VoIP.ms Wiki - User contributions [en]

FreePBX (PJSIP)

2023-12-16T17:28:31Z

Edward: /* General */

= Prerequisites =
== Sub account creation ==
Before configuring your FreePBX Trunk, you will need to have a sub-account ready to be connected.
If you don’t have a sub-account yet, you need to create a new one.
To do so, Log into your VoIP.ms portal, navigate to “Sub-Account” and click on “Create a sub-account”.
Your sub-account needs to have the minimum required in order to work with FreePBX.
: - Username
: - Password
: - Device Type: (Asterisk, IP PBX, Gateway, VoIP Switch)
: - CallerID Number: (I use a system capable of passing its own CallerID)

== Set a DID for Incoming calls ==
In order to received calls from a specific DID and leverage all your PBX functionalities, you needs to route directly your DID to your sub-accounts.
To do so, navigate through the navigation bar and go under “DID Numbers” then “Manage DID”.
To edit a DID, you will need to edit this DID by clicking on the yellow button and apply the proper SIP/IAX account.
Note or select your preferred PoP Server; your DID Numbers need to reflect the same PoP server in order to receive incoming calls.

= Configuration of FreePBX =
== Creating a new trunk ==
On your FreePBX panel, Click on the menu [Connectivity] menu, then [Trunk]. When you are on the trunk page, Click on [+ Add Trunk] and select [+ Add SIP (Chan_pjsip) Trunk].

[[File:FreePBX_PJSIP_Trunk_ADD-PJSIP.png|750px]]

=== General Tab ===
: Trunk Name: This is only to identify your trunk for your own purposes.
: Outbound CallerID: '''"CALLERID NAME" <##########>''' (your 10 digit DID # without any dot)
: Note: The name should be '''max 15 characters''', Must be in '''CAPITAL LETTERS''' without specials characters. Spaces are allowed.

[[File:FreePBX_PJSIP_Trunk_ADD-PJSIP-general.png|750px]]

=== pjsip Settings ===
==== General ====
: Username: ######_username (This will be your VoIP.ms sub-account)
: Secret: This will be the password of your sub-account.
: SIP Server: Your preferred PoP Server.
: SIP Server Port: 5060 or one of our alternative port 5080 or 42872

[[File:FreePBX_PJSIP_Trunk_ADD-PJSIP_pjsipSettings-General.png|750px]]

==== Advanced ====
: From domain: Specify your WAN IP address for your PBX.

== Inbound Route Setup ==
In order to route your DID in your PBX you will need to create a inbound route. To do so, Go to the menu [Connectivity] then [Inbound Route], on this page, click on [Add Inbound Route]
: Description: You can add a description for your own purposes.
: DID Number: This will be your 10 digits DID that you have routed to your sub-account.

[[File:FreePBX_PJSIP_Inbound_ADD-PJSIP.png|750px]]

== Outbound Routes ==
=== Route settings ===
: Route name: Give your outbound route a name for your own purposes. 
: Route CID: '''"CALLERID NAME" <##########>''' (your 10 digit DID # without any dot).
: Note: The name should be '''max 15 characters''', Must be in '''CAPITAL LETTERS''' without specials characters. Spaces are allowed. If a CID in your trunk has not been forced, you need to specify the outbound caller ID used with this outbound route.
: Trunk sequence for Matched routes: You will need to the new VoIP.ms trunk you have created.

[[File:FreePBX_PJSIP_Outbound_ADD-PJSIP.png|750px]]

=== Dial Patterns ===

== Outbound Caller ID Name for call to Canada ==
Since Canada’s provider are not using the CNAM database to pass the caller ID Name, you will need to provide this information directly from your trunk. The way to display a name and a number in a trunk is: "CALLERID NAME" <##########>.

<div style="border-left: 6px solid blue; background-color: lightgrey; width: 85%;">
'''IMPORTANT''':
- We suggest entering your outbound Caller ID Name must be in '''CAPITAL LETTERS'''. This will appears more clearly/visible on some devices.
- You must NOT use any special characters, they will not be displayed.
- Do not exceed '''15 characters''' max! Some of regular Canadian providers will not show more than '''15 characters'''. We suggest shrinking or adapt your caller ID.
- Spaces are allowed in a caller id name.
</div>

For USA destinations, you will need to request a CNAM Database update through the live chat or by submitting a ticket request. Note that the '''same criteria must be respected'''.

''This article was made in collaboration with Chris Sherwood from Crosstalk Solution. You can find his Youtube explanatory video about PJSIP Trunk here : https://www.youtube.com/watch?v=Mu1OxktwURg
and his blog article here : https://crosstalksolutions.com/voip-ms-setup-using-pjsip-on-freepbx/''

Caller ID

2021-09-07T21:01:58Z

Edward:

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/ID_de_l%27appelant Français] ||
[https://wiki.voip.ms/article/Numero_Identificador_(Caller_ID) Español]
|}

== Blog Article ==
[https://wiki.voip.ms/article/What_Is_CNAM_and_How_to_Leverage_It_for_Your_Business%3F What Is CNAM and How to Leverage It for Your Business?]

== 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 need to pass a Caller ID to ensure proper termination of your calls.

There are two types of caller ID and it is important to differentiate them: Caller ID Number (CID) and Caller ID Name (CNAM).

Please note that the Caller ID is only guaranteed when using Premium routes, and only for US48 and Canadian calls. In Canada you may find Caller ID working on some Value routes.

Incoming Caller ID (from people calling you) is addressed below.

__TOC__

== Outgoing Caller ID number ==

Caller ID number is the most common Caller ID type passed. If a more complex system capable of passing its own Caller ID is being used, such as a [[Welcome#PBX|PBX]], the Caller ID field is likely set from the trunk, or one of its extensions.

If you are using devices like [[Welcome#Devices|Analogue Telephone Adapters]], [[Welcome#Devices|IP phones]], or [[Welcome#Softphones|softphones]], the Caller ID number is available to be set from your VoIP.ms account via the customer portal.

=== Caller ID Rules ===

In line with all the rulings in favor of protecting and empowering consumers, VoIP.ms enforces valid Caller IDs as per the NANPA and ITU-T E.164 standards.

Therefore, calls displaying the following Caller ID information will be rejected:

*International calls with Caller ID containing less than 7 digits and greater than 15 digits;
*North America calls with Caller ID not containing exactly 10 digits (exception will be made for 7 digits for 310-xxx numbers);
*North America calls with Caller ID containing 10 digits, but with an unassigned NPA (first 3 digits of the number);
*North America calls with Caller ID containing 10 digits, but where an unassignable NXX is used (i.e. the second block of 3 digits where the first digit is either zero (0) or one (1)).

=== Main account ===

To set the Caller ID number for your Main account, access:
*"[[Account Settings]]" from '''Main Menu''' -> '''Account settings'''
*Go to '''General''' Tab and there set the '''CallerID number'''

<div><ul>
<li style="display: inline-block;"> [[File:AccountSettingsMenu.png|thumb|none|800px|Account settings]] </li>
<li style="display: inline-block;"> [[File:CIDmain.jpg|thumb|none|700px|CallerID field]] </li>
</ul></div>

=== Sub accounts ===

The Caller ID number for a sub account can be set during the creation process, or later by clicking Edit on the [[Sub Accounts|subaccount]].

[[File:CallerIDSubAcc.png|thumb|none|700px|CallerID field in a sub account]]

* '''For calls to US and Canada, a 10-digit Caller ID number is required to ensure proper call termination. The portal's Caller ID field only supports numerical digits.'''
* '''Toll free Caller IDs are not recommended, especially when Calling Toll free numbers, to reduce potential connection issues.'''

If you are using an analog telephone adapter (ATA), IP phone, or softphone, this is where you need to adjust the settings. It is important to ensure that you transmit a valid caller ID to ensure proper termination. Here you will have the option of having one of your incomings DID numbers displayed as an outgoing call identification number. You can also use a personalized number that does not appear in your lists of DID numbers. However, you must respect the standards in force by the regulations to this effect. Always use a number that belongs to you or to that you have permission to use as the outgoing call display number. If you have a PBX that manages the outgoing call identification number, you will have an option for this.

''Note: Using a Toll free outbound caller id is not recommended, especially when calling Toll-Free numbers.''

=== Listen to the current account's Caller ID Number ===

To listen to the caller ID set in the account you're dialing from, please dial '''822''' and the Caller ID number will be read back. Notice that it will depend on your device's dial plan or dial pattern for this to work.

These test calls won't have any cost for you and will appear in your [[Call_Detail_Records | CDR]] as "CallerID Testing".

===Outbound caller ID number using Call Forwarding===

When you chose to forward all your incoming calls from your VoIP.ms number to an external phone number, the original's caller ID number will be transmitted to this external number. However, if the incoming caller is hiding their phone number this call will fail due to the new regulations for CID, on where sending a valid caller ID number is mandatory.

A workaround is to create a filter for '''"Anonymous"''' calls along with another call forwarding entry but this one using the '''"Caller ID number override"'''. This way, incoming calls using a valid CID will be transmitted to your external number showing the original's CID and incoming calls hiding their number will use the filter and the forward with caller ID Override, showing the CID override number and completing the call.

'''To receive "Anonymous" calls when using call forwarding, follow these steps:'''

1. Go to '''DID Numbers>> Call forwarding''' section and create a new call forwarding entry to the same number you're forwarding to, but this new entry will contain the '''"Caller ID override"'''. For the caller ID override, you can type your VoIP.ms number or any valid number you want to see when you receive an anonymous call. You might want to fill the '''"Description field"''' so you can identify this entry on the next step.

2. Thereafter, go to '''DID Numbers>> CallerID filtering''' section and create a new filter.

* For step 1, mark '''"Anonymous CallerID number"''' checkbox
* For step 2, this is optional you can choose to apply this filter to all your numbers or just to a specific one.
* For Step 3, mark the '''"Call forwarding"''' checkbox and make sure to use the call forwarding entry with the caller ID override number.
* Save your filter.

== Outgoing Caller ID name ==

The Caller ID name is an additional information you can pass along with your Caller ID number. This will be received on the callee's end and it could be your given name or the name of your business.

For example: ''"John Smith" <9145551234>''

The sample above 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 a Caller ID name from the VoIP.ms portal.

If you plan to make calls to Canadian numbers, you can simply pass the Caller ID name from your device or system as most of them support this.'''*''' 
You will need to check for a field on the interface from the device to enter this setting, and in case you are using a more advanced system, get assistance to set the outgoing Caller ID name set up.

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.
When you make a call to a US number, you will send a Caller ID number and the system will check the CNAM database for a name matching that same Caller ID number in order to display both name and number to the final phone.

CNAM is only available for some USA numbers. In order to update your Caller ID'''*''' on the CNAM database for your US calls, there is a process to follow which has a cost of $10 USD (one time only fee).
Please contact the VoIP.ms support staff to get further details on what information you need to submit and to confirm if your local US DID is available for a CNAM update.

'''CNAM update is only available for some Local US DIDs. Toll frees cannot have their Caller ID name updated'''.

<div style="border-left: 6px solid blue; background-color: lightgrey; width: 85%;">
'''IMPORTANT''':
- We suggest entering your outbound Caller ID Name must be in '''CAPITAL LETTERS'''. This will appears more clearly/visible on some devices.
- You must NOT use any special characters, they will not be displayed.
- Do not exceed '''15 characters''' max! Some of regular Canadian providers will not show more than '''15 characters'''. We suggest shrinking or adapt your caller ID.
- Spaces are allowed in a caller id name.
</div>

== Incoming Caller ID number and name ==

You will receive the Caller ID number and Caller ID name that the VoIP.ms server receives from the caller, this is exactly what will be sent to you on Incoming calls.
You can always check what Caller ID number VoIP.ms receives by going into your [[Call Detail Records]] to check the incoming calls.

The incoming Caller ID name works almost the same way, except that this is an optional setting that you need to enable per DID number on the [[Manage DID|DID settings]] page. This option is called "CallerID Name Lookup". When enabled, the system will perform a query on the LIBD/CNAM Database, for callers with Canadian or US CID number, in order to find a name matching that CID number. The system then will display the result of this query in the Caller ID name portion of the '''Caller ID''', leading to a "Caller ID name"<5551231234> when people call your number.

If the calling number is already in your [[phone book]], the name will be taken from there instead of doing a CNAM lookup on an external database.

'''IMPORTANT NOTE FOR CANADIAN DIDs'''

The majority of the Canadian DID numbers support CNAM Pass-Through. This means that for your incoming calls the system '''won't do a CNAM query''' (and also not charge you) if the incoming call already has a Caller ID name, even if the DID receiving the call has the CNAM queries enabled in your VoIP.ms customer portal.
----

== Notes ==

:'''Outgoing Caller ID is not guaranteed on calls to Canadian cellular numbers, even when using the Premium route. This is due to the way Canadian carriers work - they sometimes pass a random Caller ID that they have on record, changing the original. This is out of our control as it is the way Canadian carriers handle calls to cellular numbers. There are also issues with incorrect Caller ID being sent on outbound [[toll-free number]] calls from Bell Mobility (and its resellers); the display shows a disconnected Bell number in an area code corresponding to the caller's location instead of the caller's mobile number.'''

[[category:guides]]

OBi300

2021-08-05T20:47:17Z

Edward: /* Manual Configuration Details */

[[File:obi300.png|none|200px|center|link=https://www.polycom.com/voice-conferencing-solutions/voip-adapters/obi300.html|OBi 300]]

'' 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 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.'' 

== Manual Configuration Details ==

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'''.

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.

If done properly, the following window should appear on your screen:

[[File:Obi300_Login.png|300px|thumb|left]]
 

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:Obi300_AutoProv.png|450px|thumb|left|Disabling Auto Provisioning]]
 
After this, save all changes and you are ready to move on to the actual configuration.
 
===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) 
:'''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|<1>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|011xx.|***xxx|xx.|(Mipd)|[^*]@@.)

[[File:Step2.png|550px|thumb|left|ITSP profile, General - click to enlarge]]
 

====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]]
 

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]]
 

===Configuring Voice Services===
 
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]]
 

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.

===NOTE for similar devices===
 
For the Obi212, "PSTN Line" is the default setting under "Physical Interfaces -> PHONE1 Port -> "Primary Line", however the correct option should be "SP1 service" or the SP line that we are configuring.
This may also apply for other models, so please make sure the right option is selected for your case.

[[File:Obiprimaryline.png|550px|thumb|left|ITSP profile, SIP (Register Expires)- click to enlarge]]
 

===Configuring a Voice line using TLS===

'''NOTE''': This section is optional. It explains in detail how to use encrypted traffic in the device. If you are not certain about how encrypted traffic works, or the benefits of encrypting SIP traffic, please contact technical support for more information.

These devices are compatible with TLS, however, some settings need to be adjusted to have it working. This section will be assuming that your SIP account is already enabled to use TLS, if you have not enabled it yet, please follow these instructions before going further:

For more information on how to enable encrypted traffic for the main account, please click on [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Main_Account | Main account]] or more information on how to enable encrypted traffic for the sub-account [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Sub_Account | sub account]]

=== Configuring your OBi device ===

In order to use TLS/call encryption with OBi devices, you'll need to modify the following parameters accordingly. Please note that this is only available for some OBi devices, and the screenshots are from the most recent firmware version.

Under Service Providers > ITSP Profile > SIP use the following values:
:'''ProxySeverPort''': 5061
:'''ProxyServerTransport''': TLS
:'''RegistrarServerPort''': 5061
:'''OutboundProxyPort''': 5061
:'''X_OutboundProxyTransport''': TLS

[[File:TLSOBi1.png|550px|thumb|left|ITSP profile, SIP (TLS)- click to enlarge]]
 

Under Voice Services > SP Service use the following values:
:'''X_KeepAliveServerPort''': 5061
:'''X_SRTP''': Use SRTP Only

[[File:TLSOBi2.png|550px|thumb|left|Voice Services, SP Service (SRTP)- click to enlarge]]

File:Obiprimaryline.png

2021-08-05T20:15:05Z

Edward:

Virtual Fax

2021-05-25T14:24:06Z

Edward: /* Email to Fax */

[[File:Faxhomelogo.png|center]]

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/T%C3%A9l%C3%A9copieur_virtuel Français] ||
[https://wiki.voip.ms/article/Fax_Virtual Español]
|}

The Virtual Fax feature is used to send and receive Faxes (facsimile) with the VoIP.ms service using a DID number specifically dedicated to Fax. 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__

== Blog Article ==
[https://wiki.voip.ms/article/Fax_over_IP_(FoIP)_using_T.38_Protocol Fax over IP (FoIP) using T.38 Protocol]

== Tutorial Video ==

[[Image:VfaxThumbnail.png|200px|link=https://youtu.be/k0WDphvlxbk]]

== 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 currently only be used to send Faxes to Canadian and U.S. numbers. We also cannot guarantee that international will be properly received.'''
* '''Virtual Fax DID Numbers cannot receive regular voice calls nor SMS, but only faxes.'''

== Costs 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 want to calculate how much this service could cost per page faxed, converted into per minute charges, you'll realize that this depends a lot on the destination fax speed and the content faxed (how much content is on the page compared to the blank space). Unfortunately, there are no definite ways to calculate precisely these costs . If you are sending just text, a page can take from 30s to 1 min. If you are faxing multiple pages documents, this will go faster because you don't have 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 increase 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 currently available from other providers.

== 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 25 MB

*The file cannot contain more than 200 pages

*Only one document per fax message can be sent

== Virtual Fax DID Number ==

Virtual Fax works specifically with Fax Numbers only acquired from the VoIP.ms Customer Portal or numbers ported in 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 tiff
-'''IMPORTANT: The maximum file size is 25 MB'''
*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 choice 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 has been completed.

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

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:VfaxDel.png|900px]]

== My Folders ==

In the 'My Folders' section, you can create folders by typing in the folder name of your choice under 'New Folder' and clicking 'Create'.
You will have an overview of your folders, see the dates they were created, the number 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 Address''': If selected, a notification will be sent to the inputted email address that a new fax has been received. You can enter several email address to receive a notification by separating them with a comma ( , ).

'''Attach PDF File''': If selected, a PDF file with the contents of the fax message will be attached to the email.

'''URL Callback''': If selected, fax messages received by your number will sent a POST request to the URL callback provided.

'''Callback Retry''': When selected, we will be expecting an ''OK'' output from your URL callback page as an indicator that you have received the fax message correctly. If we do not receive the ''OK'' letters from your callback page, we will keep sending you the same fax message every 30 minutes for up to 7 days.

'''Fax to SIP''': If selected, fax messages received by your number will be sent to your device (fax machine or PBX) registered on one of our fax servers. For more information, please verify this part of the article: [[Virtual_Fax#T.38_Faxing_.28BETA.29 | T38 faxing]]

[[File:FaxInboundSetup.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.

'''Important:'''

*Only the file attached will be faxed, the body of the email won't be transmitted'''
*If you have a signature at your email, avoid using pictures (logos, photos or images) on it, as it will be set as another attachment at your email and that could cause issues at the outbound fax your email will generate.

'''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|]]

== Forward Inbound faxes to mail ==

To have your inbound faxes forwarded to a given email address, on your customer portal, refer to:
DID Numbers --> Manage DID's. When here, select and edit the given Virtual Fax number and on the first screen you'll be presented with, enable the tick boxes: "Email address" and "Attach PDF file", as well as filling in the email address field with the desired address for the faxes to be attached to.

== T.38 Faxing (BETA) ==

Due to the nature of the technology, '''the number of pages, as well as a copy of the fax document''',
will '''not''' be available in the customer portal when '''sending''' a fax over T.38.
'''Inbound faxing''' over T.38 will still have this information available.

The following are the requirements in order to be able to send and receive faxes via T.38 directly from your own fax phone setup.

:* A VirtualFAX DID number
:* An analog telephone adapter (ATA) with T.38 support.
:* A Fax machine connected to an ATA
:* The ATA must be configured with a SIP account and registered to one of the fax dedicated servers: '''fax1.voip.ms, fax2.voip.ms'''
:* The caller ID number of the SIP account must be configured to match any of the Fax DIDs available in your account. This will be the number used as the sender.
:* A T.38 enabled fax machine or a T.38 enabled SIP trunk are also supported. These will not require an ATA.

To '''receive''' fax messages, edit your Fax DID number, and choose the SIP account where all the incoming faxes will be forwarded.

[[File:faxdiden.png]]

'''Additional notes:'''

:* T.38 is only supported over SIP. IAX2 is not supported.
:* TCP, UDP, TLS/SRTP are fully supported.
:* T.38 Faxing is not limited to 100 outbound faxes per day nor 25mb file size limit.
:* Because the system merely does a pass-through when sending a fax via T.38, is not possible to know if the fax was successful or not.
:* The Fax message will be charged by the duration of the entire call, from the moment it was answered until it was hung up.

== Fax using the Reseller Interface ==

The feature is available for your client through the Reseller interface. You must enable this feature in your package in order to give them the ability to leverage this.

Go under the navigation bar on '''[Reseller]''' then click on '''[Manage Rates & Packages]'''
: [[File:FaxO_Reseller_1.png|thumb|none|300px]]

Click on the Edit button to edit your package, or click on '''[Create a new package]''' to create a new one.

: [[File:FaxO_Reseller_2.png|thumb|none|700px]]

Go under the '''[Reseller System Configuration]''' Tab, and on the section "Type of configuration" select: '''[Package Configuration]''',

: [[File:FaxO_Reseller_3.png|thumb|none|700px]]

Then scroll down and find the feature '''"Manage Faxes"''', and enable it. (If you would like to allow
If you wish to give the possibility to your client to buy new Fax DID via the Reseller interface, enable '''"Order Faxes"'''.

: [[File:FaxO_Reseller_4.png|thumb|none|500px]]

=== Order Faxes ===
----
1) To Order a new Fax Number for your client, or to help your client adding one. Go under the '''[Virtual Faxes]''' menu, then '''[Order Faxes]'''.

[[File:FaxO_Add.png|thumb|none|300px]]

[[File:FaxO_Add_2.png|thumb|none|500px]]

[[File:FaxO_Add_3.png|thumb|none|500px]]

The new Fax DID ordered, will be available in the "Manage Fax Numbers" section.

=== Manage Fax Numbers ===
----
To manage your inbound Fax DID, and send the FAX to Email (PDF). Notifications when recieved. Go under '''[Virtual Faxes]''' then '''[Manage Fax Numbers]'''.
[[File:FaxM_Add.png|thumb|none|300px]]

Where your DID, click on the '''[Edit DID]''' button.

[[File:FaxM_Add_2.png|thumb|none|500px]]

=== Send Fax ===
----
To send a Fax using the reseller interface. Go under the '''[Virtual Faxes]''' at the left navigation bar, then '''[Send Fax]'''.

[[File:FaxS_Add.png|thumb|none|300px]]

Enter the destination number select a Fax number

[[File:FaxS_Add_2.png|thumb|none|500px]]

Click '''[Send Fax]'''

=== My Faxes ===
----
To manage the Incoming/Outgoing Faxes in order to View a fax, delete a fax, send a fax to an email, transfer a fax to a folder... Go under the '''[Virtual Faxes]''' at the left navigation bar, then '''[My Faxes]'''.

[[File:FaxF_Add.png|thumb|none|300px]]

[[File:FaxF_Add_2.png|thumb|none|500px]]

=== Email to Fax ===
----
To use the feature Email To Fax. Go under the '''[Services]''' at the left navigation bar, then on '''[Virtual Faxes]''', then '''[Email to Fax]'''.

[[File:FaxE_Add.png|thumb|none|300px]]

Click on the "'''Add Email to Fax'''" tab. Check the '''Enable''' box. Enter the email address. Select the fax number that should be associated with this email.

[[File:FaxE_Add_2.png|thumb|none|500px]]

Click '''[Save Email to Fax]'''

To send a fax via email through the reseller's "email to fax" option, the instructions and parameters are basically the same than regular email to fax with the exception of the destination email address, which for this case is fax@voipinterface.net .

Choosing Server

2021-03-03T17:12:16Z

Edward: /* IPs */

[[File:ChooseServerImg.png|thumb|none|1280px|VoIP.ms servers]]

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Choisir_un_serveur Français] ||
[https://wiki.voip.ms/article/Elegir_servidor Español]
|}

= 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.

'''Note:''' Please bear in mind that some servers might not be available for your DID number to be used as POP (Point of presence) at the ''Manage DIDs'' section.
Make sure that your SIP/IAX device and your phone number are pointing to the same server.

=== IPs ===

<div><ul>
<li style="display: inline-block; vertical-align:top;">
'''Canada'''
*Montreal 1, QC ('''montreal.voip.ms''') 192.175.96.73
*Montreal 2, QC ('''montreal2.voip.ms''') 192.175.96.74
*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
*Montreal 9, QC ('''montreal9.voip.ms''') 67.205.74.184
*Montreal 10, QC ('''montreal10.voip.ms''') 67.205.74.187
*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
*Toronto 9, ON ('''toronto9.voip.ms''') 158.85.70.154
*Toronto 10, ON ('''toronto10.voip.ms''') 158.85.70.158
*Vancouver 1, BC ('''vancouver.voip.ms''') 162.213.157.220
*Vancouver 2, BC ('''vancouver2.voip.ms''') 162.213.157.117
*Vancouver 3, BC ('''vancouver3.voip.ms''') 162.213.157.82
</li>

<li style="display: inline-block; vertical-align:top;">
'''United States'''
*Atlanta 1, GA ('''atlanta.voip.ms''') 75.127.65.130
*Atlanta 2, GA ('''atlanta2.voip.ms''') 209.217.224.50
*Chicago 1, IL ('''chicago.voip.ms''') 69.162.175.27
*Chicago 2, IL ('''chicago2.voip.ms''') 69.162.175.28
*Chicago 3, IL ('''chicago3.voip.ms''') 69.162.175.29
*Chicago 4, IL ('''chicago4.voip.ms''') 208.100.39.55
*Chicago 5, IL ('''chicago5.voip.ms''') 50.31.115.149
*Chicago 6, IL ('''chicago6.voip.ms''') 50.31.115.150
*Chicago 7, IL ('''chicago7.voip.ms''') 50.31.115.151
*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''') 23.239.211.90
*Denver 2, CO ('''denver2.voip.ms''') 64.27.52.226
*Houston, TX ('''houston.voip.ms''') 173.193.85.18
*Houston 2, TX ('''houston2.voip.ms''') 173.193.85.19
*Los Angeles 1, CA ('''losangeles.voip.ms''') 96.44.149.186
*Los Angeles 2, CA ('''losangeles2.voip.ms''') 96.44.149.202
*Los Angeles 3, CA ('''losangeles3.voip.ms''') 64.188.6.162
*Los Angeles 4, CA ('''losangeles4.voip.ms''') 64.188.6.170
*New York 1, NY ('''newyork.voip.ms''') 72.251.239.196
*New York 2, NY ('''newyork2.voip.ms''') 72.251.239.205
*New York 3, NY ('''newyork3.voip.ms''') 72.251.239.206
*New York 4, NY ('''newyork4.voip.ms''') 72.251.239.207
*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
*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''') 104.129.57.250
*Seattle 2, WA ('''seattle2.voip.ms''') 173.205.93.122
*Seattle 3, WA ('''seattle3.voip.ms''') 173.205.93.226
*Tampa, FL ('''tampa.voip.ms''') 162.254.144.173
*Tampa 2, FL ('''tampa2.voip.ms''') 162.254.144.176
*Tampa 3, FL ('''tampa3.voip.ms''') 23.111.187.139
*Tampa 4, FL ('''tampa4.voip.ms''') 23.111.166.202
*Washington 1, DC ('''washington.voip.ms''') 169.62.41.189
*Washington 2, DC ('''washington2.voip.ms''') 169.62.41.187
</li>

<li style="display: inline-block; vertical-align:top;">
'''International'''
*Amsterdam, NL ('''amsterdam.voip.ms''') 66.212.22.42
*London, UK ('''london.voip.ms''') 159.8.157.212
*Sydney, AU ('''sydney1.voip.ms''') 168.1.73.84
*Paris, FR ('''paris.voip.ms''') 159.8.85.180
</li>
</ul></div>

===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.

= 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|thumb|none|600px|Ping]]

==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) =
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)
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.
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.

=== Bash Script To Handle The Mac Ping Output Format ===

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.

<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 server list text file should be formatted with one host name per line.
# The list of voip.ms servers is available at https://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 $USER_FILE ]]
then
# Bash 3.x in macOS does not support readarray, need to do cumbersome array loops instead
while IFS= read -r servers; do
serverList+=( "$servers" )
done < <(grep -Eo '^\b[[:alpha:]]+?[[:alnum:]]\.voip\.ms\b$' "$USER_FILE" | \
grep -v '^\s*#' | awk NF | sort)
else
# N.B. The script looks for the html boldface tags 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 servers; do
serverList+=( "$servers" )
done < <(curl -sm 10 https://wiki.voip.ms/article/Choosing_Server | \
grep -E '([[:alpha:]]+?[[:alnum:]]\.voip\.ms)' | \
tr "<>" " " | awk '{print $(NF-3)}' | sort )
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 "^${server}" | cut -d ":" -f1)
while IFS= read -ra idx; do
keys+=( "${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 pings; do
pingList+=( "$pings" )
printf "%-64s %5d %2d/%-2d\n" "$pings" \
"$((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: 169.62.41.189] is alive ( 98.28 ms)
washington2.voip.ms [ip: 169.62.41.187] 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 ===

Dec 2017 - A bug in the code shown washington2.voip.ms as the best server, this was corrected.

<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"
Clear-Variable best* -Scope Global #Clear the best* variables in case you run it more than once...
#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","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)
{
#Add one to the counting variable....we are on server #1...then server 2, then server 3 etc...
$k++
#Update the progress bar
Write-Progress -Activity "Testing Server: ${server}" -status "Testing Server $k out of $servercount" -percentComplete ($k / $servercount*100)
#Counting variable for number of times we tried to ping a given server
$i = 0
Do{
#assume a failure
$pingsuccess = $false
$i++ #Add one to the counting variable.....1st try....2nd try....3rd try etc...
Try{
#Try to ping
$currentping = (test-connection $server -count 1 -ErrorAction Stop).responsetime
#If success full, set success variable
$pingsuccess = $true
}
#Catch the failure and set the success variable to false
Catch {
$pingsuccess = $false
}
}
#Try everything between Do and While up to 5 times, or while $pingsuccess is not true
While($pingsuccess -eq $false -and $i -le 5)
#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 = $currentping
If($pingsuccess -and ($currentping -lt $bestping -or (!($bestping)))){
#If this is the best ping...save it
$bestserver = $server #Save the best server
$bestping = $currentping #Save the best ping results
}
write-host "tested: $server at $currentping ms after $i attempts" #write the results of the test for this server
}
write-host "`r`n The server with the best ping is: $bestserver at $bestping ms`r`n" #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]]

Asterisk SIP

2019-11-19T16:13:50Z

Edward: /* Asterisk TLS/SRTP (SIP) */

==Asterisk (SIP)==

===sip.conf===

<nowiki>
[general]
register => 100000:johnspassword@atlanta.voip.ms:5060

[voipms]
canreinvite=no
context=mycontext
host=atlanta.voip.ms ;(one of our multiple servers, you can choose the one closer to your location)
secret=johnspassword ;your password
type=peer
username=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
trustrpid=yes
sendrpid=yes
insecure=invite
nat=yes
</nowiki>

*Note: As of FreePBX/Asterisk version 2.9.0.7 if ;comments are placed after the config entries as above it may prevent outgoing call's with the end points getting "All circuits are busy". Remove the ;comments and the trunk will send the calls with no errors.

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk TLS/SRTP (SIP)==

1. In order to use these devices with encryption, besides having to [[Call Encryption - TLS/SRTP | enable the SIP account in your VoIP.ms customer portal]], there are some settings you will have to modify in your device's configuration.

2. Once your account/sub-account has Encrypted traffic enabled, the system has to be configured use/send the traffic through TLS and SRTP.

For the registration over TLS, you need to define the protocol the PBX will use in the general config.

[general]
register => tls://100000:johnspassword@atlanta1.voip.ms:5061

For the outbound part, add the following lines to the peer details.

encryption=yes
transport=tls

See example below:

<nowiki>
[voipms]
encryption=yes
transport=tls
canreinvite=no
context=mycontext
host=atlanta1.voip.ms ;(one of our multiple servers, you can choose the one closer to your location)
secret=johnspassword ;your password
type=peer
username=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
trustrpid=yes
sendrpid=yes
insecure=invite
nat=yes
</nowiki>

'''Note:''' When using TLS is very important to specify the number of the server, in case the name you have chosen doesn't use the number 1 you need to add it.

==Asterisk IP Auth. (SIP)==

===sip.conf===

Note: You'll need to create a sub account to use IP Auth

<nowiki>
[voipms]
canreinvite=nonat
context=mycontext
host=atlanta.voip.ms ;(one of our multiple servers, you can choose the one closer to your location)
type=peer
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support g729
nat=yes
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>

==Known Issues==

===Getting Message Waiting Indicator to Work with VoIP.ms Voicemail===

VoIP.ms sends MWI notifications unsolicited. Please make the following changes in your configurations.

Remove this Line please.

[general] Section

mwi => 123456:mypassword@losangeles.voip.ms/65000

Add this Line please.

[voipms] Section

unsolicited_mailbox=65000

== Local extension calling ==

(User submitted) In order for this to work, you must fill out "User Context and User Details"

User Context Name: 100000 (your account)

[100000] your account
type=user
auth=md5
notransfer=yes
disallow=all
allow=gsm&ulaw
trunk=yes
secret=****
context=from-trunk

Asterisk SIP

2019-11-19T15:41:46Z

Edward:

==Asterisk (SIP)==

===sip.conf===

<nowiki>
[general]
register => 100000:johnspassword@atlanta.voip.ms:5060

[voipms]
canreinvite=no
context=mycontext
host=atlanta.voip.ms ;(one of our multiple servers, you can choose the one closer to your location)
secret=johnspassword ;your password
type=peer
username=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
trustrpid=yes
sendrpid=yes
insecure=invite
nat=yes
</nowiki>

*Note: As of FreePBX/Asterisk version 2.9.0.7 if ;comments are placed after the config entries as above it may prevent outgoing call's with the end points getting "All circuits are busy". Remove the ;comments and the trunk will send the calls with no errors.

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk TLS/SRTP (SIP)==

1. Make sure your Main account or sub-account has "Encrypted SIP Traffic" enabled. Bear in mind, if this setting is enabled and you use UDP/TCP you will be rejected with error code 488. Enable this for the Main Account at '''Main Menu>> Account settings>> Advanced tab''' and for a sub-account at '''Sub accounts>> Manage sub-accounts''' and by clicking on the orange icon with a pen and click at "Advanced Options Click here to display"

[[File:Mainacc encryp.png|thumb|none|300px|Click to enlarge]]

[[File:Subacc encryp.png|thumb|none|300px|Click to enlarge]]

2. Now that your account/sub-account has this setting enabled, your device only needs to send TLS and SRTP.

For the registration over TLS, you need to define the protocol the PBX will use in the general config.

[general]
register => tls://100000:johnspassword@atlanta.voip.ms:5061

For the outbound part, add the following lines to the peer details.

encryption=yes
transport=tls

See example below:

<nowiki>
[voipms]
encryption=yes
transport=tls
canreinvite=no
context=mycontext
host=atlanta.voip.ms ;(one of our multiple servers, you can choose the one closer to your location)
secret=johnspassword ;your password
type=peer
username=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
trustrpid=yes
sendrpid=yes
insecure=invite
nat=yes
</nowiki>

==Asterisk IP Auth. (SIP)==

===sip.conf===

Note: You'll need to create a sub account to use IP Auth

<nowiki>
[voipms]
canreinvite=nonat
context=mycontext
host=atlanta.voip.ms ;(one of our multiple servers, you can choose the one closer to your location)
type=peer
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support g729
nat=yes
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>

==Known Issues==

===Getting Message Waiting Indicator to Work with VoIP.ms Voicemail===

VoIP.ms sends MWI notifications unsolicited. Please make the following changes in your configurations.

Remove this Line please.

[general] Section

mwi => 123456:mypassword@losangeles.voip.ms/65000

Add this Line please.

[voipms] Section

unsolicited_mailbox=65000

== Local extension calling ==

(User submitted) In order for this to work, you must fill out "User Context and User Details"

User Context Name: 100000 (your account)

[100000] your account
type=user
auth=md5
notransfer=yes
disallow=all
allow=gsm&ulaw
trunk=yes
secret=****
context=from-trunk

Asterisk SIP

2019-11-19T15:34:24Z

Edward:

==Asterisk (SIP)==

===sip.conf===

<nowiki>
[general]
register => 100000:johnspassword@atlanta.voip.ms:5060

[voipms]
canreinvite=no
context=mycontext
host=atlanta.voip.ms ;(one of our multiple servers, you can choose the one closer to your location)
secret=johnspassword ;your password
type=peer
username=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
trustrpid=yes
sendrpid=yes
insecure=invite
nat=yes
</nowiki>

*Note: As of FreePBX/Asterisk version 2.9.0.7 if ;comments are placed after the config entries as above it may prevent outgoing call's with the end points getting "All circuits are busy". Remove the ;comments and the trunk will send the calls with no errors.

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk TLS/SRTP (SIP)==

1. Make sure your Main account or sub-account has "Encrypted SIP Traffic" enabled. Bear in mind, if this setting is enabled and you use UDP/TCP you will be rejected with error code 488. Enable this for the Main Account at '''Main Menu>> Account settings>> Advanced tab''' and for a sub-account at '''Sub accounts>> Manage sub-accounts''' and by clicking on the orange icon with a pen and click at "Advanced Options Click here to display"

[[File:Mainacc encryp.png|thumb|none|300px|Click to enlarge]]

[[File:Subacc encryp.png|thumb|none|300px|Click to enlarge]]

2. Now that your account/sub-account has this setting enabled, your device only needs to send TLS and SRTP.

For the registration over TLS, you need to define the protocol the PBX will use in the general config.

<nowiki>
[general]
register => tls://100000:johnspassword@atlanta.voip.ms:5061
</nowiki>

For the outbound part, add the following lines to the peer details.

encryption=yes
transport=tls

See example below:

<nowiki>
[voipms]
encryption=yes
transport=tls
canreinvite=no
context=mycontext
host=atlanta.voip.ms ;(one of our multiple servers, you can choose the one closer to your location)
secret=johnspassword ;your password
type=peer
username=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
trustrpid=yes
sendrpid=yes
insecure=invite
nat=yes
</nowiki>

==Asterisk IP Auth. (SIP)==

===sip.conf===

Note: You'll need to create a sub account to use IP Auth

<nowiki>
[voipms]
canreinvite=nonat
context=mycontext
host=atlanta.voip.ms ;(one of our multiple servers, you can choose the one closer to your location)
type=peer
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support g729
nat=yes
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>

==Known Issues==

===Getting Message Waiting Indicator to Work with VoIP.ms Voicemail===

VoIP.ms sends MWI notifications unsolicited. Please make the following changes in your configurations.

Remove this Line please.

[general] Section

mwi => 123456:mypassword@losangeles.voip.ms/65000

Add this Line please.

[voipms] Section

unsolicited_mailbox=65000

== Local extension calling ==

(User submitted) In order for this to work, you must fill out "User Context and User Details"

User Context Name: 100000 (your account)

[100000] your account
type=user
auth=md5
notransfer=yes
disallow=all
allow=gsm&ulaw
trunk=yes
secret=****
context=from-trunk

Dialing Codes

2019-11-08T21:39:41Z

Edward: /* iNums */

[http://voip.ms/ VoIP.ms] has different dialing codes depending on the destination you want to reach, and you can use these codes from your main or sub account depending on your needs. It is important to mention that there are two main distinctions between the routes when dialing local and international calls. Calls to United States 48 (except Hawaii & Alaska) and Canada are considered local calls. All the other destinations will be considered International and will use the corresponding route.

== Local calls (USA48/Canada) ==

For calls to USA48 & Canada, you only need to dial the complete 10 digits number, optionally you can dial using 11 digits, adding the prefix 1 before the number.

Example: 1514-316-xxxx or 514-316-xxxx (do not use the dashes when dialing)

On the case of Canada, from the Account Settings >> Account Routing, it can be set a route for these calls, there is an option to use value or premium route, and also, it's possible to switch the routing the call will use by dialing the prefix 033 (Value) and 044 (Premium) followed by the 11 digits number. The US is only available on the Premium route.

*033 + 1 + Area Code + number: Canada Value (override account setting)

*044 + 1 + Area Code + number: Canada Premium (override account setting)
(+ signs used as reference only. Do not include + signs when dialing)

Cost with the '''Canada''' '''Value''' route: Starting at $0.0052
Cost with the '''Canada''' '''Premium''' route: $0.0090
Cost with the '''USA48''' route ('''Premium'''): $0.0100

''Note (except Yukon, North West Territories & Nunavut are considered in this section but have a different rate ($0.1900 per minute).''

'''USA48: The contiguous United States are the 48 U.S. states on the continent of North America that are south of Canada, plus the District of Columbia. The term excludes the states of Alaska and Hawaii, and all off-shore U.S. territories and possessions, such as Puerto Rico.
'''

== International calls ==

There are different codes that can be used to dial International numbers (Outside US48 & Canada). Hawaii and Alaska are considered in this section, since any change made on the [[Account Settings]] >> Account Routing for International calls, will affect also these destinations, this means if we set to use premium route, calls to these destinations will use also premium route, but they can be dialed as local US numbers (10 or 11 digits).

Note: Some International destinations can be dialed using only the prefix 1, this applies for countries part of the NANPA. 011 & 00 are for the rest of countries.
E.g. when dialing Hawaii, Alaska, Caribbean countries an U.S. Territories.

For dialing to countries outside US & Canada, we can follow these codes:

*011 + Country Code + number: International
*00 + Country Code + number: International
*033 + Country Code + number: International Value (override account setting)
*044 + Country Code + number: International Premium (override account setting)

(+ signs used as reference only. Do not include + signs when dialing)

Example Dialing to Mexico: 011+country code+number 011-52-9999xxxxxx (do not use the dashes when dialing)

== Special codes ==

Some special codes that you can use with the service:

=== [[Voicemail]] Access Codes ===
*97 to access directly the Mailbox associated to the account you are dialing from. (Will prompt for Password only)

*98 to access your Voicemail and choose one of your Mailbox accounts. (Will prompt for Mailbox ID and Password)

If you don't have access to our VoIP network and would like to check your Voicemail, you can simply dial your number. Once the Voicemail system answers your call, press the asterisk key (*).

=== Account Balance ===

Dial *225 (*bal): This code allow you to access your VoIP.ms Account Balance. It can be Enable or Disable on [[Sub Accounts]].

=== Echo & DTMF test ===

4443 (Echo Test): This code is used to access the echo test, with a new account this code can be dialed even without funds, it is useful to verify the quality on your line.

4747 (DTMF Test): This code is used to access the dtmf test, with a new account this code can be dialed even without funds, it is useful to verify if the dtmf is configured properly.

=== iNums ===

iNums can be ordered for free from your account, on the order DID section, one iNum is available per account and calls (incoming) are free of charge. You can get more information also at http://www.inum.net/. iNum stands for international Number, making use of the +883 global country code. iNums are reachable from a growing list of providers.

Example 011+883+iNum or 00+883+iNum: 011-883-51000134xxxx (do not use the dashes when dialing)

== Service codes for Canada ==

'''1-555-555-0911:''' Test CallerID and [[e911]] Test: to test if your caller Id is working properly you can dial this number, also this recording will let you know if your number is activated with the e911 service.

'''411:''' Directory Assistance (Must be enabled in [[Account Settings]]), when enabled users can dial the directory assistance at a cost of $0.99 per call.

'''311:''' Non-Emergency Police, Municipal and Other Governmental Services (Canadian Servers)

The service has been made available in the following communities (with starting date), please let us know if you experience an issue:

Calgary, Alberta (18 May 2005)
Edmonton, Alberta (16 December 2008)
Fort St. John, British Columbia (14 November 2006)
Gatineau, Quebec (22 June 2005)
Greater Sudbury, Ontario (12 February 2007)
Halifax Regional Municipality, Nova Scotia (15 November 2012)[8]
''Halton Region, Ontario (18 March 2008). The calls to 311 from the Halton Region are not currently supported with VoIP.ms. Please use 1-866-4HALTON
(1-866-442-5866) or 905-825-6000''
Laval, Quebec (3 October 2007)
Montreal, Quebec (mid-December 2007)
Ottawa, Ontario (19 September 2005)
Regional Municipality of Peel, Ontario (5 October 2009)
St. John's, Newfoundland and Labrador (27 June 2006)
Toronto, Ontario (24 September 2009)
Vancouver, British Columbia
Windsor, Ontario (22 August 2005)
Winnipeg, Manitoba (16 January 2009)[9]

'''511:''' Provision of Weather and Traveler Information Services (Canadian Servers)

'''811:''' Non-Urgent Health Teletriage / telehealth Services (Canadian Servers)

'''Important Notes:'''

* The services can only be Called from Canadian servers. There are plans on adding service codes to US servers as soon as possible. This doesn't apply to the 411 service.
* The services rely on your [http://wiki.voip.ms/article/Caller_ID CallerID] to route your call to the proper service line, therefore, some services may not be supported in your province or for your specific area.
* The are no charges applied to your account when you call 311, 511 or 811.

== Canada 310 service ==

Exchange 310 is a special code, these numbers are only reachable from the area where they belong, from the service, you will need to dial only the 7 digits to properly reach these numbers, pass a valid Canadian CallerID and be registered in Chicago, Seattle or any of the Canadian servers.

Example: Dial 310-xxxx (do not use the dashes when dialing), if you pass a caller id in area code 514, dialing seven digits 310-xxxx is like dialing 514-310-xxxx.

== Special cases ==

Termination is not supported to 1900 numbers, premium numbers and International Toll Free numbers (out of USA and Canada). If you have questions about please fell free to contact our support staff via [mailto:support@voip.ms | ticket support ] or live chat.

Cisco SPA112

2019-08-29T15:21:20Z

Edward:

[[File:SPA112.jpg|300px|thumb|left|Cisco SPA112]]
<div style="width:100%;overflow:hidden;clear:both"></div>

: '''There have been some reports of issues with this device from customers of both VoIP.ms and other providers.'''
: '''Make sure to install the latest firmware from [https://software.cisco.com/download/release.html?mdfid=283998771&softwareid=282463187&release=1.4.1%20SR3&relind=AVAILABLE&rellifecycle=&reltype=latest Cisco Software].'''
: '''Version 1.1 or later should be used for proper Caller ID support. '''
: '''Some People have reported issues using Firefox to Configure this device; please try Chrome or IE. '''

==Documentation==
[https://www.cisco.com/c/en/us/support/unified-communications/spa112-2-port-phone-adapter/model.html#~tab-documents Official Configuration guide / User guide / Data Sheets]

== Configuration Details ==

=== Getting the IP address of your device ===

There are two ways to retrieve the IP address of your Cisco SPA112: via analog phone menu, and via your internet router.

{| class="wikitable"
! style="text-align: center;" | Analog phone interface
! style="text-align: center;" | Internet Router
|-
|
# Attach the Cisco SPA112 to your network and attach an analog phone to one of the phone ports, then do the following:
## Dial **** from the phone, even though there is no dial tone.
## When you hear "System Configuration Menu," dial 1 1 0 # slowly. The current IP address will be read back. (e.g. 192.168.X.X)

'''If you hear 0.0.0.0, check your network connection and DHCP server. If necessary, a static IP address'''
'''can be assigned by using option 111# at the IVR, then entering the IP address with your phone's keypad'''
'''(for example, 10*1*27*2 for 10.1.27.2). The network mask can be set with option 121# and the default'''
'''gateway can be sent with option 131#'''
Learn more about the IVR menu options from the https://supportforums.cisco.com/docs/DOC-9900 document.

Be sure to allow at least a minute or two for the box to initialize; even a correctly configured and installed SPA112/122 will give no power or dialtone to the phone until initialization is complete.

Note that the SPA122 is basically a SPA112 with a second network port, intended for installation between a local network hub (LAN) and an upstream Internet (WAN) connection. The SPA122 may be configured as either a "NAT" or "bridge". Depending on configuration, this leaves the SPA122 with two addresses; a local area network address (such as 192.168.15.1) and an outside Internet address. Dialing ****110# will give one address, ****210# will give the other.
|
# Attach the Cisco SPA112 to your network
# Access your router's remote administration interface via your web browser (typical addresses may be <code>192.168.0.1</code> or <code>192.168.1.1</code>). Refer to your router instructions for more information.
# Enter your username/password if asked. If you have not set one, then it is likely the unchanged default password.
# In the router's menu, there should be a page showing a list of connected clients, with their internal IP address. Find the entry corresponding to the Cisco SPA112. It should identify itself in the list as "SPA112"
# Navigate to this IP address via your web browser
|}

=== Accessing to the device's settings page ===

Open your web browser and go to the IP address you obtained in step 1 (for example, http://192.168.2.1).
The default username is admin, and the default password is also admin.

For the SPA122, if one address does not return the web interface (or has some functions greyed/disabled), try the other.

=== Configuring the Quick Setup screen ===

Go to Quick Setup and configure Line 1 as follows:

'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms [http://wiki.voip.ms/article/Choosing_Server servers])

'''Display Name:''' Your name

'''User ID:''' 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)

'''Password:''' Your VoIP.MS SIP Password

'''Dial Plan:''' (911S0|310xxxx|<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|***xxx|*xx.|[3468]11|822|0|00|4xxx|**275*x.|xxxxxxxxxxxx.)

('''''Note''''': Replace 555 in the dial plan with your area code, See [[Dial Plan for Linksys ATAs]] for details.)

Click Submit to save settings.

[[File:quick_setup_test.png|800px|thumb|left|Quick Setup Page - Click to enlarge]]
 

=== Configuring the Voice Line ===
==== Nat Settings ====

Click on Voice, then Line 1

Set '''NAT Mapping Enable''' to Yes, then set '''NAT Keep Alive Enable''' to Yes. If your environment does not use NAT, you can leave these settings disabled. These features can usually be disabled on the SPA122 if it is connected directly to your modem since its traffic will not be subject to NAT in this configuration.

If using the second phone line on an SPA122 device, change the SIP Port for one of the lines to e.g. 5080.

[[File:VL_1_nat_settings.png|800px|thumb|left|NAT Settings - Click to enlarge]]
 

==== Proxy and Registration ====

Under '''Proxy and Registration''', set the server you will use as registration server and the proper values for the Register Expires and Proxy Fallback Intvl:

'''Proxy''': atlanta.voip.ms (one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location)
'''Register Expires''' to 300
'''Proxy Fallback Intvl''' to 300

Also confirm the following settings: 
'''Register:''' YES 
'''Use DNS SRV:''' NO 
'''DNS SRV Auto Prefix:''' NO

[[File:VL_2_proxyAndRegistration.png|800px|thumb|left|Proxy and Registration - Click to enlarge]]
 

Click Submit to save these changes

==== Subscriber Information ====

In this section please confirm that you have the proper account information:

'''Display Name''': Your name (that will be shown as callerID name)
'''User ID''': 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)
'''Password''': Your VoIP.ms SIP Password

[[File:VL_3_subscriberInformation.png|800px|thumb|left|Subscriber Information - Click to enlarge]]
 

==== Audio Configuration ====

You can verify or change the audio codec that will be used with the calls. Please verify that you have the same codec selected in your SIP account's settings.

Preferred codec: g711u (or G729)

[[File:VL_4_audioConfig.png|800px|thumb|left|Audio configuration - Click to enlarge]]
 

==== Dial Plan ====

We recommend to use this dial plan.

(911S0|310xxxx|<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|***xxx|*xx.|[3468]11|822|0|00|4xxx|**275*x.|xxxxxxxxxxxx.)

('''''Note''''': Replace 555 in the dial plan with your area code, See [[Dial Plan for Linksys ATAs]] for details.)

[[File:VL_5_dialPlan.png|800px|thumb|left|Dial Plan - Click to enlarge]]
 

You can create your own dial plan if you need it. See [[Dial Plan for Linksys ATAs]]

=== Optional settings ===

==== Outbound audio "breaking up". ====

Cisco's defaults (SIP T1 = 0.5 sec, RTP packet size 0.030 on most Sipura adapters) respectively may cause unnecessary retransmission of commands over connections with high latency and create issues with outbound audio "breaking up".

Click '''Voice''', then go to '''SIP'''.

Set SIP Timer Values (sec)

SIP T1: 1

Set RTP Parameters

RTP Packet Size: 0.02
RTP Port Min: 10000
RTP Port Max: 20000

Click Submit to save the changes

[[File:VS_sipAndRTP.png|800px|thumb|left|SIP Values - Click to enlarge]]
 

==== Caller ID display showing incorrect time ====

Sometimes the hour shown in your caller ID is incorrect. Following this suggestion usually solves the issue:

Enter your device's settings and click '''Network Setup''', then go to '''Basic Setup''', then click '''Time Settings'''

Set your time zone and (optional) NTP settings. A good time server choice is 0.pool.ntp.org. Setting the proper time zone will ensure that the time which appears on your Caller ID display is correct.

===Configuring a Voice line using TLS===

'''NOTE''': This section is optional. It explains in detail how to use encrypted traffic in the device. If you are not certain on how encrypted traffic works, or the benefits of encrypting SIP traffic, please contact technical support for more information.

These devices are compatible with TLS, however, some settings need to be adjusted to have it working. This section will be assuming that your SIP account is already enabled to use TLS, if you have not enable it yet, please follow these instructions before going further:

For more information on how to enable encrypted traffic for the main account, please click on [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Main_Account | Main account]] or more information on how to enable encrypted traffic for the sub account [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Sub_Account | sub account]]

====Verifying the device's Firmware version====

First, check your firmware's version. This is, from your Device's Configuration Utility at ''Status >> System Information''

We '''strongly recommend''' to use the latest firmware version available, up today is the ''1.4.1 (SR3) Apr 3 2019''. If you do not have this version, you may consider its upgrade.

[[File:SPA_FW_Ver.png|800px|thumb|left|Click to enlarge]]
 

==== Enabling TLS for the line ====

Go to the User's line you will use (If you use the Line 1 go to User 1) and navigate to the '''''Supplementary Service Settings''''', there set:
:''Secure Call Setting'': '''''yes'''''

[[File:SPA_User_SCS.png|800px|thumb|left|Click to enlarge]]
 

==== Configuring the transport and port ====

Go to the line you will be using with TLS and navigate to the section ''SIP Settings'', then set:
:''SIP Transport'' : '''''TLS'''''
:''SIP Port'' : '''''5061'''''

[[File:SPA_Voice_Line.png|800px|thumb|left|Click to enlarge]]
 

==== CA Certificate ====

As per CISCO's requirements, a CA certificate is needed to use Secure calls with the SPA112's device. To achieve this you will need to import the CA Cert.

Go to ''Voice >> Provisioning'' and once there navigate to '''''CA Settings''''', at ''Custom CA URL'' enter the following:

http://spa1xx.voip.ms/cca.pem

[[File:SPA_Prov_CA.png|800px|thumb|left|Click to enlarge]]
 

Click ''Submit'', the device will reboot and after that will register and you will be ready to use it with TLS

==== Secure Call Indication Tone ====

Once the secure call feature is enabled, during all the duration of your calls you will hear a couple of tones (beeps), this is normal and beyond ''VoIP.ms'' control, however you can disable this notification going to: ''Voice >> Regional >> Secure Call Indication Tone''

'''''Note:''''' Please notice that this setting is not an on/off, you will need to remove all the line
In case you need to set it back, the default value is
397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)

[[File:SPA_Regional_Tone.png|800px|thumb|left|Click to enlarge]]
 

Click Submit to save the changes

==Known Issues==

=== '''Phone will not ring on handset''' ===

Sometimes the Phone you are using is designed for a certain voltage and ring waveform. If someone tries to call you and the phone appears to be ringing for the caller but your phone never rings, please follow these steps to hopefully resolve this issue for you.

Step 1: First access the SPA web interface.

Step 2: Click on the '''Admin Login''' and then click on '''(switch to advanced view)'''

Step 3: Click on your Regional tab on the top menu.

Step 4: Go halfway down the page until you see the heading '''Ring and Call Waiting Tone Spec'''

[[File:Pap2Ring.jpg|800px|thumb|left| Ring and Call Waiting - Click to enlarge]]
 

Step 5: Change the Ring Waveform setting to Sinusoid or Trapezoid, the opposite of what you have set. You can also change the Ring Voltage in increments of 5 to 90 or 95.

Step 6: Save settings and test an incoming call.

=== Receiving Unwanted Calls in the middle of the Night (i.e. CallerID 100) that do not appear in your CDR: ===

These calls are not going through our Network but rather through the internet directly to your ATA Device.

Please look under the Voice>> Line 1 page in your SPA device for the following setting: Restrict Source IP and make sure it's enabled.

This way the ATA device will block any traffic not coming from our servers.

[[File:VL_1_restrictSourceIP.png|800px|thumb|left|Restrict IP - Click to enlarge]]
 

=== Firmware Upgrade ===

SPA112 and SPA122 adapters were distributed with outdated (1.0.x) firmware at least as late as 2012; affected boxes will not show Caller ID on any inbound call, even though the caller names and numbers are visible in the call detail record on VoIP.ms (or other provider's) web interface.

Updated firmware is available from the Cisco site [https://software.cisco.com/download/release.html?mdfid=283998771&softwareid=282463187&release=1.4.1%20SR3&relind=AVAILABLE&rellifecycle=&reltype=latest Cisco Firmware] as a .ZIP archive which contains two files (a .BIN with the actual firmware and a .PDF with documentation). Download and unzip this file. Go to the 'administration' tab on the web interface (on the SPA122, this needs to be done from the LAN side with SPA122's built-in networking set to NAT mode). On the left sidebar, click 'update firmware' (as most of the administration menu does not appear for Firefox users, downgrade to MS IE or another browser temporarily). Click the 'upload' button and indicate the location of the unzipped .BIN file. A box will appear with a progress indicator and a warning not to interrupt the upgrade. When the upgrade is completed, the SPA112/122 will reset and will likely take a minute or more to reinitialize, reconnect to the network and restore dial tone. SPA122 users who have installed the device in-line between the local PCs and the Internet will be disconnected from the Internet until reinitialization is complete.

Once the new firmware is deployed, call display will operate normally and the configuration web page will display in Firefox without missing options in the administration menu.

A manual for Cisco's SPA100 series adapters is online at http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/spa100-200/admin_guide_SPA100/spa100_ag.html

* You can check the most commonly used [[Cisco/Linksys Star Codes]]

=== SPA Star Codes ===

{| class="wikitable" style="margin-left: center; margin-right: auto; border: none;"
! colspan="3" | SPA Star Codes
|-
! scope="col" | Star Code
! scope="col" | Name
! scope="col" | Result
|-
| *69 || Call Return Code || This code calls the last caller.
|-
| *07 || Call Redial Code || Redials the last number called. (Not in pap2t)
|-
| *98 || Blind Transfer Code || Begins a blind transfer of the current call to the extension specified after the activation code.
|-
| *66 || Call Back Act Code || Starts a callback when the last outbound call is not busy.
|-
| *86 || Call Back Dea t Code || Cancels a callback.
|-
| *05 || Call Back Busy Act Code || Starts a callback when the last outbound call is busy. (Not in pap2t)
|-
| *72 || Cfwd All Act Code || Forwards all calls to the extension specified after the activation code.
|-
| *73 || Cfwd All Deact Code || Cancels call forwarding of all calls.
|-
| *90 || Cfwd Busy Act Code || Forwards busy calls to the extension specified after the activation code.
|-
| *91 || Cfwd Busy Deact Code || Cancels call forwarding of busy calls.
|-
| *92 || Cfwd No Ans Act Code || Forwards no-answer calls to the extension specified after the activation code.
|-
| *93 || Cfwd No Ans Deact Code || Cancels call forwarding of no-answer calls.
|-
| *63 || Cfwd Last Act Code || Forwards the last inbound or outbound calls to the extension specified after the activation code.
|-
| *83 || Cfwd Last Deact Code || Cancels call forwarding of the last inbound or outbound calls.
|-
| *60 || Block Last Act Code || Blocks the last inbound call.
|-
| *80 || Block Last Deact Code || Cancels blocking of the last inbound call.
|-
| *64 || Accept Last Act Code || Accepts the last outbound call. It lets the call ring through when do not disturb or call forwarding of all calls are enabled.
|-
| *84 || Accept Last Deact Code || Cancels the code to accept the last outbound call.
|-
| *56 || CW Act Code || Enables call waiting on all calls.
|-
| *57 || CW Deact Code || Disables call waiting on all calls.
|-
| *71 || CW Per Call Act Code || Enables call waiting for the next call.
|-
| *70 || CW Per Call Deact Code || Disables call waiting for the next call.
|-
| *67 || Block CID Act Code || Blocks caller ID on all outbound calls.
|-
| *68 || Block CID Deact Code || Removes caller ID blocking on all outbound calls.
|-
| *81 || Block CID Per Call Act Code || Blocks caller ID on the next outbound call.
|-
| *82 || Block CID Per Call Deact Code || Removes caller ID blocking on the next inbound call.
|-
| *77 || Block ANC Act Code || Blocks all anonymous calls.
|-
| *87 || Block ANC Deact Code || Removes blocking of all anonymous calls.
|-
| *78 || DND Act Code || Enables the do not disturb feature.
|-
| *79 || DND Deact Code || Disables the do not disturb feature.
|-
| *65 || CID Act Code || Enables caller ID generation.
|-
| *85 || CID Deact Code || Disables caller ID generation.
|-
| *25 || CWCID Act Code || Enables call waiting, caller ID generation.
|-
| *45 || CWCID Deact Code || Disables call waiting, caller ID generation.
|-
| *26 || Dist Ring Act Code || Enables the distinctive ringing feature.
|-
| *46 || Dist Ring Deact Code || Enables the distinctive ringing feature. The default is *46 ||.
|-
| *74 || Speed Dial Act Code || Assigns a speed dial number.
|-
| *16 || Secure All Call Act Code || Makes all outbound calls secure.
|-
| *17 || Secure No Call Act Code || Makes all outbound calls not secure.
|-
| *18 || Secure One Call Act Code || Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default.)
|-
| *19 || Secure One Call Deact Code || Secure One Call Deact Code Makes the next outbound call not secure. (It is redundant if all outbound calls are not secure by default.)
|}

[[category:Analog Telephone Adapters]]

Grandstream DP715/DP710

2019-05-14T16:55:27Z

Edward: /* Grandstream DP715 Configuration Detail */

[[File:Grandstream 715-710.png|400px|thumb|left|Grandstream DP715/DP710]]
<div style="width:100%;overflow:hidden;clear:both"></div>

== Grandstream DP715 Configuration Detail ==

Configure using Interactive Voice Prompt:

1. From the registered Handset, press *** to get into the IVR
menu. Enter option 02 to obtain the DP715 IP address.

2. Type the DP715 Base Station IP address in your PC browser e.g. http://192.168.2.1

3. Log in using password “admin” to configure the DP715 Base Station.

[[File:Grandstream7xxlogin.png|400px]]

Once that you login you will see the following window:

[[File:Granstream7xxstatus.png|400px]]

4. Click on '''PROFILE 1''' to configure your first line you will see the following window:

[[File:DP750correction.png|600px]]

Complete the followings fields.

* '''Primary SIP Server:''' One of our multiple servers (e.g. atlanta.voip.ms )
* '''Register Expiration:''' Set to 5 minutes (300 seconds)

Then go to the bottom and press "Update"

5. Click on '''HANDSETS''' to configure your Handsets, you will see the following window:

[[File:Granstream7xxHandsets.png|600px]]

Complete the followings fields.

* '''Enable Handset''' Set to Yes
* '''SIP User ID:''' Your VoIP.ms SIP username (e.g. 100000)
* '''Authenticate ID:''' Your VoIP.ms SIP username (e.g. 100000)
* '''Authenticate Password:''' ********* (account password)
* '''Name:''' Your Name

Then go bottom and press "Update"

===Setting Your Time Zone===

Click on the " Basic Settings " tab link at the top of the Grandstream Device Configuration window.

Scroll down to the bottom of the Basic Settings page to the Time Zone drop-down list.

From the drop-down menu list, select your local Time Zone.

Click on the Update button at the bottom of the configuration window to save your changes.

===Voice Mail Waiting Indicator Or MWI===

By default, the MWI indicator is disabled. In which case, the only way you know you have a voice mail is by picking up the handset and pressing the off-hook button. Normally you will just hear a dial tone. But, when you have voice mail waiting, you will hear a stutter tone.

To enable MWI indicator on the DP715 base unit:

*Login to the phone via web browser.
*Navigate to the Advanced Settings page.
*Scroll down to the very bottom of the Advanced Settings page.
*Change the MWI Blinking setting from Disable to Enable.
*Click the Update button. Then Reboot.

Once done, you need to associate a mailbox to the registered account from the VoIP.ms customer portal. For the main account it is done from Main Menu>>Account Settings>>General Tab and for a sub account it is chosen as the internal extension voicemail from Sub Accounts>>Manage Sub Accounts>>Edit. Next time you receive a voice mail, you will see the base unit's green LED (handset off-hook indicator) will blink. When the voicemail is deleted, the handset off-hook indicator will no longer blink.

File:DP750correction.png

2019-05-14T16:53:56Z

Edward: A correction to the NAT setting

A correction to the NAT setting

File:GrandstreanProfile7152.jpg

2019-05-14T16:52:35Z

Edward: uploaded a new version of "File:GrandstreanProfile7152.jpg"

File:GrandstreanProfile7152.jpg

2019-05-14T16:49:18Z

Edward: uploaded a new version of "File:GrandstreanProfile7152.jpg": Correction to the NAT option, was showing to no

Account Settings

2019-05-10T14:33:50Z

Edward: /* Security */

The Account Settings page is one of the most important pages in the Customer Portal, it will help you out as its name points out, to set all the proper ways to use your account, and these settings will affect the global settings on your account, from Device type to routing and even security of your account.

__TOC__

== Account Routing ==

These settings define the routes the system will use when you place calls to either Canada, Toll Free or International Numbers.

[[File:Accsettings1.png|thumb|none|600px|Account Routing]]

'''Canada Routing''': You can choose between the [[Value vs Premium|'''Value''' or '''Premium''']] Route. If you are looking for the best wholesale price without any volume commitment you can use the '''Value''' route, however the CallerID number is not always guaranteed to work in this route.
The '''Premium''' route has a better level of quality, at a price that is a little higher than the value option. The CallerID Number is guaranteed to work for this route. Please note that the '''Canada Route''' can also be set independently per sub account, under the [[Sub Accounts]] settings.

Please note the US route is only available on the Premium route with a rate of $0.009 per minute.

'''International Routing''': You can choose also the [[Value vs Premium|'''Value''' or '''Premium''']] route. Both routes are reliable, however you're going to find better quality if you're using the '''Premium''' route. The CallerID Number is not guaranteed to work for international calls even if you use the '''Premium''' route.

'''Toll Free Routing''': This define the route the system will use when you place a Toll-Free call to Canada or USA. You can choose between the '''Value''' route (free to use) or the '''Premium''' (at 1.06 cent per minute). As the other settings, the '''Premium''' route give you better quality and the CallerID Number is guaranteed to work.

'''Toll Free Carrier''': This setting defines the type of carrier that will be used when you dial a Toll-Free number. You can choose between '''Server Default''', '''American Carrier''' or '''Canadian Carrier'''.

'''Server Default:''' The Carrier for the termination of your toll-free call will be chosen depending on the geographical location of the server
you are registering or connecting to. American Servers will dial through an American Carrier and Canadian servers will use a Canadian Toll-Free carrier.
'''American Carrier:''' Your Toll-Free calls will always go through an American Carrier, even if you use a Canadian server.
'''Canadian Carrier:''' Your Toll-Free calls will always go through a Canadian Carrier, even if you use an American server.

For example, this feature can be useful for west based Canadian customers who wish to use the Seattle server but have their toll-free termination calls go through a Canadian carrier.

'''Note''': Please refer to [[Service Cost]] for information about the rates of the outgoing calls.

== Account Restrictions ==

These settings define the restrictions the system will use when you place calls to either USA, Canada or International Numbers.

[[File:Account Restrictions.jpg|thumb|none|600px|Account Restrictions]]

'''Allow 411 dialing''': When enabled, you can call 411 directory assistance at the cost of $0.99 per call.

'''Note''': XXX-555-1212 is disabled within VoIP.ms. The only way to reach directory assistance is by dialing 411.

'''Allow International Calls''': When Disabled, calls placed to destinations outside USA/Canada will be automatically rejected. If you are using [[Sub Accounts]] you need to setup this directly in the sub account.

'''Note''': This option is set to "No" by default for New accounts.

'''Max. Call Time for US48/Canadian Calls''': With these setting you can choose how long the call will last when you call to US or Canada Numbers. If a call to a US or Canada Number exceeds the maximum time you have set in this option, it will be automatically hung up.

'''Max. Call Time for International Calls''': Works exactly as the '''Max. Call Time for US48/Canadian Calls''' setting. If a call to an International Number exceeds the maximum time set, it will be automatically hung up.

'''International Amount Restriction''': With this option you can set the maximum amount per minute for calls to international destinations. If the Per minute rate for an International destination exceeds the value you have set in this option, the call will not be connected.

For example, let say you can set this option to $0.150 and you want to call a UK mobile number with area code 4470, the call will not go through because the per minute rate for this area code is $0.3048.

'''Note:''' This settings depends on the International Routing you have chosen between Value or Premium.

'''Allow Calls to Countries''': Here you can select Regions or Specific Countries to allow outgoing calls from your account.

In order to allow calls to a specific country you can click the name of the region to expand the countries within it and select the desired countries only. You can select '''Select All''' so all the countries within the continent will be selected. You will also be able to search for a specific country using the '''Search Country''' field. If you call to a country not allowed in this section, the call will not be connected.

Please make sure to only allow those destinations that you are planning to dial and avoid having unplanned destinations allowed.

[[File:Allow Inter Count.jpg|thumb|none|600px|Allow Calls to Countries]]

'''Note''': If you have international calls disabled, it will prevent calling to the international countries you have enabled in this list.

== General ==

These are the general settings used by the system when you make or receive calls.

[[File:AcctSet Gen.jpg|thumb|none|600px|General]]

'''e911 Default CallerID''': This enables you to use any CallerID you want to with your accounts but when you call E911 the number you specify here is used as your CallerID and the call is placed and completed successfully.

'''Dialing Mode''': This setting allows you to set the way you're going to dial other numbers.

*North America (Recommended): You can dial to countries part of the North American Numbering Plan Administration by dialing 10 or 11 digits. (with or without the 1 prefix). You'll need to use the 00 or 011 prefix to call international numbers.

*E164: You dial numbers by using Country Code first. You don't need to send the 00 or 011 prefix when using this mode, but it's still supported.

'''CallerID Number''': You can set the [[Caller ID|callerID Number]] you want to pass if you are using an [[Devices|ATA]], IP Phone or [[Softphones|Softphone]]. It is important to pass a valid [[Caller ID|caller id]] to ensure proper termination. If you have a a device capable of passing its own [[Caller ID|CallerID number]] such as a soft switch or [[PBXs|PBX]], you can leave this blank to set the [[Caller ID|CallerID Number]] on your side.

'''Voicemail Associated to the Main Account''': You can select the mailbox associated to the main account.

*Accessing Voice Mail: When you dial *97 from your main account, it will not prompt for the [[voicemail]] ID. It will also not prompt for the password if you have selected "Skip Password" option in the [[voicemail]] configuration.

*Message Waiting Indicator: When there are new messages, notice of new message(s) will be sent. This will lead to different results depending on your type of adapter, soft phone or IP phone. For example, when using a Linksys [[Devices|ATA]] adapter, it's usually a stutter tone with periodic ring, IP phones usually use a stutter dial tone when you pick up the line and are equipped with a blinking light, soft phones usually show up a Voicemail Icon in the dial display.

'''Note:'''This setting only affects the device registered with the main account.
If you're using a [[Sub Accounts|sub account]] you need to set the '''Internal Extension Voicemail''' for that subaccount.

'''Music On Hold''': Most IP Phones and [[Softphones|Softphones]] inform the VoIP server they are connected to when the HOLD Button is pressed. If you would like to present music to the person you place on hold, you can select it here. 
The following options for music are available: 

*'''No Music, Intermittent Bleep:''' Silent, but with a subtle intermittent bleep sound to let your callers know that they are still on the line while they hold.
*'''Away in the Tropics:''' From Hawaii to the Caribbean, these tracks deliver sounds of ukulele, steel drums, and steel guitars.
*'''Coffee and Sunrise:''' Uplifting without being perky, and positive without being too smiley.
*'''Coffee Shop Acoustic:''' Soothing, acoustic guitar tracks makes for a relaxed feel and sets a great atmosphere.
*'''Easy Listening:''' Smooth, casual tunes.
*'''Guitar Alchemy:''' Clever harmonics and progressive chord sequences to create a joyful and warming musical experience.
*'''Happy Endings:''' Uplifting, commercial style. Guitar, drums, ukulele, harmonica and bells.
*'''Light and Casual:''' Soothing and peaceful songs with a light, positive feeling.
*'''Orchestral Moods:''' Emotional and dramatic tales spun by violins, pianos and full orchestras.
*'''Piano Mix:''' Smooth Piano
*'''Rock Me Easy:''' Feel-good music to create a relaxing atmosphere.
*'''Spa Sounds:''' Soft, slow and serene instrumentals.

You can test the categories by dialing the following codes: 

'''*** 89''' Away in the Tropics 
'''*** 90''' Coffee and Sunrise 
'''*** 91''' Coffee Shop Acoustic 
'''*** 92''' Easy Listening 
'''*** 93''' Guitar Alchemy 
'''*** 94''' Happy Endings 
'''*** 95''' Light and Casual 
'''*** 96''' Orchestral Moods 
'''*** 97''' Piano Mix 
'''*** 98''' Rock Me Easy 
'''*** 99''' Spa Sounds 
'''***100''' No Music, Intermittent Bleep 

== Security ==

Here you can change the password to access your Customer Portal as well your SIP/IAX passwords for your main account.

[[File:Accset sec.jpg|thumb|none|600px|Security]]

'''Customer Portal Password''': Here you can change the password used to log in your VoIP.ms Customer Portal.

'''Main SIP/IAX Password''': Here is where you can change the password used to register your Main Account to one of the VoIP.ms servers.

'''Note:''' For default, the Main SIP/IAX Password is the same you use to login in your Customer Portal.

== Inbound Settings==

In this tab, you can set the protocol used for inbound calls and the device you're using for your main account.

[[File:Inbound.jpg|thumb|none|600px]]

'''Protocol for inbound DIDs''': Here you can select either SIP or IAX2 protocol. This will depend on the device you want to use with your Main account. SIP is the recommended protocol.

'''Device type''': Make sure to select the correct type of device you're going to use with the main account to properly receive incoming calls in your device.

== Notifications ==

The notifications tab holds a complete list of alerts based upon most of the features that you will be able to see in the customer portal, in regards of changes over your configuration system structure, e911 dialing and API configuration changes.

You will be able to select rather to receive an email or not when these events occur within your configuration, added to the always useful Balance Threshold.

[[File:Accsettings2.png|thumb|none|600px|Notifications]]

'''Balance Threshold''': Here you can set an amount and when your balance goes below this amount you will receive an email to the address set in the option below. You can select a value between $1 up to $200, or if you like you can disable the notifications (this is not recommended).

'''Note''': Set this according to the monthly use you have in your account.

'''Email''': Enter the email address in which you want to receive the notifications. Make sure to enter a proper email address. If you want to receive notifications in more than one email address, you can list the emails separated with commas (,). This will only work for the Balance Threshold

'''Note''': While the balance is below your threshold, you're going to keep receiving emails until you add new funds in your account or you change the balance threshold.

== Default DID Routing ==

These settings let you define the preconfigured Routing options that are going to be applied to each new DIDs that you order. However, you are still able to change the settings for the DID in the order page.

[[File:DefaultDIDRouting.png|thumb|none|600px|Default DID routing]]

'''Select Plan''': Incoming Calls and Monthly Fee for the DID will be charged according to the Plan selected here. Plan prices change depending on the rate center. These prices will be shown when purchasing numbers.

'''CallerID Name Lookup''': When activated, the system will perform a lookup in the LIBD/CNAM database to find the name matching the number of callers with a US/Canadian CallerID Number. The result of this query will be displayed with the following format "Caller ID name portion" <Caller ID number>.

'''DID POP''': Here you can set the server in which your software/device is registered to or receiving the call from.
'''Note''': Always make sure to register your software/device to the same server you set in this option to properly receive incoming calls.

[[File:DefaultRouting.png|thumb|none|600px|Default DID POP]]

'''Routing''': Here you can select the destination of the incoming calls to your DID number. You can routed directly to one account/sub account, an [[Digital Receptionist (IVR)]], [[Calling Queues]], [[Time Conditions]], [[Ring Groups]], etc.

'''Failover Options''': This setting let you define where to redirect the call when the destination is '''Busy''', '''Unreachable''' or '''No Answer'''. These options can be displayed by clicking the '''Show Failover Options''' button

Note: You can change these settings later or enable additional settings in your Customer Portal>>DID Numbers>>Manage DID(s)

== Newsletter ==

'''Newsletter Subscription''': You can set if you want to receive news from voip.ms

[[File:Accsettings3.png|thumb|none|600px|Newsletter]]

== Advanced ==

These options allow you to set the allowed codecs, DTMF modes and NAT options. These settings are recommended for advanced users only.

[[File:Accset adv.jpg|thumb|none|600px|Advanced]]

'''NAT (Network Address Translation)''': This setting should be set to Yes if you're behind a NAT, if not set to No. If you're unsure what setting means, is highly recommended that you leave it to Yes.

'''DTMF Mode''': This allows you to select the DTMF mode that is going to be used for your account. If you set this to AUTO the RFC2833 (AVT) is going to be used and automatically switch to INBAND if the other end doesn't support RFC2833.
Note: Its recommended that you select the same DTMF mode in your device.

'''Allowed Codecs''': This setting allows you to select the codecs that you're going to use, you can choose to allow one or more codecs at the time.
Note: It's recommended that you select Allow All and only change it if you have an specific reason to do so.

'''Encrypted SIP Traffic''': This setting allows you to encrypt the communication between your device and our server, by using the SIP-TLS ''(Transport Layer Security)'' and SRTP ''(Secure Real-Time Transport Protocol)'' protocol.

Before using this feature, please consult the article about the [[Call Encryption - TLS/SRTP]]

'''Note:''' If enabled, all the SIP traffic will be encrypted for the main account.
''' ''' Please note that if encrypted calls are enabled then you need to configure your device to make and receive encrypted calls.

[[category:guides]]

Call Encryption - TLS/SRTP

2019-04-02T19:07:37Z

Edward: /* Configuration on SIP Client */

This feature allows you to encrypt the communication between your device and our server, by using the SIP-TLS ''(Transport Layer Security)'' and SRTP ''(Secure Real-Time Transport Protocol)'' protocol.

This adds a security layer when the packets are being transmitted between you and our server, it encapsulates and encrypt the transmission. In other words, when your device is configured with this encryption method, your device asks to our server a dedicated certificate to establish a trust and fully secure communication from each part.

This is ideal if you are using a softphone on a public network. ''(We strongly recommend you to use this function in this case.)''

Once encrypted calls are enabled for your account or sub account, the SIP-TLS and SRTP must be used.
Your account or sub account will no longer be able to use regular SIP communication method.

__TOC__

== Activate This Option on Your Main Account ==

1) To active this feature go to your Customer portal home and click on “'''Main Menu'''” > “'''Account Settings'''”

:[[File:TLS-SRTP-Steps0001.png|border]]

2) Once you are in the “'''Account Settings'''” section, navigate through the submenu and go to “'''Advanced'''” and find the field “'''Encrypted SIP Traffic'''”, set to '''Yes''' and press '''Apply'''

:[[File:TLS-SRTP-Steps0002.png|border]]

'''Note:''' If enabled, all the SIP traffic will be encrypted for the main account.
''' ''' Please note that if encrypted calls are enabled then you need to configure your device to make and receive encrypted calls.

== Activate This Option on Your Sub Account ==

1) You may also activate this feature on a sub account. You need to navigate through the navigation bar and select “'''Sub Accounts'''” > “'''Manage Sub accounts'''”.

:''If you don't have a sub account yet, you can create one by clicking on the tab “'''Create Sub Account'''”''

:[[File:TLS-SRTP-Steps0003.png|border]]

2) In the column “'''Actions'''” click on the edit button [[File:edit_icon.png]], for the row of the sub account you need to activate this feature.

:[[File:TLS-SRTP-Steps0004.png|border]]

3) Once you are in the Edit screen, find the “'''Advanced Options'''” and “'''Click here to display'''”. Then set “'''Encrypted SIP Traffic'''” to '''Yes''', and press '''Update Account'''

:[[File:TLS-SRTP-Steps0005.png|border]]

'''Note:''' If enabled, all SIP traffic calls will be encrypted for this sub account.
''' ''' Please note that if encrypted calls are enabled then you need to configure your device to make and receive encrypted calls.

== Configuration on SIP Client ==

Once you have activated the feature on your main account/sub account, you need to configure your SIP client.

On some devices, you will have to configure some settings to '''enable''' the SIP-TLS communication method.
In your settings you must select '''TLS''' as your transport protocol and activate ''media encryption or SRTP*'' as '''Mandatory*'''. Without mandatory ''media encryption*'', this would result in a call rejection with the SIP error 488.

'''''Some technical considerations that you need to know for using this feature.'''''
Please take note, when using encrypted calls with a server, you must always use the server name with a '''number''' at the end.
For example, you must use chicago'''1'''.voip.ms instead of chicago.voip.ms.

This also applies to cities with only one server.
For example, you select the POP server london.voip.ms in the portal, but write london'''1'''.voip.ms when you configure your [[Devices]], [[Softphones]] or [[PBXs]].

When you are using the TLS protocol, it is implied to be using TCP as packet transport. The reason is using TLS over UDP is not supported by the TLS specification.
The TLS by TCP will use the port '''5061''' instead of 5060. We also have an alternative port such as '''5081''' and '''42873'''

''*The configuration and the terminology may vary from each device/PBX.''

''*Take note; some SIP clients do not support the call encryption, in some cases is a paid feature, or is available only in the paid version.''

Some old systems/devices require to configure the certificate manually. In case is needed, please contact technical support via live chat or email to support@voip.ms (from the email address associated to your account), requesting the client certificate and advising the SIP server.

'''NOTE''': The certificate expires every 90 days, so requesting a new one every period will be necessary to keep call encryption working under these circumstances

To know more about that, please refer to the device configuration page or your device manual.

== TLS/SRTP Registration Status Validation ==

When your device is fully registered by using SIP-TLS protocol, you will be able to see the registration status in the portal Home page “'''Main menu'''” > "'''Portal Home'''" for each account/sub account registered or in “'''Sub Accounts'''” > “'''Manage Sub accounts'''” tab to see all of your Sub Accounts registration status.

A green padlock [[File:green_padlock.png]] will appears on the right of "Registered" in green. If you don’t see the padlock, you need to revalidate some configuration.

:[[File:TLS-SRTP-Steps0006.png|border]]

:[[File:TLS-SRTP-Steps0007.png|border]]

== Notes ==

*It has been confirmed that ATA Obi100 does not support encrypted calls.

Call Encryption - TLS/SRTP

2019-04-02T19:06:22Z

Edward: /* Configuration on SIP Client */

This feature allows you to encrypt the communication between your device and our server, by using the SIP-TLS ''(Transport Layer Security)'' and SRTP ''(Secure Real-Time Transport Protocol)'' protocol.

This adds a security layer when the packets are being transmitted between you and our server, it encapsulates and encrypt the transmission. In other words, when your device is configured with this encryption method, your device asks to our server a dedicated certificate to establish a trust and fully secure communication from each part.

This is ideal if you are using a softphone on a public network. ''(We strongly recommend you to use this function in this case.)''

Once encrypted calls are enabled for your account or sub account, the SIP-TLS and SRTP must be used.
Your account or sub account will no longer be able to use regular SIP communication method.

__TOC__

== Activate This Option on Your Main Account ==

1) To active this feature go to your Customer portal home and click on “'''Main Menu'''” > “'''Account Settings'''”

:[[File:TLS-SRTP-Steps0001.png|border]]

2) Once you are in the “'''Account Settings'''” section, navigate through the submenu and go to “'''Advanced'''” and find the field “'''Encrypted SIP Traffic'''”, set to '''Yes''' and press '''Apply'''

:[[File:TLS-SRTP-Steps0002.png|border]]

'''Note:''' If enabled, all the SIP traffic will be encrypted for the main account.
''' ''' Please note that if encrypted calls are enabled then you need to configure your device to make and receive encrypted calls.

== Activate This Option on Your Sub Account ==

1) You may also activate this feature on a sub account. You need to navigate through the navigation bar and select “'''Sub Accounts'''” > “'''Manage Sub accounts'''”.

:''If you don't have a sub account yet, you can create one by clicking on the tab “'''Create Sub Account'''”''

:[[File:TLS-SRTP-Steps0003.png|border]]

2) In the column “'''Actions'''” click on the edit button [[File:edit_icon.png]], for the row of the sub account you need to activate this feature.

:[[File:TLS-SRTP-Steps0004.png|border]]

3) Once you are in the Edit screen, find the “'''Advanced Options'''” and “'''Click here to display'''”. Then set “'''Encrypted SIP Traffic'''” to '''Yes''', and press '''Update Account'''

:[[File:TLS-SRTP-Steps0005.png|border]]

'''Note:''' If enabled, all SIP traffic calls will be encrypted for this sub account.
''' ''' Please note that if encrypted calls are enabled then you need to configure your device to make and receive encrypted calls.

== Configuration on SIP Client ==

Once you have activated the feature on your main account/sub account, you need to configure your SIP client.

On some devices, you will have to configure some settings to '''enable''' the SIP-TLS communication method.
In your settings you must select '''TLS''' as your transport protocol and activate ''media encryption or SRTP*'' as '''Mandatory*'''. Without mandatory ''media encryption*'', this would result in a call rejection with the SIP error 488.

'''''Some technical considerations that you need to know for using this feature.'''''
Please take note, when using encrypted calls with a server, you must always use the server name with a '''number''' at the end.
For example, you must use chicago'''1'''.voip.ms instead of chicago.voip.ms.

This also applies to cities with only one server.
For example, you select the POP server london.voip.ms in the portal, but write london'''1'''.voip.ms when you configure your [[Devices]], [[Softphones]] or [[PBXs]].

When you are using the TLS protocol, it is implied to be using TCP as packet transport. The reason is using TLS over UDP is not supported by the TLS specification.
The TLS by TCP will use the port '''5061''' instead of 5060. We also have an alternative port such as '''5081''' and '''42873'''

''*The configuration and the terminology may vary from each device/PBX.''

''*Take note; some SIP clients do not support the call encryption, in some cases is a paid feature, or is available only in the paid version.''

Some old systems/devices require to configure the certificate manually. In case is needed, please contact technical support via live chat or email to support@voip.ms (from the email address associated to your account), requesting the client certificate and advising the SIP server. '''NOTE''': The certificate expires every 90 days, so requesting a new one every period will be necessary to keep call encryption working under these circumstances

To know more about that, please refer to the device configuration page or your device manual.

== TLS/SRTP Registration Status Validation ==

When your device is fully registered by using SIP-TLS protocol, you will be able to see the registration status in the portal Home page “'''Main menu'''” > "'''Portal Home'''" for each account/sub account registered or in “'''Sub Accounts'''” > “'''Manage Sub accounts'''” tab to see all of your Sub Accounts registration status.

A green padlock [[File:green_padlock.png]] will appears on the right of "Registered" in green. If you don’t see the padlock, you need to revalidate some configuration.

:[[File:TLS-SRTP-Steps0006.png|border]]

:[[File:TLS-SRTP-Steps0007.png|border]]

== Notes ==

*It has been confirmed that ATA Obi100 does not support encrypted calls.

File:CallRecordingsPage.png

2019-03-20T16:23:03Z

Edward:

File:RecordingIconSubs.png

2019-03-20T16:02:30Z

Edward: uploaded a new version of "File:RecordingIconSubs.png"

File:IconPortalHome.png

2019-03-20T15:52:41Z

Edward: uploaded a new version of "File:IconPortalHome.png"

File:IconPortalHome.png

2019-03-20T15:50:01Z

Edward:

Call Encryption - TLS/SRTP

2019-03-15T17:48:21Z

Edward:

This feature allows you to encrypt the communication between your device and our server, by using the SIP-TLS ''(Transport Layer Security)'' and SRTP ''(Secure Real-Time Transport Protocol)'' protocol.

This adds a security layer when the packet being transmitted between you and our server, it encapsulates and crypt the transmission. In other words, when your device is configured with this encryption method, your device asks to our server a dedicated certificate to establish a trust and fully secure communication from each part.

This is ideal if you are using a softphone on a public network. ''(We strongly recommend you to use this function in this case.)''

Once encrypted calls are enabled for your account or sub account, the SIP-TLS and SRTP must be used.
Your account or sub account will no longer be able to use regular SIP communication method.

__TOC__

== Activate This Option on Your Main Account ==

1) To active this feature go to your main account by navigating through the “'''Main Menu'''” > “'''Account Settings'''”

:[[File:TLS-SRTP-Steps0001.png|border]]

2) Once you are in the “'''Account Settings'''” section, navigate through the submenu and go to “'''Advanced'''” and find the field “'''Encrypted SIP Traffic'''” to '''Yes''', and press '''Apply'''

:[[File:TLS-SRTP-Steps0002.png|border]]

'''Note:''' If enabled, all the SIP traffic will be encrypted for the main account.
''' ''' Please note that if encrypted calls are enabled then you need to configure your device to make and receive encrypted calls.

== Activate This Option on Your Sub Account ==

1) You may also activate this feature on a sub account. You need to navigate through the navigation bar and select “'''Sub Accounts'''” > “'''Manage Sub accounts'''”.

:''If you don't have a sub account yet, you can create one by clicking on the tab “'''Create Sub Account'''”''

:[[File:TLS-SRTP-Steps0003.png|border]]

2) In the column “'''Actions'''” click on the edit button [[File:edit_icon.png]], in a row of the sub account you need to activate this feature.

:[[File:TLS-SRTP-Steps0004.png|border]]

3) Once you choose, your sub account or have created a new one. Find the “'''Advanced Options'''” and “'''Click here to display'''”. Then “'''Encrypted SIP Traffic'''” to '''Yes''', and press '''Update Account'''

:[[File:TLS-SRTP-Steps0005.png|border]]

'''Note:''' If enabled, all SIP traffic calls will be encrypted for this sub account.
''' ''' Please note that if encrypted calls are enabled then you need to configure your device to make and receive encrypted calls.

== Configuration on SIP Client ==

Now you have activated the feature on your main account/sub account you need to configure your SIP client.

On some device, you will have to configure some settings to '''enable''' the TLS-SIP communication method.
In your settings you must select '''TLS''' as your transport protocol and activate ''media encryption or SRTP*'' as '''Mandatory'''. Without mandatory ''media encryption'', this would result in a call rejection with the SIP error 488.

'''''Some technical precisions that you need to know by using this feature.'''''
Please take note, when using encrypted calls with a server, you must always use the server name with a '''number''' at the end.
For example, you must use chicago'''1'''.voip.ms instead of chicago.voip.ms.

This also applies to cities with only one server.
For example, you select london.voip.ms in the portal, but write london'''1'''.voip.ms when you configure your device, softphone or PBX.

When you are using the TLS protocol, it is implied to using TCP as packet transport. The reason is using TLS over UDP is not supported by the TLS specification.
The TLS by TCP will use the port '''5061''' instead of 5060. We also have an alternative port such as '''5081''' and 42873

''*The configuration and the terminology may vary from each device/PBX.''

''*Take note; some SIP clients do not support the call encryption, in some cases is a paid feature, or is available only in the paid version.''

To know more about that, please refer to the device configuration page or your device manual.

== TLS/SRTP Registration state validation ==

When your device is fully registered by using SIP-TLS protocol, you will be able to see the registration status in the portal Home page “'''Main menu'''” > "'''Portal Home'''" for each account/sub account registered or in “'''Sub Accounts'''” > “'''Manage Sub accounts'''” tab to see all of your Sub Accounts registration status.

A green padlock [[File:green_padlock.png]] will appears on the right of registered in green. If you don’t see the padlock, you need to revalidate some configuration.

:[[File:TLS-SRTP-Steps0006.png|border]]

:[[File:TLS-SRTP-Steps0007.png|border]]

File:Actions column.png

2019-03-15T15:20:03Z

Edward:

File:RecordingsCDRwIndictations.png

2019-03-15T14:25:38Z

Edward: uploaded a new version of "File:RecordingsCDRwIndictations.png": Updated to show recent changes

File:RecordingsCDRwIndictations.png

2019-02-12T20:43:15Z

Edward:

File:DeleteRecording.png

2019-02-11T20:38:27Z

Edward:

File:RecordingIconSubs.png

2019-02-11T17:26:20Z

Edward: uploaded a new version of "File:RecordingIconSubs.png"

File:RecordingIconSubs.png

2019-02-11T17:24:45Z

Edward:

File:RecordingMain.png

2019-02-11T17:11:18Z

Edward:

File:RecordingSubs.png

2019-02-11T17:01:23Z

Edward:

File:EnablediconDID.png

2019-02-08T21:50:07Z

Edward:

File:RecordingDIDs.png

2019-02-08T20:48:21Z

Edward: Recordings for DID

Recordings for DID

File:12exportFR.png

2019-01-22T18:45:32Z

Edward:

File:11Order groupFR.png

2019-01-22T18:44:39Z

Edward:

File:10Search familyFR.png

2019-01-22T18:43:56Z

Edward:

File:9Edit entryFRpng.png

2019-01-22T18:43:30Z

Edward:

File:8Edit group2FR.png

2019-01-22T18:43:05Z

Edward:

File:7Edit groupFR.png

2019-01-22T18:42:42Z

Edward:

File:6Family groupFR.png

2019-01-22T18:42:21Z

Edward:

File:5Add groupFR.png

2019-01-22T18:41:55Z

Edward:

File:4Manage groupsFR.png

2019-01-22T18:41:20Z

Edward:

File:3Phonebook entry with SIPUriFR.png

2019-01-22T18:40:57Z

Edward:

File:2Phonebook entryFR.png

2019-01-22T18:40:32Z

Edward:

File:1Phonebook main screenFR.png

2019-01-22T18:40:09Z

Edward:

Phone book

2019-01-22T17:31:41Z

Edward: /* Setup a 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.

Additionally you can have a [[Caller ID]] name (CNAM) override to identify the calls of an important customer if his number doesn't have a proper [[Caller ID]] name (CNAM) linked to it.

You can also use the Phone Book entries with our Virtual Fax and SMS features. When sending from both features, the field will instantly show you the available entries to choose from when you start inputting either a number or a name from the Phone Book. You will also see the name configured in the Phone Book for received messages.

This guide will help you to configure and learn how to use properly the Phone Book.

'''NOTE: You can have up to a maximum of 500 Phone Book entries and assign Speed Dial codes to 99 of those entries.'''

__TOC__

== Phone Book Purposes ==

Before creating the Phone Book entry, we are going to explain the main purposes of this feature:

=== Speed-Dial ===

You can create a Phone Book entry to work as a Speed-Dial, allowing you to place a call by pressing a reduced number of keys. This function is particularly useful if you dial certain numbers on a regular basis. You can program a speed dial to local, long distance or international numbers, you can also have a [[SIP URI]] (like *7501 to dial john@other-sip-provider.com).

The prefix to dial your speed dial entries is *75. Example: If you want to dial entry 01, you need to dial *7501 from your phone. It's not currently possible to use a different prefix. You may have up to 99 Speed-Dial entries.

=== Group your contacts together ===

Create and manage different groups to have your contacts listed in an ordered fashion and make your search through your phone book contacts more efficient. You can create now different groups for "family", "friends", "office", etc and have your lists of contacts ordered by groups.

=== CallerID-name (CNAM) Override ===

When you receive an incoming call to one of your numbers that matches a phone number in the Phone Book, the CNAM of the incoming call can be set in the entry on the Phone Book. For example, let´s say that you have an entry with the number 5554443322 associated with the name "John Smith", whenever you receive a phone call for that number you will see the CNAM as "John Smith". The Phone Book feature overrides the CNAM look up feature (if you have enabled it for your DID number).

For the name display from Phone Book to operate correctly, the displayed number must be written as it appears in the [[Call Detail Records]] - for North American numbers this will usually be ten digits without the leading +1.

=== CallerID Number Override ===

When you call one number using the Speed-Dial if you have setup a "CallerID Number Override" the default [[Caller ID|caller ID number]] will be changed for the one you have configured in your phone book entry.

=== Virtual Fax and SMS ===

You can also use the Phone Book entries with our Virtual Fax and SMS features. When sending from both features, the field will instantly show you the available entries to choose from when you start inputting either a number or a name from the Phone Book. You will also see the name configured in the Phone Book for received messages.

== Setup a Phone Book ==

First go to your Customer Portal and follow the menu option "DID Numbers" >> "Phone Book"

[[File:Phonebook mainScreen.png|thumb|none|800px]]

From this screen you can create, edit and delete Phone Book entries. Your Phone Book may contain a total of 500 entries.

=== Create a Phone Book Entry with a Phone Number ===

To create an entry, first click on the button "Add Phone Number", you will be prompted to the next screen:

[[File:Phonebook entry.png|thumb|none|800px]]

'''Speed Dial''': Here you only need to select which Speed-Dial you want to assign to this number. ''Optional Field.''

'''Name''': You can enter here the name that you want to use as CNAM Override. ''Mandatory field.''

'''Phone Number''': Enter here the phone number. ''Mandatory field.''

You can also use an International Number, you only need to make sure to use the prefix 00 or 011 to make the call to the number. Make sure that you have enabled the International Calls in your account.

'''Note''': The CallerID Number is not 100% guaranteed to be passed properly for International Routes at the moment.

'''CallerID Number Override''': Enter here the number you want to pass. This overrides the default callerID Number. ''Optional field.''

'''Group''': Select one of your phone book groups. You can also create a new Phone book group and the contact will be part of the group you are creating from the new entry directly.

'''Note''': The '''"General"''' group is the group existing by default. Any phone book entry created not assigned to a customized group, is assigned to the General group.

'''Note''': Here you can set a note to identify the entry. ''Optional field.''

=== Create a Phone Book Entry with a SIP URI ===

Creating a Phone Book Entry for a [[SIP URI]] is basically the same as creating the entry for a Phone Number. The difference is that you can select or create the [[SIP URI]] entry in this point in order to assign a Speed-Dial code . The information is selected for the [[SIP URI]] feature within your Customer Portal.

[[File:Phonebook SIPUri.png|thumb|none|800px]]

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

== Manage your groups ==

=== Create a Phone Book group ===

To create a new Phone book group, click on "Manage groups".

[[File:Manage group.png|thumb|none|600px]]

Click on "Add group" to create a new group.

[[File:Add group.png|thumb|none|600px]]

You will now be prompted a new window that allows you to set the Name of the group (we recommend using an easy to remember name like "Family").

Then click on '''Available members''' to see all your Phone book entries available to that moment. You can also see some basic information of the available members, including if they are part of another existing group.

Check the boxes of all the members you want to include in this group, and apply by clicking on the '''Save Phone Group''' option

[[File:Familygroup.png|thumb|none|600px]]

=== Edit your groups ===

If you have other groups you want to manage, you can click on the "Edit" icon (pen and paper icon), under the "Actions" column.

In "Group members" you will see all the members that are already part of the group you editing, and can remove them by clicking the Red X icon.

[[File:Edit group1.png|thumb|none|800px]]

In "Available members" you will see all the phone book contacts that are part of the general phone book or part of other groups, allowing you to change group they currently are for the one you are editing.

[[File:Edit group2.png|thumb|none|800px]]

=== Change the group of a contact ===

At any point, you can change the group of individual phone book entries. This is useful in case you change your mind for the group of a contact or if you created group after the contact.

From the main Phone Book screen, select the '''Edit''' icon (pen and paper icon under "Actions" column), then click on the '''Group''' option to see your list of available groups, select the one you desire and '''Save.'''

[[File:Edit contact2.png|thumb|none|800px]]

=== Search or order your contacts based on the group ===

The Phone book group feature introduces a new search criteria when going through your Phone book entries. You can type the name of a group in the search bar, and the customer portal will display only the Phone book entries that match the name of the group.

[[File:Search group.png|thumb|none|800px]]

To order the Phone book entries by groups, you simply click once on the "Group" column, and the portal will re-order your list of contacts based on the group name in alphabetic order.

[[File:Order groups.png|thumb|none|800px]]

== Import or Export the Phone Book ==

[[File:Phonebook export importpng.png|thumb|none|800px]]

Additionally you can import or export your Phone Book, the format is CSV (Comma-separated values). Here's an example of what the information should look like when you export your Phone Book.

*7501,"John Smith",5554443322
*7502,"John Doe",johndoe@other-provider.com

'''Note''': If you mark the box '''Overwrite Existing Phone Book Entries''' while Importing your Phone
Book, then all your existing Phone Book entries with a Speed Dial will be updated only with a valid match
for the same Speed Dial in your uploaded file.

[[category:guides]]

File:Phonebook mainScreen.png

2019-01-22T17:30:31Z

Edward:

Phone book

2019-01-21T20:25:05Z

Edward: Updated to reflect the addition of the Group feature

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.

Additionally you can have a [[Caller ID]] name (CNAM) override to identify the calls of an important customer if his number doesn't have a proper [[Caller ID]] name (CNAM) linked to it.

You can also use the Phone Book entries with our Virtual Fax and SMS features. When sending from both features, the field will instantly show you the available entries to choose from when you start inputting either a number or a name from the Phone Book. You will also see the name configured in the Phone Book for received messages.

This guide will help you to configure and learn how to use properly the Phone Book.

'''NOTE: You can have up to a maximum of 500 Phone Book entries and assign Speed Dial codes to 99 of those entries.'''

__TOC__

== Phone Book Purposes ==

Before creating the Phone Book entry, we are going to explain the main purposes of this feature:

=== Speed-Dial ===

You can create a Phone Book entry to work as a Speed-Dial, allowing you to place a call by pressing a reduced number of keys. This function is particularly useful if you dial certain numbers on a regular basis. You can program a speed dial to local, long distance or international numbers, you can also have a [[SIP URI]] (like *7501 to dial john@other-sip-provider.com).

The prefix to dial your speed dial entries is *75. Example: If you want to dial entry 01, you need to dial *7501 from your phone. It's not currently possible to use a different prefix. You may have up to 99 Speed-Dial entries.

=== Group your contacts together ===

Create and manage different groups to have your contacts listed in an ordered fashion and make your search through your phone book contacts more efficient. You can create now different groups for "family", "friends", "office", etc and have your lists of contacts ordered by groups.

=== CallerID-name (CNAM) Override ===

When you receive an incoming call to one of your numbers that matches a phone number in the Phone Book, the CNAM of the incoming call can be set in the entry on the Phone Book. For example, let´s say that you have an entry with the number 5554443322 associated with the name "John Smith", whenever you receive a phone call for that number you will see the CNAM as "John Smith". The Phone Book feature overrides the CNAM look up feature (if you have enabled it for your DID number).

For the name display from Phone Book to operate correctly, the displayed number must be written as it appears in the [[Call Detail Records]] - for North American numbers this will usually be ten digits without the leading +1.

=== CallerID Number Override ===

When you call one number using the Speed-Dial if you have setup a "CallerID Number Override" the default [[Caller ID|caller ID number]] will be changed for the one you have configured in your phone book entry.

=== Virtual Fax and SMS ===

You can also use the Phone Book entries with our Virtual Fax and SMS features. When sending from both features, the field will instantly show you the available entries to choose from when you start inputting either a number or a name from the Phone Book. You will also see the name configured in the Phone Book for received messages.

== Setup a Phone Book ==

First go to your Customer Portal and follow the menu option "DID Numbers" >> "Phone Book"

[[File:Phonebook Screen.png|thumb|none|800px]]

From this screen you can create, edit and delete Phone Book entries. Your Phone Book may contain a total of 500 entries.

=== Create a Phone Book Entry with a Phone Number ===

To create an entry, first click on the button "Add Phone Number", you will be prompted to the next screen:

[[File:Phonebook entry.png|thumb|none|800px]]

'''Speed Dial''': Here you only need to select which Speed-Dial you want to assign to this number. ''Optional Field.''

'''Name''': You can enter here the name that you want to use as CNAM Override. ''Mandatory field.''

'''Phone Number''': Enter here the phone number. ''Mandatory field.''

You can also use an International Number, you only need to make sure to use the prefix 00 or 011 to make the call to the number. Make sure that you have enabled the International Calls in your account.

'''Note''': The CallerID Number is not 100% guaranteed to be passed properly for International Routes at the moment.

'''CallerID Number Override''': Enter here the number you want to pass. This overrides the default callerID Number. ''Optional field.''

'''Group''': Select one of your phone book groups. You can also create a new Phone book group and the contact will be part of the group you are creating from the new entry directly.

'''Note''': The '''"General"''' group is the group existing by default. Any phone book entry created not assigned to a customized group, is assigned to the General group.

'''Note''': Here you can set a note to identify the entry. ''Optional field.''

=== Create a Phone Book Entry with a SIP URI ===

Creating a Phone Book Entry for a [[SIP URI]] is basically the same as creating the entry for a Phone Number. The difference is that you can select or create the [[SIP URI]] entry in this point in order to assign a Speed-Dial code . The information is selected for the [[SIP URI]] feature within your Customer Portal.

[[File:Phonebook SIPUri.png|thumb|none|800px]]

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

== Manage your groups ==

=== Create a Phone Book group ===

To create a new Phone book group, click on "Manage groups".

[[File:Manage group.png|thumb|none|600px]]

Click on "Add group" to create a new group.

[[File:Add group.png|thumb|none|600px]]

You will now be prompted a new window that allows you to set the Name of the group (we recommend using an easy to remember name like "Family").

Then click on '''Available members''' to see all your Phone book entries available to that moment. You can also see some basic information of the available members, including if they are part of another existing group.

Check the boxes of all the members you want to include in this group, and apply by clicking on the '''Save Phone Group''' option

[[File:Familygroup.png|thumb|none|600px]]

=== Edit your groups ===

If you have other groups you want to manage, you can click on the "Edit" icon (pen and paper icon), under the "Actions" column.

In "Group members" you will see all the members that are already part of the group you editing, and can remove them by clicking the Red X icon.

[[File:Edit group1.png|thumb|none|800px]]

In "Available members" you will see all the phone book contacts that are part of the general phone book or part of other groups, allowing you to change group they currently are for the one you are editing.

[[File:Edit group2.png|thumb|none|800px]]

=== Change the group of a contact ===

At any point, you can change the group of individual phone book entries. This is useful in case you change your mind for the group of a contact or if you created group after the contact.

From the main Phone Book screen, select the '''Edit''' icon (pen and paper icon under "Actions" column), then click on the '''Group''' option to see your list of available groups, select the one you desire and '''Save.'''

[[File:Edit contact2.png|thumb|none|800px]]

=== Search or order your contacts based on the group ===

The Phone book group feature introduces a new search criteria when going through your Phone book entries. You can type the name of a group in the search bar, and the customer portal will display only the Phone book entries that match the name of the group.

[[File:Search group.png|thumb|none|800px]]

To order the Phone book entries by groups, you simply click once on the "Group" column, and the portal will re-order your list of contacts based on the group name in alphabetic order.

[[File:Order groups.png|thumb|none|800px]]

== Import or Export the Phone Book ==

[[File:Phonebook export importpng.png|thumb|none|800px]]

Additionally you can import or export your Phone Book, the format is CSV (Comma-separated values). Here's an example of what the information should look like when you export your Phone Book.

*7501,"John Smith",5554443322
*7502,"John Doe",johndoe@other-provider.com

'''Note''': If you mark the box '''Overwrite Existing Phone Book Entries''' while Importing your Phone
Book, then all your existing Phone Book entries with a Speed Dial will be updated only with a valid match
for the same Speed Dial in your uploaded file.

[[category:guides]]

File:Order groups.png

2019-01-21T20:21:53Z

Edward:

File:Search group.png

2019-01-21T20:16:23Z

Edward:

File:Edit contact2.png

2019-01-21T20:09:50Z

Edward: