Services ===

View Setting: You can set if you want to display the information in the Client Interface on 'Simple View' here. Simple view is a feature that allows you to show your voipinterface.voip.ms client inferface in a simpler way to your clients. This hides some features like :

*Call Types (busy, answered, failed & noanswer) filters in the CDR
*This month and today`s amount spent on the Balance page
*Services specific information (Register type, Allowed codecs, etc) on the services page
*And some other information that would not be useful for "vonage style" clients

Services*: In this section you can select which services will be available and which services will not be available for your customers, if you are using the "Default Configuration" option in the Type of Configuration section of the Package settings.

<nowiki>*Options only available in the New Version of the reseller system</nowiki>

=== Text and Messages ===

In this tab you will be able to personalize the Welcome Text that will appear in the Reseller Customer Portal. You are also able to customize the Emails that will be sent when a Customer's Balance is running low, as well as when they request and complete a Password Reset.

== Payment ==

If you want your clients to deposit directly in your paypal account, you can activate the paypal integration. All payments will be sent to your paypal account and automatically credited in your client interface.

[[File:PaymentIntegration.png|center|thumb|600px]]
<br clear="all" />

Instructions :
#Enter your paypal email address in the "paypal email account" field
#Enter the amounts that you want to offer, without decimal and separated by a coma
#In your paypal account, go to "Profile" , and under the "Selling preference" section, activate the Instant payment notification(IPN). Enter the url : http://www.voipinterface.net/paynotify.php .
#We strongly recommend to make a live test with an amount of $1.

== Host Name ==

In order to personalize the interface with a hostname

#Configure a host with an A RECORD in your dns that points to 174.120.50.58
#Enter in the field bellow the fully qualified host + domain name (e.g: interface.mydomain.com)
#Click "submit"

There is a one time fee of $10.00 for adding your hostname to our system.

== Reports ==

Here you can see the Call Detail Records for the calls your customers have made and received, with the rates we have established on the packages created, showing a net total of the benefit earned as well.

Reports are divided in the following categories:

*'''Global CDR''': Display cdr report for all your reseller clients calls. You can see all the calls (incoming and outgoing) from all your clients.

*'''Per User CDR''': Display cdr report for a specific client. This filter help you to search calls only for one specific client.

*'''Financial report''': Month to month revenues ,deposits, costs related to the reseller clients. You can see the financial information by month.

*'''Sub accounts report''': List of all sub accounts connected to the reseller clients ,with related information. Here you will find a list of your clients, [http://wiki.voip.ms/article/Sub_Accounts sub account], package and DID number associated with each one.

*'''DID(s) Report''': List of all DID's connected to the reseller clients, with related information.

[[category:guides]]

New Reseller Guide

2016-10-19T19:29:39Z

Rod:

The Reseller feature is intended for users who want to initiate a business in the VoIP world. This useful tool can help you create and manage accounts for your own clients, you will be able to assign specific rates for them and also provide credentials to register a DID number. You will be able to handle the billing for your customers depending on your needs and how much you want to earn with your business, setting rates and adding credits for them. Additionally we will provide you with information about how to customize your Reseller portal to have a more personalized site.

The Reports section will allow you to check the Call detail records from the customers and review the Financial Reports related to their accounts. To activate your Reseller section, you need first to login to the Customer Portal then go to Reseller >> Reseller Main >> Activate

This guide will focus on the basic information to run the Reseller section properly. Here you can learn how to:

'''1. Manage Rates and Packages'''

'''2. Manage your clients'''

'''3. Use the client branding and access'''

'''4. Generate Call and Profit Reports'''

== Manage Rates and Packages. ==

This section will focus on how to create packages for the reseller feature, how to edit the rates depending on the destination and how to set a benefit for the calls your clients make. It is important to note that all the calls your customers make are charged to you as per our regulars fees, however on the reseller portal, they will see the charges according to the rates you set here.

=== Create a new package ===

From the Customer Portal go to Reseller >> Manage Rates and Packages >> Create a new Package.

[[File:CreatePackage2.png|right|650px]]

'''Package name''': Set a package name.

'''Outbound Markup''': Here you set the benefit you want to have on the calls that your clients make, you can set a fixed amount or a percentage, for this example if you set a 20%, your package will be set as follows ([cost rate] + $ fixed amount) * 20 %.

Example : for 1 cent per minute, the sell rate will (be 0.01 + 0.0000) + 20 % = 0.012

'''Pulse''': This is the charge increments that you will use for this package. For example, a pulse of 60 will make calls charged by the minute. You can read more about this in our [[Calls Cost]] article.

'''Canada Route & International Route''': These are the routes that the package will use on outgoing calls. For more information between the differences between Value and Premium, please read our article [[Value vs Premium]]

'''Voip.ms to Voip.ms Calls''': This option will make it so calls between VoIP.ms DID numbers are free. Applies to calls to USA and Canada VoIP.ms Numbers.

Under the '''Fees''' Tab you will be able to enter a value that you would charge as monthly base rate and a setup rate when connecting the package to a client.

The '''Free Minutes''' Tab will let you set a total of free calls, inbound & outbound for the package. If a vlue is entered in this field, the client will have free minutes (inbound & outbound calculated) up to the number of minutes entered here. Free minutes will be calculated from the invoice date for the package and will be calculated in total rounded minutes( for example, 3 minutes 11 seconds will be calculated as 4 minutes). The destinations added to the ''Free Zones'' at Reseller > Manage Rates & Packages > Edit Rates > Manage free zone will be able to spend these Free Minutes.

The '''Reseller System Configuration''' tab offers several advanced options for the package:

* '''User can edit Sub Account''': If enabled, the end user will be able to modify details of the sub account such as Password, Device Type, Authentication Type, CallerID Override, Music On Hold, Allowed Codecs, DTMF Mode and NAT.

* '''Type of Configuration''': This option determines the configuration of the services that will be shown in the Reseller Customer Portal. If the Default Configuration is set, the Services displayed will be the ones configured at Reseller > Client Branding and Access > Services tab. If the 'Package Configuration' is set, you will be able to set them in the Services section below.

* '''View Setting''': You can set if you want to display the information in the Client Interface on ''Simple View'' or ''Standard View'' here. Simple View is a feature that allows you to show your voipinterface.voip.ms client inferface in a simpler way to your clients. This hides some features like :
**Call Types (busy, answered, failed & noanswer) filters in the CDR
**This month and today`s amount spent on the Balance page
**Services specific information (Register type, Allowed codecs, etc) on the services page
**And some other information that would not be useful for "vonage style" clients

* '''Services''': In this section you can select which services will be available and which services will not be available for your customers, if you are using the "Package Configuration" option in the Type of Configuration section. (These Options are only available in the New Version of the reseller system)

=== Edit Rates ===

If you want to set different rates depending on the destination of the call, you can do this on the Edit Rates section from the package, all the custom rates you enter there will override the global markup. You can set a custom rate for all the calls to USA or Canada, by country or Calling code. You can also set your '''Free Zones''' here, which will be the destinations that will be included in the Free Minutes offered in the package.

[[File:EditRatesPackage.png|center|thumb|600px]]
<br clear="all" />

== Manage your clients ==

Now that you have a package, you need to add a client, this section will focus on how to create a client and associate it with a [http://wiki.voip.ms/article/Sub_Accounts sub account], as well as adding payments and charges for each client, assigning a DID number and also the proper way to delete a client account that is not longer needed.

[[File:ManageClientsAccounts.png|center|thumb|800px]]
<br clear="all" />

=== Adding a client ===

From the customer portal go to Reseller menu >> Manage Clients acct >> Create a new client account.

[[File:CreateClient.png|center|600px]]

Fill the form with the client information and choose between soft and hard balance, when finished click on create account. A customer in HARD balance will be suspended if he is on zero or negative balance, while a customer in SOFT balance will be able to keep making use of the service even if their balance is zero or negative.

On the Manage section, you can add a new package or a DID number. To add numbers to your clients you need to order the number from your Customer Portal first. Don't forget to set a Per Minute rate and/or Monthly fee for your number, as incoming calls are charged independently from the outgoing calls.

Once the client account has been created, you will need to create a [http://wiki.voip.ms/article/Sub_Accounts sub account] for it as well.

Go to Sub accounts menu >> Create Sub account. During the Sub Account creation process, you will reach the Reseller configuration section, here you can choose:

[[File:ResellerConfiguration.png|right]]

*Reseller client

*Reseller client Package

*Next Billing date

*And if you decide to charge the set up fees now, make sure to check the appropriate box.

If you already have a [http://wiki.voip.ms/article/Sub_Accounts sub account] created, you can modify it at Sub Accounts >> Edit Sub Account >> Manage and add a package there.

=== Delete a client ===

You can delete a client on the Edit section from the Client's account, however note that before doing this you need to be sure to unconnect any package or DID number related to the client, you can do this from the Manage section on Client´s account.

=== Adding Payments manually ===

You can add payments manually from the customer portal >> Reseller menu >> Manage Clients acct >> Manage Client, then select the Payment button under the Balance panel.

=== Add Charges manually ===

You can add charges manually from the customer portal >> Reseller menu >> Manage Clients acct >> Manage Client, then select the Charge button under the Balance panel.

=== Add phone numbers to your clients (DIDs) ===

You can add phone numbers to your clients from the Customer portal >> Reseller menu >> Manage Clients acct >> Manage Client >> Package for this clients >> Add Phone number. Please note that you need to order a DID and have it on your account before you can assign it to a client. Don't forget to set a Per Minute rate and/or Monthly fee for your number, as incoming calls are charged independently from the outgoing calls.

[[File:AddPhone.png|center]]

== Use the client branding and access ==

It is always important to show a good brand image to your customer, for that reason we offer the option to customize our reseller interface to have the best presentation possible. There is also a useful feature on the site that can allow us to integrate your client's payments with a paypal account, so they can be added automatically as funds to the client.

=== Branded Interface ===

At the top of the screen you will see two available options, these are:

'''Version of the reseller system:''' This is the version of the Reseller System that will be displayed to your clients. The New Version of the Reseller System offers several customization options as well access to most of the VoIP.ms services, while the Old Version offers limited customization and access to basic services such as direct SIP routing and [[Call Forwarding]]

'''Configure Reseller Version:''' Determines which version of the Reseller System you will be configuring on this section. Enabling the New Version will offer you the option to modify the color of the panels, buttons and background of your Client Portal, as well as previewing it before making any changes.

=== Interface and Access ===

'''Clients Interface Access:''' This section will list the URL that you can give your clients to access their Account.

'''Auto Connect Form for Clients:''' This is an HTML form that you can use to automatically send your clients to their interface without needing authorization.

'''DID Access:''' You can set the pricing here for any DIDs that your clients purchase using the option at Services > Order DIDs on their Client Portal, if you have enabled this service for them.

=== Branding Management ===

In this section you will be able to change the Title that will appear in your Reseller Customer Portal page, as well as any of the colors of the interface, including the buttons* and the panels*, as well as selecting a Header, logo* and login background* for the interface.

<nowiki>*Options only available in the New Version of the reseller system.</nowiki>

=== Display Elements ===

Here you can choose whether or not you'd like to display the SIP and IAX password(s) in the Reseller Interface, as well as allowing your customers the option to reset their passwords in the Login Reseller Interface.

Lastly, you will be presented with an option to integrate an automated signup form into your website. If so, you can link your signup menu option to the URL presented there. You can also select which fields will be shown in order to sign up with a new account with you.

=== Services ===

View Setting: You can set if you want to display the information in the Client Interface on 'Simple View' here. Simple view is a feature that allows you to show your voipinterface.voip.ms client inferface in a simpler way to your clients. This hides some features like :

*Call Types (busy, answered, failed & noanswer) filters in the CDR
*This month and today`s amount spent on the Balance page
*Services specific information (Register type, Allowed codecs, etc) on the services page
*And some other information that would not be useful for "vonage style" clients

Services*: In this section you can select which services will be available and which services will not be available for your customers, if you are using the "Default Configuration" option in the Type of Configuration section of the Package settings.

<nowiki>*Options only available in the New Version of the reseller system</nowiki>

=== Text and Messages ===

In this tab you will be able to personalize the Welcome Text that will appear in the Reseller Customer Portal. You are also able to customize the Emails that will be sent when a Customer's Balance is running low, as well as when they request and complete a Password Reset.

== Payment ==

If you want your clients to deposit directly in your paypal account, you can activate the paypal integration. All payments will be sent to your paypal account and automatically credited in your client interface.

[[File:PaymentIntegration.png|center|600px]]

Instructions :
#Enter your paypal email address in the "paypal email account" field
#Enter the amounts that you want to offer, without decimal and separated by a coma
#In your paypal account, go to "Profile" , and under the "Selling preference" section, activate the Instant payment notification(IPN). Enter the url : http://www.voipinterface.net/paynotify.php .
#We strongly recommend to make a live test with an amount of $1.

== Host Name ==

In order to personalize the interface with a hostname

#Configure a host with an A RECORD in your dns that points to 174.120.50.58
#Enter in the field bellow the fully qualified host + domain name (e.g: interface.mydomain.com)
#Click "submit"

There is a one time fee of $10.00 for adding your hostname to our system.

== Reports ==

Here you can see the Call Detail Records for the calls your customers have made and received, with the rates we have established on the packages created, showing a net total of the benefit earned as well.

Reports are divided in the following categories:

*'''Global CDR''': Display cdr report for all your reseller clients calls. You can see all the calls (incoming and outgoing) from all your clients.

*'''Per User CDR''': Display cdr report for a specific client. This filter help you to search calls only for one specific client.

*'''Financial report''': Month to month revenues ,deposits, costs related to the reseller clients. You can see the financial information by month.

*'''Sub accounts report''': List of all sub accounts connected to the reseller clients ,with related information. Here you will find a list of your clients, [http://wiki.voip.ms/article/Sub_Accounts sub account], package and DID number associated with each one.

*'''DID(s) Report''': List of all DID's connected to the reseller clients, with related information.

[[category:guides]]

New Reseller Guide

2016-10-19T19:16:16Z

Rod: /* Interface and Access */

The Reseller feature is intended for users who want to initiate a business in the VoIP world. This useful tool can help you create and manage accounts for your own clients, you will be able to assign specific rates for them and also provide credentials to register a DID number. You will be able to handle the billing for your customers depending on your needs and how much you want to earn with your business, setting rates and adding credits for them. Additionally we will provide you with information about how to customize your Reseller portal to have a more personalized site.

The Reports section will allow you to check the Call detail records from the customers and review the Financial Reports related to their accounts. To activate your Reseller section, you need first to login to the Customer Portal then go to Reseller >> Reseller Main >> Activate

This guide will focus on the basic information to run the Reseller section properly. Here you can learn how to:

'''1. Manage Rates and Packages'''

'''2. Manage your clients'''

'''3. Use the client branding and access'''

'''4. Generate Call and Profit Reports'''

== Manage Rates and Packages. ==

This section will focus on how to create packages for the reseller feature, how to edit the rates depending on the destination and how to set a benefit for the calls your clients make. It is important to note that all the calls your customers make are charged to you as per our regulars fees, however on the reseller portal, they will see the charges according to the rates you set here.

=== Create a new package ===

From the Customer Portal go to Reseller >> Manage Rates and Packages >> Create a new Package.

[[File:CreatePackage2.png|right|650px]]

'''Package name''': Set a package name.

'''Outbound Markup''': Here you set the benefit you want to have on the calls that your clients make, you can set a fixed amount or a percentage, for this example if you set a 20%, your package will be set as follows ([cost rate] + $ fixed amount) * 20 %.

Example : for 1 cent per minute, the sell rate will (be 0.01 + 0.0000) + 20 % = 0.012

'''Pulse''': This is the charge increments that you will use for this package. For example, a pulse of 60 will make calls charged by the minute. You can read more about this in our [[Calls Cost]] article.

'''Canada Route & International Route''': These are the routes that the package will use on outgoing calls. For more information between the differences between Value and Premium, please read our article [[Value vs Premium]]

'''Voip.ms to Voip.ms Calls''': This option will make it so calls between VoIP.ms DID numbers are free. Applies to calls to USA and Canada VoIP.ms Numbers.

Under the '''Fees''' Tab you will be able to enter a value that you would charge as monthly base rate and a setup rate when connecting the package to a client.

The '''Free Minutes''' Tab will let you set a total of free calls, inbound & outbound for the package. If a vlue is entered in this field, the client will have free minutes (inbound & outbound calculated) up to the number of minutes entered here. Free minutes will be calculated from the invoice date for the package and will be calculated in total rounded minutes( for example, 3 minutes 11 seconds will be calculated as 4 minutes). The destinations added to the ''Free Zones'' at Reseller > Manage Rates & Packages > Edit Rates > Manage free zone will be able to spend these Free Minutes.

The '''Reseller System Configuration''' tab offers several advanced options for the package:

* '''User can edit Sub Account''': If enabled, the end user will be able to modify details of the sub account such as Password, Device Type, Authentication Type, CallerID Override, Music On Hold, Allowed Codecs, DTMF Mode and NAT.

* '''Type of Configuration''': This option determines the configuration of the services that will be shown in the Reseller Customer Portal. If the Default Configuration is set, the Services displayed will be the ones configured at Reseller > Client Branding and Access > Services tab. If the 'Package Configuration' is set, you will be able to set them in the Services section below.

* '''View Setting''': You can set if you want to display the information in the Client Interface on ''Simple View'' or ''Standard View'' here. Simple View is a feature that allows you to show your voipinterface.voip.ms client inferface in a simpler way to your clients. This hides some features like :
**Call Types (busy, answered, failed & noanswer) filters in the CDR
**This month and today`s amount spent on the Balance page
**Services specific information (Register type, Allowed codecs, etc) on the services page
**And some other information that would not be useful for "vonage style" clients

* '''Services''': In this section you can select which services will be available and which services will not be available for your customers, if you are using the "Package Configuration" option in the Type of Configuration section. (These Options are only available in the New Version of the reseller system)

=== Edit Rates ===

If you want to set different rates depending on the destination of the call, you can do this on the Edit Rates section from the package, all the custom rates you enter there will override the global markup. You can set a custom rate for all the calls to USA or Canada, by country or Calling code. You can also set your '''Free Zones''' here, which will be the destinations that will be included in the Free Minutes offered in the package.

[[File:EditRatesPackage.png|center|600px]]

== Manage your clients ==

Now that you have a package, you need to add a client, this section will focus on how to create a client and associate it with a [http://wiki.voip.ms/article/Sub_Accounts sub account], as well as adding payments and charges for each client, assigning a DID number and also the proper way to delete a client account that is not longer needed.

[[File:ManageClientsAccounts.png|center|800px]]

=== Adding a client ===

From the customer portal go to Reseller menu >> Manage Clients acct >> Create a new client account.

[[File:CreateClient.png|center|600px]]

Fill the form with the client information and choose between soft and hard balance, when finished click on create account. A customer in HARD balance will be suspended if he is on zero or negative balance, while a customer in SOFT balance will be able to keep making use of the service even if their balance is zero or negative.

On the Manage section, you can add a new package or a DID number. To add numbers to your clients you need to order the number from your Customer Portal first. Don't forget to set a Per Minute rate and/or Monthly fee for your number, as incoming calls are charged independently from the outgoing calls.

Once the client account has been created, you will need to create a [http://wiki.voip.ms/article/Sub_Accounts sub account] for it as well.

Go to Sub accounts menu >> Create Sub account. During the Sub Account creation process, you will reach the Reseller configuration section, here you can choose:

[[File:ResellerConfiguration.png|right]]

*Reseller client

*Reseller client Package

*Next Billing date

*And if you decide to charge the set up fees now, make sure to check the appropriate box.

If you already have a [http://wiki.voip.ms/article/Sub_Accounts sub account] created, you can modify it at Sub Accounts >> Edit Sub Account >> Manage and add a package there.

=== Delete a client ===

You can delete a client on the Edit section from the Client's account, however note that before doing this you need to be sure to unconnect any package or DID number related to the client, you can do this from the Manage section on Client´s account.

=== Adding Payments manually ===

You can add payments manually from the customer portal >> Reseller menu >> Manage Clients acct >> Manage Client, then select the Payment button under the Balance panel.

=== Add Charges manually ===

You can add charges manually from the customer portal >> Reseller menu >> Manage Clients acct >> Manage Client, then select the Charge button under the Balance panel.

=== Add phone numbers to your clients (DIDs) ===

You can add phone numbers to your clients from the Customer portal >> Reseller menu >> Manage Clients acct >> Manage Client >> Package for this clients >> Add Phone number. Please note that you need to order a DID and have it on your account before you can assign it to a client. Don't forget to set a Per Minute rate and/or Monthly fee for your number, as incoming calls are charged independently from the outgoing calls.

[[File:AddPhone.png|center]]

== Use the client branding and access ==

It is always important to show a good brand image to your customer, for that reason we offer the option to customize our reseller interface to have the best presentation possible. There is also a useful feature on the site that can allow us to integrate your client's payments with a paypal account, so they can be added automatically as funds to the client.

=== Branded Interface ===

At the top of the screen you will see two available options, these are:

'''Version of the reseller system:''' This is the version of the Reseller System that will be displayed to your clients. The New Version of the Reseller System offers several customization options as well access to most of the VoIP.ms services, while the Old Version offers limited customization and access to basic services such as direct SIP routing and [[Call Forwarding]]

'''Configure Reseller Version:''' Determines which version of the Reseller System you will be configuring on this section. Enabling the New Version will offer you the option to modify the color of the panels, buttons and background of your Client Portal, as well as previewing it before making any changes.

=== Interface and Access ===

'''Clients Interface Access:''' This section will list the URL that you can give your clients to access their Account.

'''Auto Connect Form for Clients:''' This is an HTML form that you can use to automatically send your clients to their interface without needing authorization.

'''DID Access:''' You can set the pricing here for any DIDs that your clients purchase using the option at Services > Order DIDs on their Client Portal, if you have enabled this service for them.

=== Branding Management ===

In this section you will be able to change the Title that will appear in your Reseller Customer Portal page, as well as any of the colors of the interface, including the buttons* and the panels*, as well as selecting a Header, logo* and login background* for the interface.

<nowiki>*Options only available in the New Version of the reseller system.</nowiki>

=== Display Elements ===

Here you can choose whether or not you'd like to display the SIP and IAX password(s) in the Reseller Interface, as well as allowing your customers the option to reset their passwords in the Login Reseller Interface.

Lastly, you will be presented with an option to integrate an automated signup form into your website. If so, you can link your signup menu option to the URL presented there. You can also select which fields will be shown in order to sign up with a new account with you.

=== Services ===

View Setting: You can set if you want to display the information in the Client Interface on 'Simple View' here. Simple view is a feature that allows you to show your voipinterface.voip.ms client inferface in a simpler way to your clients. This hides some features like :

*Call Types (busy, answered, failed & noanswer) filters in the CDR
*This month and today`s amount spent on the Balance page
*Services specific information (Register type, Allowed codecs, etc) on the services page
*And some other information that would not be useful for "vonage style" clients

Services*: In this section you can select which services will be available and which services will not be available for your customers, if you are using the "Default Configuration" option in the Type of Configuration section of the Package settings.

<nowiki>*Options only available in the New Version of the reseller system</nowiki>

=== Text and Messages ===

In this tab you will be able to personalize the Welcome Text that will appear in the Reseller Customer Portal. You are also able to customize the Emails that will be sent when a Customer's Balance is running low, as well as when they request and complete a Password Reset.

== Payment ==

If you want your clients to deposit directly in your paypal account, you can activate the paypal integration. All payments will be sent to your paypal account and automatically credited in your client interface.

[[File:PaymentIntegration.png|center|600px]]

Instructions :
#Enter your paypal email address in the "paypal email account" field
#Enter the amounts that you want to offer, without decimal and separated by a coma
#In your paypal account, go to "Profile" , and under the "Selling preference" section, activate the Instant payment notification(IPN). Enter the url : http://www.voipinterface.net/paynotify.php .
#We strongly recommend to make a live test with an amount of $1.

== Host Name ==

In order to personalize the interface with a hostname

#Configure a host with an A RECORD in your dns that points to 174.120.50.58
#Enter in the field bellow the fully qualified host + domain name (e.g: interface.mydomain.com)
#Click "submit"

There is a one time fee of $10.00 for adding your hostname to our system.

== Reports ==

Here you can see the Call Detail Records for the calls your customers have made and received, with the rates we have established on the packages created, showing a net total of the benefit earned as well.

Reports are divided in the following categories:

*'''Global CDR''': Display cdr report for all your reseller clients calls. You can see all the calls (incoming and outgoing) from all your clients.

*'''Per User CDR''': Display cdr report for a specific client. This filter help you to search calls only for one specific client.

*'''Financial report''': Month to month revenues ,deposits, costs related to the reseller clients. You can see the financial information by month.

*'''Sub accounts report''': List of all sub accounts connected to the reseller clients ,with related information. Here you will find a list of your clients, [http://wiki.voip.ms/article/Sub_Accounts sub account], package and DID number associated with each one.

*'''DID(s) Report''': List of all DID's connected to the reseller clients, with related information.

[[category:guides]]

Reseller Basic Guide

2016-10-19T19:15:09Z

Rod:

'''Reseller Feature – Basic Guide'''

'''This guide is for the Old Version of the Reseller System. For our New Version of the Reseller System, please check our article: [[New Reseller Guide]]'''

The Reseller feature is intended for our users who wants to initiate a business in the VoIP world. This useful tool in our system can help you create and manage accounts for your own clients, you will be able to assign specific rates for them and also provide credentials to register a DID number. You will be able to handle the billing for your customers depending on your needs and how much you want to earn with your business, setting rates and adding credits for them. Additionally we will provide you information about how to customize your Reseller portal to have a more personalized site.

Reports section will help us basically to check the Call detail records from the customers and to view some financial reports related. To activate your Reseller section, you need first to refer from the customer portal go to Reseller menu >> Reseller Main >> Activate

Congratulations, now you have activated the Reseller mode, from now on there is plenty of work to do, this guide will focus on the basic information to run the Reseller section properly. Here you can learn how to:

'''1. Manage Rates and Packages'''

'''2. Manage your clients'''

'''3. Use the client branding and access'''

'''4. Reports'''

== Manage Rates and Packages. ==

On this section we will learn first how to create packages for the reseller feature and how to edit the rates depending on the destination and how to set a benefit for the calls our clients make. It is important to note that all the calls your customers make, are charged to you as per our regulars fees, however on the reseller portal, they will see the charges differently, according to the rates you set here.

=== Create a new package ===

From the customer portal go to Reseller menu >> Manage Rates and Packages >> Create a new Package.

[[File:Package.jpg]]

Package name: Set a package name.

Global Markup: Here you set the benefit you want to have on the calls that your clients make, you can set a fixed amount or a percentage, for this example if you set a 20%, your package will be set as follows ([cost rate] + $ fixed amount) * 20 %.

Example : for 1 cent per minute, the sell rate will (be 0.01 + 0.0000) + 20 % = 0.012

=== Edit Rates ===

If you want to set different rates depending on the destination of the call, you can do this on the Edit Rates section from the package, all the custom rates you enter there will override the global mark up. You can set a custom rate for all the calls to USA or Canada, by country or Calling code.

[[File:Rates.jpg]]

== Manage your clients ==

Now that you have a package, you need to add a client, we will review here how to create a client and associate it with a [http://wiki.voip.ms/article/Sub_Accounts sub account], also we will see where we can add payments and charges for each client, assign a DID number and also the proper way to delete a client account that is not longer needed.

=== Adding a client ===

From the customer portal go to Reseller menu >> Manage Clients acct >> Create a new client account.

[[File:Client.jpg]]

Fill the form with the client information and choose between soft and hard balance, when finished click on create account. Important note: when having the client on hard balance note that the service will be suspended for the client if he is on zero or negative balance.

On the Manage section, you can add a new package or a DID number, to add numbers to your clients you need to order the number first from your customer porta.

The client account has been created, however a [http://wiki.voip.ms/article/Sub_Accounts sub account] per client is needed.

Go to Sub accounts menu >> Create Sub account >>

Create the sub account as usual , when you get to Reseller configuration section, you can choose:

[[File:Resellersub.jpg]]

*Reseller client

*Reseller client Package

*Next Billing date

*And if you decide to charge the set up fees now, check on the box.

If you alreay have a [http://wiki.voip.ms/article/Sub_Accounts sub account] created, you can refer to the client account on the Manage section and add a package, when doing this
you will be able to connect it with an existing sub account.

=== Delete a client ===

You can delete a client on the Edit section from the Client's account, however note that before doing this you need to be sure to unconnect any package or DID number related to the client, you can do this from the Manage section on Client´s account.

=== Adding Payments manually ===

You can add payments manually from the customer portal >> Reseller menu >> Manage Clients acct >> Manage Client >> On client information select, Add payment.

=== Add Charges (manually) ===

You can add charges manually from the customer portal >> Reseller menu >> Manage Clients acct >> Manage Client >> On client information select, Add Charge.

=== Add phone numbers to your clients (DIDs) ===

You can add phone numbers to your clients from the customer portal >> Reseller menu >> Manage Clients acct >> Manage Client >> Package for this clients >> select , Add Phone number >> Connect a DID for charging (please note that you may need to order a DID and have it on your account before you can assign it to a client).

[[File:Addpayment.jpg]]

== Use the client branding and access ==

To show a good image to our customers is always important, for that reason we will see here how to customize our reseller interface to have the best presentation possible. We also see that there is a useful feature on the site that can allow us to integrate our client's payments with a paypal account, so they can be added automatically to the client.

=== Adding a host name ===

In order to personalize the interface with a hostname

1.Configure a host with an A RECORD in your dns that points to 174.120.50.58

2.Enter in the field bellow the fully qualified host + domain name (e.g: interface.mydomain.com)

3.Click "submit"

There is a one time fee of $10.00 for adding your hostname to our system.

=== Branding Management ===

From these section you can set your own header, change colors, Titles background, Titles text color, Content background color, content text color.

[[File:Branding.jpg]]

=== PayPal Integration ===

If you want your clients to deposit directly in your paypal account, you can activate the paypal integration. All payments will be sent to your paypal account and automatically credited in your client interface.

[[File:Paypal.jpg]]

Instructions :
1.Enter your paypal email address in the "paypal email account" field
2.Enter the amounts that you want to offer, without decimal and separated by a coma
3.In your paypal account, go to "Profile" , and under the "Selling preference" section, activate the Instant payment notification(IPN). Enter the url : http://www.voipinterface.net/paynotify.php .
4.We strongly recommend to make a live test with an amount of 1 $.

== Reports ==

Here we can see the Call Detail Records for the calls our customers are making, with the rates we establish on the packages created, also a quick review on a Financial report and the DID numbers associated.

*Reports are divided in the follow categories:

*Global CDR: Display cdr report for all your reseller clients calls. You can see all the calls (incoming and outgoing) from all your clients.

*Per User CDR: Display cdr report for a specific client. This filter help you to search calls only for one specific client.

*Financial report: Month to month revenues ,deposits, costs related to the reseller clients. You can see the financial information by month.

*Sub accounts report: List of all sub accounts connected to the reseller clients ,with related information. Here you will find a list of your clients, [http://wiki.voip.ms/article/Sub_Accounts sub account], package and DID number associated with each one.

*DID(s) Report: List of all DID's connected to the reseller clients, with related information.

[[category:guides]]