VoIP.ms Wiki - User contributions [en]

3CX Phone System

2016-12-16T19:01:03Z

Jeff: /* Inbound Rules */

3CX Phone System for Windows
Version 9.

3CX Phone System is a software-based IP PBX that replaces a traditional PBX and delivers employees the ability to make, receive and transfer calls. The IP PBX supports all traditional PBX features. An IP PBX is also referred to as a VOIP Phone System, IP PBX or SIP server.

Calls are sent as data packets over the computer data network instead of via the traditional phone network. Phones share the network with computers and separate phone wiring can therefore be eliminated. With the use of a VOIP gateway, you can connect existing phone lines to the IP PBX and make and receive phone calls via a regular PSTN line. The 3CX phone system uses standard SIP software or hardware phones, and provides internal call switching, as well as outbound or inbound calling via the standard phone network or via a VOIP service.

'''NOTICE:''' The latest update of 3CX, as of this writing, introduced a feature that might block Call Requests from Voip.ms. In order to solve this you need to go into 3CX, open system|parameters, find "SEC_IGNORE_USER_AGENT" and remove "voip"
from the Value Section. This only applies to 3CX V15 SP4.

==Adding VoIP.ms to 3CX==
Please make sure NAT is enabled on both your side and on the account you are using.

In the 3CX Phone System Portal, in the middle of the upper-part of the screen, we can see the '''Add Voip Provider Wizard''' , please click on the link. Once we are there we can put the '''Name of the Provider''', in this case we are going to enter''' Voip.ms''' and in the Provider list we can select '''Generic Voip Provider'''.

[[File:3CX trunk 1.jpg|1024px]]

Once we already click on continue, we will see the '''Voip Provider Details''' configure it according to the following instructions:

[[File:3CXtrunk2.jpg|1024px]]

*'''SIP server hostname or IP''': any of the 10+ VoIP.ms servers, all have the naming convention server.voip.ms (e.g: newyork.voip.ms).
*'''SIP Server port''': 5060
*'''Outbound proxy hostname or IP''': Leave in blank
*'''Outbound proxy port (default is 5060)''': 5060

For a full list of the servers and their IP's please check [http://wiki.voip.ms/article/Choosing_Server#IPs[Servers]]

Account details section, to complete this section please do it according to the following instructions:

[[File:3CXtrunk3.jpg|1024px]]

*'''External Number''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
*'''Authentication ID''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
*'''Authentication Password''': Your voip.ms password. (the password you set when you signup is the default SIP password too)
*'''Maximum simultaneous calls''': Specify how many concurrent calls your account supports.

'''Calls Routing'''

[[File:3CX trunk 4.jpg|1024px]]

Now specify how calls from this VOIP provider should be routed. You can specify a different route outside office hours.

End call
Connect to Extension
Connect to Queue
Connect to Digital Receptionist
Voicemail Box for extension
Forward to Outside Number

On the next page, you will be asked for a prefix so as to create an outbound rule for Voip.ms . Enter the dialing prefix in the “Calls to numbers starting with (prefix)” text box. To make calls via this provider, precede the number to be dialed with this prefix.

==Outbound Rules==

An outbound rule defines on which provider an outbound call should be placed,based on who is making the call, the number that is being dialed and the length of the number.

[[File:3CXOutboundRules.jpg|1024px]]

Specify for which calls to apply the outbound route. In the „Apply this rule to these calls‟ section, specify any of these options:

*'''Calls to Numbers starting with''': apply this rule to all calls starting with the number you specify. For example, specify 1 to specify that all calls starting with a 1 (usually a prefix) are outbound calls. Callers would dial „123456789‟ to reach the number „23456789‟

*'''Calls from extensions''': Select this option to define particular extensions or extension ranges for which this rule applies. Specify one or more extensions separated by commas, or specify a range using a -, for example 105-140.

*'''Calls with a Number length of''': Select this option to apply the rule to numbers with a particular digit length, for example 10 digits. This way you can capture calls to local area numbers or national numbers.

*Now specify how the outbound calls should be made. In the '''Make outbound calls on''' section, select up to 3 routes for the call. Each defined gateway or provider will be listed as a possible route. If the first route is not available or busy, 3CX Phone System will automatically try the second route.

==Inbound Rules==

With the Inbound rules we can configure calls made to that particular DID number to go to a particular extension, digital receptionist or other destination.
The latest update of 3CX introduced a feature that might block Call Requests from Voip.ms. In order to solve this you need to go into 3CX, 3CX Management Console > Settings > Parameters > Customer Parameters tab ,
find "SEC_IGNORE_USER_AGENT" and remove "voip" from the Value Section. This only applies to 3CX V15 SP4.

[[File:Create DID.jpg|1024px]]

*Click on the "Create DID‟ button in the 3CX Management Console in the toolbar.
*Enter a name for the DID (for example support).
*Now enter the DID number as it will appear in the SIP “to” header.

* Now select for which ports you wish to add this DID. If the DID number is associated with multiple ISDN ports, then you must select each. An inbound rule will be created for each port that you select.

*Now specify where you wish to direct calls made to this DID:
End Call
Connection to extension
Connect to Queue/Ring Group
Connect to Digital receptionist
Voicemail box for extension
Forward to outside number
Send fax to email of extension

You can specify that an incoming call is routed differently if it is received outside office hours. De-select the "Same as during office hours‟ option to specify a different route.

'''Click OK''' to create the DID / Inbound rule. The newly created DID‟s will be listed as inbound rules.

*There is record of an '''issue''' where you can't get incoming calls, and only outbound will work, like the issue mentioned [http://www.3cx.com/forums/cant-get-inbound-routes-from-voip-ms-did-to-work-30255.html Here]
The option '''"Source Identification by DID"''' can be found [http://www.3cx.com/blog/docs/did-voip-provider/ Here]

==Creating Extensions==

To add an extension, click on "Add Extension‟ from the toolbar.

[[File:Extensions 1.jpg|1024px]]

*'''Enter the extension number''': first and last name and the Email address (Optional) of the user. The email address will be used for voice mail notifications and as the default SIP ID. You can leave the field empty if you wish.

*'''Now specify an authentication ID and password''':

ID – The SIP "Username‟. e.g. 200.
Password –The SIP Password (password can be hidden from the user).

* '''Voicemail''': You can enable different features from your Voicemail

Enable voice mail
Play Caller ID – the voicemail system will play the number of the caller who left the voice message
Read out date/time of message – the voicemail system will play back the time of the voice message to be played
PIN number – this pin number is used to protect the voice mailbox and is used by the user to access the mailbox. The PIN number is also used
as a password to logon to 3CX Assistant or the MyPhone User portal.

* '''Click OK''' to create the extension.

After you have created the extension, a summary page will appear, which shows the information that the SIP phone will need:

* '''Proxy server IP or FQDN''': Host name of 3CX Phone System
* '''User ID''': Extension number created
* '''Authentication ID''': As specified in Authentication ID field
* '''Password''': As specified in Authentication password field

==Security Measures==

'''We strongly recommend you to change the password in your account, PBX system and extensions on it, periodically'''

'''As a preventive measure you also can disable International calls on your account'''.
'''From the customer portal >> Main Menu >> Account Settings >> Account Restrictions.
'''These settings define the restrictions the system will use when you place calls to either USA48, Canada or International Numbers.'''

''We strongly recommend to specifically select only the countries that you on your regular traffic (outgoing calls). You can do this by clicking in Currently Allowed: All Countries Allowed >> Click here to manage list of allowed countries''
[[Category:PBXes]]

3CX Phone System

2016-12-15T20:49:15Z

Jeff: /* Inbound Rules */

3CX Phone System for Windows
Version 9.

3CX Phone System is a software-based IP PBX that replaces a traditional PBX and delivers employees the ability to make, receive and transfer calls. The IP PBX supports all traditional PBX features. An IP PBX is also referred to as a VOIP Phone System, IP PBX or SIP server.

Calls are sent as data packets over the computer data network instead of via the traditional phone network. Phones share the network with computers and separate phone wiring can therefore be eliminated. With the use of a VOIP gateway, you can connect existing phone lines to the IP PBX and make and receive phone calls via a regular PSTN line. The 3CX phone system uses standard SIP software or hardware phones, and provides internal call switching, as well as outbound or inbound calling via the standard phone network or via a VOIP service.

==Adding VoIP.ms to 3CX==
Please make sure NAT is enabled on both your side and on the account you are using.

In the 3CX Phone System Portal, in the middle of the upper-part of the screen, we can see the '''Add Voip Provider Wizard''' , please click on the link. Once we are there we can put the '''Name of the Provider''', in this case we are going to enter''' Voip.ms''' and in the Provider list we can select '''Generic Voip Provider'''.

[[File:3CX trunk 1.jpg|1024px]]

Once we already click on continue, we will see the '''Voip Provider Details''' configure it according to the following instructions:

[[File:3CXtrunk2.jpg|1024px]]

*'''SIP server hostname or IP''': any of the 10+ VoIP.ms servers, all have the naming convention server.voip.ms (e.g: newyork.voip.ms).
*'''SIP Server port''': 5060
*'''Outbound proxy hostname or IP''': Leave in blank
*'''Outbound proxy port (default is 5060)''': 5060

For a full list of the servers and their IP's please check [http://wiki.voip.ms/article/Choosing_Server#IPs[Servers]]

Account details section, to complete this section please do it according to the following instructions:

[[File:3CXtrunk3.jpg|1024px]]

*'''External Number''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
*'''Authentication ID''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
*'''Authentication Password''': Your voip.ms password. (the password you set when you signup is the default SIP password too)
*'''Maximum simultaneous calls''': Specify how many concurrent calls your account supports.

'''Calls Routing'''

[[File:3CX trunk 4.jpg|1024px]]

Now specify how calls from this VOIP provider should be routed. You can specify a different route outside office hours.

End call
Connect to Extension
Connect to Queue
Connect to Digital Receptionist
Voicemail Box for extension
Forward to Outside Number

On the next page, you will be asked for a prefix so as to create an outbound rule for Voip.ms . Enter the dialing prefix in the “Calls to numbers starting with (prefix)” text box. To make calls via this provider, precede the number to be dialed with this prefix.

==Outbound Rules==

An outbound rule defines on which provider an outbound call should be placed,based on who is making the call, the number that is being dialed and the length of the number.

[[File:3CXOutboundRules.jpg|1024px]]

Specify for which calls to apply the outbound route. In the „Apply this rule to these calls‟ section, specify any of these options:

*'''Calls to Numbers starting with''': apply this rule to all calls starting with the number you specify. For example, specify 1 to specify that all calls starting with a 1 (usually a prefix) are outbound calls. Callers would dial „123456789‟ to reach the number „23456789‟

*'''Calls from extensions''': Select this option to define particular extensions or extension ranges for which this rule applies. Specify one or more extensions separated by commas, or specify a range using a -, for example 105-140.

*'''Calls with a Number length of''': Select this option to apply the rule to numbers with a particular digit length, for example 10 digits. This way you can capture calls to local area numbers or national numbers.

*Now specify how the outbound calls should be made. In the '''Make outbound calls on''' section, select up to 3 routes for the call. Each defined gateway or provider will be listed as a possible route. If the first route is not available or busy, 3CX Phone System will automatically try the second route.

==Inbound Rules==

With the Inbound rules we can configure calls made to that particular DID number to go to a particular extension, digital receptionist or other destination.
It appears there's something in the latest update of 3CX that is blocking Call Requests from Voip.ms. In order to solve this you need to go into 3CX, open system|parameters, find "SEC_IGNORE_USER_AGENT" and remove "voip"
from the Value Section. This only applies to 3CX V15 SP4".

[[File:Create DID.jpg|1024px]]

*Click on the "Create DID‟ button in the 3CX Management Console in the toolbar.
*Enter a name for the DID (for example support).
*Now enter the DID number as it will appear in the SIP “to” header.

* Now select for which ports you wish to add this DID. If the DID number is associated with multiple ISDN ports, then you must select each. An inbound rule will be created for each port that you select.

*Now specify where you wish to direct calls made to this DID:
End Call
Connection to extension
Connect to Queue/Ring Group
Connect to Digital receptionist
Voicemail box for extension
Forward to outside number
Send fax to email of extension

You can specify that an incoming call is routed differently if it is received outside office hours. De-select the "Same as during office hours‟ option to specify a different route.

'''Click OK''' to create the DID / Inbound rule. The newly created DID‟s will be listed as inbound rules.

*There is record of an '''issue''' where you can't get incoming calls, and only outbound will work, like the issue mentioned [http://www.3cx.com/forums/cant-get-inbound-routes-from-voip-ms-did-to-work-30255.html Here]
The option '''"Source Identification by DID"''' can be found [http://www.3cx.com/blog/docs/did-voip-provider/ Here]

==Creating Extensions==

To add an extension, click on "Add Extension‟ from the toolbar.

[[File:Extensions 1.jpg|1024px]]

*'''Enter the extension number''': first and last name and the Email address (Optional) of the user. The email address will be used for voice mail notifications and as the default SIP ID. You can leave the field empty if you wish.

*'''Now specify an authentication ID and password''':

ID – The SIP "Username‟. e.g. 200.
Password –The SIP Password (password can be hidden from the user).

* '''Voicemail''': You can enable different features from your Voicemail

Enable voice mail
Play Caller ID – the voicemail system will play the number of the caller who left the voice message
Read out date/time of message – the voicemail system will play back the time of the voice message to be played
PIN number – this pin number is used to protect the voice mailbox and is used by the user to access the mailbox. The PIN number is also used
as a password to logon to 3CX Assistant or the MyPhone User portal.

* '''Click OK''' to create the extension.

After you have created the extension, a summary page will appear, which shows the information that the SIP phone will need:

* '''Proxy server IP or FQDN''': Host name of 3CX Phone System
* '''User ID''': Extension number created
* '''Authentication ID''': As specified in Authentication ID field
* '''Password''': As specified in Authentication password field

==Security Measures==

'''We strongly recommend you to change the password in your account, PBX system and extensions on it, periodically'''

'''As a preventive measure you also can disable International calls on your account'''.
'''From the customer portal >> Main Menu >> Account Settings >> Account Restrictions.
'''These settings define the restrictions the system will use when you place calls to either USA48, Canada or International Numbers.'''

''We strongly recommend to specifically select only the countries that you on your regular traffic (outgoing calls). You can do this by clicking in Currently Allowed: All Countries Allowed >> Click here to manage list of allowed countries''
[[Category:PBXes]]

FAQ

2016-08-15T15:43:35Z

Jeff: /* What are the Payment Options? */

[[File:Logofaq.jpg|250px|]]

__TOC__

Pour la page en Francais veillez cliquez ici: [[Questions_Les_Plus_Fréquentes|Questions Les Plus Fréquentes]]

Haga clic aquí para Español:[[Preguntas frecuentes]]

==Do you offer a Test Account or Number?==
We currently do not have a test account or test number program however opening an account is completely free and we provide free dialing codes to test the quality of the service.
Please check our many dialing codes here: [[Dialing Codes]]

==How do I start?==
You can open an account for free following this link : https://voip.ms/signup.php
Once your account is active you will be able to log in and find all of our features and options.
We suggest to also check our [[Getting Started]] guide.

==Is there a cost to open the account?==
Opening an account is completely free. The minimum amount of funds you can add is $25 USD which will become your account balance to use the services.

==Is there a Contract?==
There are no contracts with us, you can open an account for free at any time you wish, for as long as you want.

==What are the Payment Options?==
The current options available are Paypal (No Account Needed for Paypal Guest CC Payment), VISA, Master Card, American Express, Discover (Where supported) and via Bank Wire deposit to our bank account in Montreal.
Please contact Customer Support for information about Bank Wire.

==Can I deposit less than $25 into my account?==
Currently $25 is the minimum amount you can add to your account. This credit won’t expire and will last until depleted by using the services.
You can ask for a refund of the remaining balance at any time should you choose to cancel the account.

==Do you charge taxes?==
All customers with Canada as country will pay for the tax called GST. The customers who are also from Quebec in Canada will pay an additional province tax, PST. This applies for both Paypal and Credit Card payments.
The HST will get applied on Canadian provinces that require it.
In order to get further details about how these are applied, feel free to check this link: http://en.wikipedia.org/wiki/Sales_taxes_in_Canada

==How much does the service cost?==
The cost will vary depending on the services used.
For outbound calls you are subject to the per minute rate of the destination you call and the billing increment.
For inbound calls you are subject to the rate plan of the DID, the monthly fee and the billing increment it if applies.
There may be additional monthly fees or set up fees depending on the services you subscribe to.
Please check more details about the service cost here [[Service Cost]]

==What is the Billing Increment?==
The Billing Increment is a call duration measurement unit expressed in seconds.
For example, with a $0.0100 per minute rate, and a 60 seconds billing increment, any call less than 60 seconds is rounded up to 60 seconds or 1 full minute of usage. A 5 seconds call and a 50 seconds call are both billed as 1 minute or $0.0100.

With our 6 seconds billing increment, a 5 second call will be billed as 6 seconds or 1/10 of 1 minute or $0.0010.
This way, you don't get charged for a whole minute just for a 10 seconds call.

==What is a DID?==
A Direct Inward Dial number (DID), in simple terms, is a virtual number that, for all intents and purposes, can be considered a regular phone number, with the exception that it is not attached to any POTS line (Landline). Once your configuration is ready, your DID will be the phone number that everyone in the world will call to reach you, just like any other phone number.

==What type of DID numbers you offer?==
There are different types of DID numbers. They are mostly differentiated by their geographic presence.
Local DIDs from US or Canada.
International DIDs.
Toll Free numbers.

These are the most common used by our customers. For further information about getting a DID number and other types please check our related article [[Order a DID Number]]

==Can I dial out with my DID?==
The DID numbers are a type of service that are exclusively for receiving calls.
When you place outgoing calls with our service you are not calling from the DID number but from the account instead and the number you will pass on your calls as your Caller ID will be the one that is configured on the accounts settings from the portal.

==What is Caller ID?==
Caller ID is a telephone service that transmits the calling party’s number to the called party’s telephone . When available, the Caller ID number can be complemented with Caller ID name (e.g. John Smith).
If you are placing outgoing calls you will likely need to pass a Caller ID to ensure proper termination of your calls, particularly to reach toll free numbers.
There are two types of caller ID and it is important to differentiate them: Caller ID Number (CID) and Caller ID Name (CNAM).
Please contact Technical Support for assistance or more details about Caller ID or check our related article at [[Caller ID]]

==What is CNAM?==
CNAM stands for CallerID Name and it's the information that will be displayed in the phone of the receiving party, when you place an outgoing call.
If you will be making calls to Canadian numbers, you can simply pass the Caller ID name from your device or system as most of them support this.
The Caller ID name on US calls works differently, this is controlled by a national CNAM database with records of numbers and names matching each number.
We can update the CNAM database under request. For more information please check our related article at [[http://wiki.voip.ms/article/Caller_ID#Outgoing_Caller_ID_name Caller ID Name]]

==Can I port my existing number from another provider to VoIP.ms?==
VoIP.ms does offer Local Number Portability (LNP) service and your number may be available for porting. Please contact Customer Service to find out whether your current number is portable to VoIP.ms network or if you already have an open account, please refer to the [[Porting a Number]] guide.

==Do you provide any hardware for the service?==
We don’t provide any kind of hardware device, software or system to use the service. The service is a BYOD (Bring your own device). You should be able to get one from any Communications specialized store and all SIP- compatible devices are supported.

==Can I use my existing device with VoIP.ms?==
Basically any device or system that supports SIP or IAX2 protocol will work with our service . If you bring your device (ATA, IP phone) from a previous provider, make sure it is unlocked and you are able to make changes to its configuration. You can find a list of some devices that we support, with their configuration guides, at [[Devices]]

==Do you have Configuration Samples?==
You can find configuration samples for most of the common devices and phone systems used with VoIP.ms here: [[Devices]]
If your device or system is not listed there but supports the protocol SIP or IAX2 and at least one of the following codecs: G711u, G729a or GMS, you should be able to use it, and you can always contact our Technical Support staff for additional assistance.

==Do you offer Technical Support by phone?==
Due to the nature of the troubleshooting which can result time consuming we chose our main ways to offer support to be via Ticket System (Email) and Live chat.
At this point, Technical Support over the phone is not available.

==What is my Main Account SIP password?==
The main account SIP password is, by default, the same as your customer portal password. If you have not changed the SIP password, it is the same password you use to log into your portal. You can change your SIP password from the Customer Portal at any time from the Main Menu - Account Settings page. Please check our related article for more information: [[http://wiki.voip.ms/article/Account_Settings#Security Account Security]]

==What Server should I use?==
Usually, in order to receive better results, you should choose the server physically closest to your location. You can also send a ping to any of the servers to check the best response time.
Please check our related article for more information about this subject [[Choosing Server]]

==Does VoIP.ms encrypt the communication?==
The SIP communication is secure although not encrypted. However, the passwords are MD5 hashed and are not transmitted without encryption when establishing the call.

==Do you Support e911==
We offer e911 service only for US and Canadian DID numbers (including USA and Canadian Toll-free numbers). This feature can be activated in the Customer Portal directly under the e911 page under the DID numbers menu.
More information available in our related wiki article here [[E911]]

==Do you offer Calling Card?==
We do not offer any Calling Card feature at this moment, however you can set up a Calling Card solution with your own system and our DID numbers as long as this is used with DID numbers in our Per Minute plan.
For more information please contact Technical Support.

==Do you offer Fax service?==
We offer a Virtual Fax feature (currently on Beta). This feature can be used with special Fax DID numbers which can be acquired and configured through our Fax portal.
Please check our related article for further details: [[Virtual Fax]]

==Do you offer SMS service?==
We do offer SMS service (currently on Beta). The SMS is available for a great number of DIDs and Cities from US and Canada and requires to be used through the SMS portal.
SMS is not available through SIP protocol for the moment.
You can read more details about this feature in our related article [[SMS]]

==Do you offer Conference Calls?==
We do not offer Conference calls as a feature in the customer portal, however if your system or device is capable of establishing a conference call or possess this feature, you can use it with our services.

==Do you support TCP for the SIP communication?==
With the exception of Atlanta1, TCP is fully supported in our servers.

==Do you offer alternative ports besides 5060?==
We offer alternative SIP ports, UDP/TCP 5080 and 42872 on all of our servers, You can try those ports in case your Internet Service Provider blocks the port 5060 UDP/TCP or if you need to use another one.

==Can I register 2 or more different devices with the same account username?==
It is strongly suggested not to do so, this can possibly cause conflicts while routing the calls to your device as well issues related with registration. If you need to register more than one device please create and use the Sub Accounts, you will get new credentials for any additional device.
For more information about Sub Accounts please check the related article on this subject [[Sub Accounts]]

==Do you accept Autodialers or Telemarketing traffic ?==
We do not accept the use of our termination (outbound) services for telemarketing purposes (Including but not limited to Automated Dialers, Call Centers and collection agencies).

==Do you offer service to Call Centers ?==
We currently provide service to Call Centers for inbound traffic, the use of this is restricted to DID numbers with a per minute plan.
Outbound traffic from call centers is not supported at this time. If you have any concern about this please contact Customer Support for more details.

==Can I resell your services ?==
Yes, our customers will be able to find a "Reseller Section" included in the Customer Portal.
We give you all the tools you need plus a White label Reseller Interface that has been developed to help our clients upsell our services under their own brand. We also recommend that you must familiarize yourself with the Voip.ms interface before reselling the service.
For more information about reseller please contact Technical Support or visit our related article here: [[Reseller Basic Guide]]

==Do you offer Wholesale Rates ?==
If you are interested on a discount based on traffic usage or volume, please send an email to sales@voip.ms providing all details about your traffic like destinations you need to call and average of minutes used per month, in order to receive additional information and quotes.

==Can I port out my number ?==
If you wish to port out a number from our service you can do so at any time by starting the porting request with the new provider. We authorize all Port Out Request matching the correct information from your account.
Remember to delete the DID number from your VoIP.ms account once the port out process is completed.
For any question regarding Port Out please send an email to our LNP Department at ports@voip.ms

==How do I delete my DID ?==
From the customer portal, go to the “DID Numbers” menu tab and then click “DID Billing”. From this page, you can cancel number(s). Remember this process cannot be reversed, please be sure that you will no longer need the number.

==Is there any Cancellation Fee ?==
We do not charge any cancellation fee of any kind, you are free to cancel your account or number at any time you wish to.

==Where are you located ?==
VoIP.ms is a Canadian Company founded back in 2007 in Montreal.
The main office is located in Montreal, QC, Canada and there is also a Technical Site in Merida, Yucatan, Mexico for the South America market.

==Do you offer a Referral Program ?==
For the moment we don't have any kind of referral program available.

[[Category: Guides]]

Voicemail

2016-08-01T18:02:42Z

Jeff: /* Voicemail Greeting Customization */

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

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

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

== Voicemail Accounts ==

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

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

== Creating a Voicemail mailbox ==

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

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

You will be prompted with the following information:

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

== Assigning your Voicemail to your DID==

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

[[File:selectVoicemail.png|750px|thumb|left|Select Voicemail - click to enlarge]]
 

== Assigning your Voicemail to your Account==

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

[[File:Mainvoicemail.jpg]]

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

[[File:Subvoicemail.jpg]]

== Manage Voicemail and Messages ==

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

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

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

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

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

== Voicemail Greeting Customization ==

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

* The '''First''' option is to do it directly from your Phone, once you have created the Mailbox and you have assigned it to your DID number and to your account, dial *97 from your phone and at the Voicemail menu select the option "0" (Voicemail options) and then the option "4" (Temporary message) there you will be able to record your greeting, and to save it.

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

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

== Voicemail Access Codes: ==

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

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

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

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

== Navigate the Voicemail Menu ==

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

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

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

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

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

'''Notes:'''
* The Unavailable message ('''Option 1''') and Busy message ('''Option 2''') are not currently working with the voicemail system of VoIP.ms.
Use the temporary greeting, which overrides these standard greetings.
* To Delete a Recording you can just record again, a moment of silence, or create a new recording overwriting the old.
* Please note that if you change the password over the phone ('''Option 5'''), it will not change at your customer portal>>Voicemail options,
and you will have to manually change the password there. If you don't don't change it at the customer portal and you modify any other value
later, when you apply the changes the old password will be set again to your mailbox, since the modify option apply all the values set at
the voicemail options.

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

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

* 4 - Play the previous message.

* 5 - Repeat the message.

* 6 - Play the next message.

* 7 - Delete the current message, without confirmation.

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

== Known Voicemail Issues or Frequent Questions ==

'''I Can No Longer Listen to My Voicemail on my Android Phone.'''

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

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

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

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

'''I Keep Receiving Cut Off Voicemails from People'''
While recording a message the system listens for 5 seconds of silence. If the systems detects no sounds for 5 seconds it will
Stop the VM Recording and send a person to VM options.

'''How Can I Delete All Voicemail Messages in My Mailbox.'''

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

'''I have Voicemail Messages but when I login the system tells me I Have No Voicemails'''

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

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

[[category:guides]]

Features

2016-07-13T22:22:46Z

Jeff:

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

One of the benefits of having a VoIP.ms account is that not only do you get both origination and termination services, but VoIP.ms also offers a complete solution for your business or home by providing you with many options and features to make your life easier than ever. Most of the Features listed below are completely free with the service and you will not need to pay extra to have these awesome benefits. To fully understand each of these features, you can click on the the feature name to access the specific guide for each topic.

{|border="1"
|+
!Feature !! Summary
|-
! [[Call Detail Records|Call Details Report and Invoice Generation]]
|You will be able to examine detailed information of all the calls coming and going from your Main account and Sub accounts simply from the portal.
Additionally you will find the '''Finances''' Menu, where you will get access to the tool to Generate Invoices according to your needs and also be able to add funds to your account.
|-
! [[Callback]]
|With this feature, you can define a number to be called by our system, in order to receive a dial tone and place outgoing calls through VoIP.ms. This could be useful if you want to place a call and you are not at home or don't have access to your voip device.
|-
! [[Caller ID]]
|Caller ID is a telephone service that transmits the calling party's number to the called party's telephone. When available, the Caller ID number can be complemented with the caller ID name (description e.g. John Smith)
|-
! [[Call Forwarding]]
|A Call Forwarding allows an incoming call to be redirected to a mobile telephone or another telephone number where the desired called party is able to answer. You can set any number, even international numbers.
|-
! [[CallerID Filtering]]
|This feature allows you to filter the incoming calls to your DID numbers given a specific Caller ID number, area code or even anonymous numbers. For example, if you receive annoying incoming calls from a telemarketing company you can create a filter to route all the calls to a recording that plays the message "That number is no longer in service, please hang-up and try again", amongst several other options.
|-
! [[Calling Queues]]
|If you want a solution to manage your incoming calls and have your customer(s) wait on the line while an agent picks up the call, you need to create a Calling Queue entry. This will permit you to have many calls on hold, queued calls in First In-First Out order until agents become available.
|-
! [[Porting a Number|DID Portability]]
|Don't lose that number that you like so much, if you are a VoIP.ms customer and have an existing number with a different provider you can optionally bring it (port it in) to VoIP.ms network. Toll free numbers, local USA/Canada and even International numbers can be ported to VoIP.ms.
The LNP (Local Number Portability) Department handles all porting processes and guides the customer through it. They are also in charge of updating the customer via email with any relevant information about the port every time there is a change on the port request status.
|-
! [[DigitalReceptionist_IVR|Digital Receptionist (IVR)]]
| Interactive Voice Response (IVR), (also known as Auto-attendant) this option can be used to present a recording to the people calling to your DID number and also give them the chance to enter the extension of the person or department they want to reach. For example, you could create an IVR and point one of your DID numbers to it, and when the IVR answers, the caller will hear "Thank you for calling XYZ Inc, for Sales press 1, for Service press 2", etc.
|-
! [[DISA]]
|Direct Inward System Access ( DISA ) allows you to use our system for placing outgoing calls, even if you are not close to any device where you are registering your account or sub account. In this case you just would need to dial to your DID number and to provide a 4 digit PIN number, you can then dial out to any number in the world under our termination rates.
|-
! [[E911]]
|VoIP 911 Service differs from traditional 911 services due to limitations brought on by VoIP technology. It is nearly impossible to detect where a call originates from when placed over the internet. e911 allows users to associate physical addresses with their DIDs, allowing them to have service similar to traditional 911.
'''Important Note: Use of our 911 Service costs a recovery setup fee of $ 1.50 on activation and a regulatory recovery fee of $ 1.50 per DID number activated per month.'''
|-
! Failover for Incoming calls
|Whether your system is unavailable or you are experiencing an Internet or power outage, VoIP.ms offers you the option to redirect the incoming calls to your DID numbers to any location where you can take the call, for example like your cell phone.
Since you are virtually able to select any regular Routing option for the failovers, you can customize your setup in different ways by selecting one of three states, No Answer, Unreachable or Busy.
|-
! Free Calls Between VoIP.ms Customers
|Save even more money with this feature, with VoIP.ms you get free calls from your account to any VoIP.ms Voice DID from US and Canada. So if your friends, family or customers have VoIP.ms DIDs from US/Can, you can be sure that the call will be FREE.
|-
! [[Sub_Accounts#Optional_Settings|Internal Extensions]]
|This feature will allow you to define a quick dial access number to contact the subaccount directly and internally, this way you will be able to reach each a sub account by dialing an extension number E.G "102". This is an amazing feature because calls between extensions are free, so you can have one extension at your office or home in Canada and another one in the US or another country in the world and reach that subaccount dialing only the extension number. With this feature your business can be more efficient and also your friends and family will be in touch easily.
|-
! IP Authentication and IAX support
|Besides the regular method of authentication (Username and password), VoIP.ms also offers the customer the option to authenticate their system via IP address, this way the customer can configure their system to fit their needs.
Even though the native and regular protocol is SIP, VoIP.ms also supports IAX protocol for Customers wanting to communicate this way.
|-
! [[Account_Settings#Notifications|Low Balance Email Notifications]]
|Since it could be time consuming to constantly keep track of the Account balance, VoIP.ms offers you the option to receive an Email notification to your email address, when your account reaches a Low balance limit, below your defined Balance Threshold.
|-
! Music on Hold
|With our services you can easily chose if you want to play "Music on Hold" when the call is on hold state, the caller will hear music from a variety of choices while they wait on line. Also you can deactivate this option if you don't want to play Music to the caller, this can be done from your '''customer portal >> Main Menu >> Account Settings >> General. '''[http://wiki.voip.ms/article/Account_Settings#General]
|-
! [[Phone book]]
|The Phone Book feature allows you to configure Speed-Dial entries and Caller ID name (CNAM) overrides. For example, let's say you have a Customer, Provider or Relative that you call frequently, you can create a phone book entry in order to make a call using a speed-dial entry of 4 digits long. You can also have a Caller ID name (CNAM) override to identify the incoming calls of an important customer if his number doesn't have a proper Caller ID name (CNAM) linked to it, for free.
|-
! [[Recordings]]
|VoIP.ms allows you to upload an audio file and use it in the different options we have under the DID numbers menu. It can be used with several of the other features of our system such as Digital receptionist, Calling queues, and others.
|-
! [[Ring Groups]]
|The Ring Group feature allows you to have incoming calls to be redirected to different destinations that are included in your Ring Group, where any member of the group is able to answer. When you receive a call to a DID routed to a Ring Group, all members of that group will ring at the same time until one of them answers the call. You can add various types of members to a ring group:
Main Account,[[Sub Accounts]],[[SIP URI]]'s,[[Call Forwarding]].
|-
! [[Reseller Basic Guide| Reseller]]
|The Reseller feature is intended for users who want to initiate a business in the VoIP world by reselling VoIP.ms services, this useful tool can help you create and manage accounts for your own clients, you will be able to assign specific rates and DIDs to them and also provide credentials to register a device or call system. You will be able to handle the billing for your customers depending on your needs and how much you want to earn with your business, setting rates and adding credits for them. Additionally you will find options to customize your reseller portal to have a more personalized site.
|-
! [[SIP_URI|SIP URI]]
|A SIP URI is the SIP addressing schema to call another person via SIP. In other words, a SIP URI is a user's SIP phone number or address. The SIP URI resembles an e-mail address.
|-
! [[SMS]]
|
This feature allows you to send and receive messages with your DID Number (US and Canada DID Numbers Only). Currently it is in beta and will remain free until further notice.
|-
! [[Sub Accounts]]
|Having a Sub Account allows you to register more than one device to make or receive calls simultaneously, you can also use it as internal extensions for your office or even your house. Many of the features within VoIP.ms make use of the sub accounts. With this guide we are going to learn how to create and use this feature properly.
|-
! [[Time Conditions]]
|Time Conditions is a feature that allows you to route your incoming calls to different destinations depending on the time of the day of the call. For example, you can have a Time condition for your Non-Business hours and all the incoming calls you receive will be routed to a recording or the voicemail directly (you can also route to other options, like IVR, Call Forwarding entry, etc.).
|-
! Vanity Toll Free Numbers
|Are you looking to have that special contact number for your business with specific digits? VoIP.ms offers the option to order a customized Toll Free number with any combination of digits you like.
Either for business or personal use, you will be able to use the tool from the portal and find that number.
|-
! [[Virtual Fax]]
|This Feature Is Currently Available For Beta Customers.You can sign up for Beta testing by clicking [https://www.voip.ms/m/beta.php Here]. The Virtual Fax feature is used for sending and receiving a Fax (facsimile) with the VoIP.ms service using a DID number specifically dedicated to Faxing. You may obtain such a number from your Customer Portal in the Fax Numbers section under the Order DID(s) of the DID Numbers menu. Regular voice DID numbers are not compatible with the Virtual Fax feature.
|-
! [[Voicemail]] ___________________________________________________
|VoIP.ms has an advanced Voicemail feature that is free to use, which also provides you the option to forward your messages to your email address as an audio attachment.
In order to use the Voicemail feature with Voip.ms you will have to create a Voicemail entry and then assign your entry to one of your DIDs or Accounts.
|-
|}

'''If you have more inquires regarding the Features, feel free to contact the support team via the Live Chat or the Ticket system.

==Contact Us==
[[Image:Download.jpg|left|50x75px|link=https://livechat.boldchat.com/aid/2947277729005480016/bc.chat?cwdid=60236691424546376&url=https%3A//www.voip.ms/]][https://livechat.boldchat.com/aid/2947277729005480016/bc.chat?cwdid=60236691424546376&url=https%3A//www.voip.ms/ '''Live Chat Support'''] [[Image:Download.png|50x75px|link=https://www.voip.ms/m/hp/tickets.php]] '''support@voip.ms'''

TP-Link TD-VG3631

2016-07-13T16:59:56Z

Jeff: Created page with "300Mbps Wireless N VoIP ADSL2+ Modem Router <div style="width:100%;overflow:hidden;clear:both"></div> == General In..."

[[File:Tplink vg3631 front.png|500px|thumb|center|300Mbps Wireless N VoIP ADSL2+ Modem Router]]
<div style="width:100%;overflow:hidden;clear:both"></div>

 

== General Information ==

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

== User's Guide ==

Here you can see the complete user guide for this device.

[http://www.tp-link.com/Resources/document/TD-VG3631_V_1_User_Guide.pdf User's Guide]

== Logging into your device ==

Please put the Default IP of 192.168.1.1 in your Internet Browser (IE: Chrome or Firefox) as http://192.168.1.1

When the Login screen comes up please set the following for the login credentials.

Login: admin

Password: admin

[[File:Tplink Login.png|600px|thumb|center|]]

== Configuration Details ==

Once you have logged into your device you will see the Status Page.

[[File:Tplink Status.png|600px|thumb|center|]]

*Please look in the Left Hand Column for '''Voice''' and click on '''SIP Account'''

[[File:Tplink Voice Sip 1.png|600px|thumb|center|]]

You will see the Sip Account Management page and you can either click on '''Add''' or click on '''Edit''' to edit an existing account.

Then you will see the Page where you can configure your TP-Link Lines.

[[File:Tplink Voice Sip 2.png|left|900px|thumb|upright=0.35|alt=ProfileName: Set any name you like to Identify the account.Test]]

*'''ProfileName:''' Set any name you like to Identify the account.

*'''Display Name:''' Set the CallerID Name you would like for your outbound calls.

*'''Authentication ID:''' 100000 (Set with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub) Shown on your Portal Home Page.

*'''Registrar Address:''' Atlanta2.voip.ms (You can choose any of our multiple VoIP.ms [http://wiki.voip.ms/article/Choosing_Server servers]

*'''Sip Proxy:''' Atlanta2.voip.ms (You can choose any of our multiple VoIP.ms [http://wiki.voip.ms/article/Choosing_Server servers]

*'''Outbound Proxy:''' Atlanta2.voip.ms (You can choose any of our multiple VoIP.ms [http://wiki.voip.ms/article/Choosing_Server servers]

*'''Phone Number:''' 100000 (Set with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub) Shown on your Portal Home Page.

*'''Password:''' Your VoIP.ms SIP Password. You can Confirm your Main Account SIP Password [https://www.voip.ms/m/samples/voxalot.php HERE]

*'''Registrar Port:''' Set 5080

*'''Sip Proxy Port:''' Set 5080

*'''Outbound Proxy Port:''' Set 5080

Now you should be able to save these settings and restart the TP-Link. After a restart please test calling into and out of your TP-Link.

File:Tplink Voice Sip 2.png

2016-07-13T16:01:39Z

Jeff:

File:Tplink Voice Sip 1.png

2016-07-13T15:58:09Z

Jeff: Voice Sip Manage Account Page

Voice Sip Manage Account Page

File:Tplink Status.png

2016-07-13T15:55:58Z

Jeff: Status Home Page

Status Home Page

File:Tplink Login.png

2016-07-13T15:46:07Z

Jeff: TP Link Login Page

TP Link Login Page

File:Tplink vg3631 front.png

2016-07-13T15:13:48Z

Jeff:

Virtual Fax

2016-06-29T18:39:20Z

Jeff: /* My Faxes */

[[File:Faxhomelogo.png|center]]

The Virtual Fax feature is used for sending and receiving a Fax (facsimile) with the VoIP.ms service using a DID number specifically dedicated to Faxing. You may obtain such a number from your Customer Portal in the Fax Numbers section under the ''Order DID(s)'' of the ''DID Numbers'' menu.
Regular voice DID numbers are not compatible with the Virtual Fax feature.

__TOC__

== Important information to know about the Virtual Fax Service ==

* '''The Virtual Fax Service is in BETA version, that means it is not fully deployed. It is important to us that you report any issues with this service by sending an email to support@voip.ms so that the developers can get involved if necessary.'''
* '''The Virtual Fax Service is only available for U.S. and Canadian DID Numbers specifically acquired from the Fax Numbers ''Order DID'' section'''
* '''It is also possible to port your Voip.ms Voice DID Numbers and Numbers from other Providers into our Virtual Fax service. For numbers from other providers, you can find this option under the ''DID Portability'' section of the Customer Portal. For Voip.ms numbers, you can request an internal port by sending an email to our LNP department at ports@voip.ms. The porting fee is $15 per number for both options.'''
* '''The Service can only be used to send Faxes to Canadian and U.S. Numbers at this time. We also cannot guarantee that International will be properly received.'''

== Cost and Rates ==

Setup Fee: $0.00 (Currently Free)

Monthly Fee: $1.99

Per Minute Fee: $0.0290 (2.9 Cents)

[[Calls Cost|Billing Increment]] : 6 seconds

If you are wanting to calculate how much this could cost, per page faxed, converted into the time of per min charges you will want to take into consideration that this depends a lot on the destination fax speed and the content (How much of the page has something other than white) being faxed. Unfortunately there are no definite ways in calculating costs. If you are sending just text, a page can go from 30 secs up to 1 min. If you are faxing multiple page documents this will go faster since you are not having to handshake or negotiate with the far side's fax machine for each page. This can result in 1.5 cents to 3 cents on a 1-2 page fax. Since we also increment the charge to every 6 seconds you are saving even more money since we do not charge you a full min for partial minute usages. You can usually get around 1-2 pages faxed per minute at 2.9 cents and $2 a month which is much less than most other Electronic Fax services being offered at this time.

== Current Limitations ==

*During the Beta test, each DID Number can only send 100 messages per day. This limit can be raised upon request and verification.

*The files sent per message cannot exceed 25mb

== Virtual Fax DID Number ==

Virtual Fax works specifically with Fax Numbers only acquired from the VoIP.ms Customer Portal or numbers ported in specifically as Fax enabled. There are '''local US and Canadian numbers''' available for order. You can order a Fax DID Number from your portal >> DID Numbers >> Order DID >> Fax Numbers. You can select the desired region and a random number from the chosen area code will be assigned to you.

[[File:FaxorderDID2.jpg|700px|]]

You can also port in a number you currently own with another provider. This process can be started from the Customer Portal at DID Numbers >> DID Portability >> Porting Fax Numbers

[[File:FaxPortability.png|700px]]

== Send a Fax ==

To start using the service you will need to head to your Customer Portal >> DID Numbers >> Virtual Fax. From the Home Page you can select ‘Send Fax’. There you will see:
*Fax Number or Contact Name: This is where you will put the destination number. You can start typing a name or a number from your Phone book and it will become available.
*From Name: Here you will put the name to send in the Fax header.
*From Number: Select the Fax DID number from which you will send your Fax.
*File: Choose a file to send as a Fax. The file must be in pdf, txt, jpg, gif, png or tif
-'''IMPORTANT: The maximum file size is 25MB'''
*Station ID: This will be the station ID you set for the header of the Fax message. It could be a specified post if your location has several stations, such as Reception, Main Office, Accounting PC, etc.
*Send Email: If selected, an email will be sent to the specified address to confirm the Fax has been sent successfully or to advise of a failed attempt.

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

== My Faxes ==
In this section of the Virtual Fax menu you will be able to view your Inbound and Outbound Faxes. You may select a date range and choose the folder you would like to view. Click 'Get My Faxes' to view your selection. You can view the Status of each Fax and select from several Actions. You can select to View the Fax directly, Download the Fax, Email the Fax to an address of your choosing or alter the location of the Fax by moving it to another folder.

Only the Status gets updated automatically. You will have to refresh the page to get the costs after a fax completes.

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

== My Folders ==

In the 'My Folders' section you can create folders by typing in the folder name of your choosing under 'New Folder' and clicking 'Create'.
You will have an overview of your Folders, see the date they were Created, the amount of Faxes in each Folder and be able to Edit the Folder or Delete it.
Any Faxes contained in a created folder will revert back to either the INBOX or SENT folder if the created folder is deleted.

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

== My Fax Numbers ==
In the 'My Fax Numbers' section you will see your Fax DID numbers and Description, the Options that have been enabled for each number, the Email address if one has been configured along with the URL if configured in the URL Callback section. You can Edit the number from the 'Actions' section or choose to Delete it. When editing you will have the option to set an Email Address to receive a notification when a new Fax is received (you can also select to have the PDF file attached in the email) and set a URL Callback (you can also enable URL Callback Retry).

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

== Email to Fax==

This feature allows you to send a Fax message using your email account.

How to send a Fax message using your email account:
*Use the email account you provided when enabling the Email to Fax service.
*Send the email to fax@voip.ms
*In the subject field type the Destination Fax Number (Example: 5148000000).
*Attach the document you wish to send to the email message. VoIP.ms supports the following formats: pdf, txt, jpg, gif, png, tif.
*Send the email.
**'''NOTE: Only the file attached will be faxed, the body of the e-mail won't be transmitted'''

'''Security Code and From Number:'''

If Security Code is enabled, you need to add a dot (.) and the Security Code after the Destination Fax Number (Example: 5148000000.Az09).

If you have more than one Fax number, you could change the From Number by adding a dot (.) and the From Number you'd like to use after the Destination Fax Number and the Security Code (Example: 5148000000.Az09.2268280000).

If you have not enabled the Security Code and want to change the From Number, you can add a dot (.) and the From Number after the Destination Fax Number (Example: 5148000000.2268280000).

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

Virtual Fax

2016-06-29T18:36:38Z

Jeff: /* Cost and Rates */

[[File:Faxhomelogo.png|center]]

The Virtual Fax feature is used for sending and receiving a Fax (facsimile) with the VoIP.ms service using a DID number specifically dedicated to Faxing. You may obtain such a number from your Customer Portal in the Fax Numbers section under the ''Order DID(s)'' of the ''DID Numbers'' menu.
Regular voice DID numbers are not compatible with the Virtual Fax feature.

__TOC__

== Important information to know about the Virtual Fax Service ==

* '''The Virtual Fax Service is in BETA version, that means it is not fully deployed. It is important to us that you report any issues with this service by sending an email to support@voip.ms so that the developers can get involved if necessary.'''
* '''The Virtual Fax Service is only available for U.S. and Canadian DID Numbers specifically acquired from the Fax Numbers ''Order DID'' section'''
* '''It is also possible to port your Voip.ms Voice DID Numbers and Numbers from other Providers into our Virtual Fax service. For numbers from other providers, you can find this option under the ''DID Portability'' section of the Customer Portal. For Voip.ms numbers, you can request an internal port by sending an email to our LNP department at ports@voip.ms. The porting fee is $15 per number for both options.'''
* '''The Service can only be used to send Faxes to Canadian and U.S. Numbers at this time. We also cannot guarantee that International will be properly received.'''

== Cost and Rates ==

Setup Fee: $0.00 (Currently Free)

Monthly Fee: $1.99

Per Minute Fee: $0.0290 (2.9 Cents)

[[Calls Cost|Billing Increment]] : 6 seconds

If you are wanting to calculate how much this could cost, per page faxed, converted into the time of per min charges you will want to take into consideration that this depends a lot on the destination fax speed and the content (How much of the page has something other than white) being faxed. Unfortunately there are no definite ways in calculating costs. If you are sending just text, a page can go from 30 secs up to 1 min. If you are faxing multiple page documents this will go faster since you are not having to handshake or negotiate with the far side's fax machine for each page. This can result in 1.5 cents to 3 cents on a 1-2 page fax. Since we also increment the charge to every 6 seconds you are saving even more money since we do not charge you a full min for partial minute usages. You can usually get around 1-2 pages faxed per minute at 2.9 cents and $2 a month which is much less than most other Electronic Fax services being offered at this time.

== Current Limitations ==

*During the Beta test, each DID Number can only send 100 messages per day. This limit can be raised upon request and verification.

*The files sent per message cannot exceed 25mb

== Virtual Fax DID Number ==

Virtual Fax works specifically with Fax Numbers only acquired from the VoIP.ms Customer Portal or numbers ported in specifically as Fax enabled. There are '''local US and Canadian numbers''' available for order. You can order a Fax DID Number from your portal >> DID Numbers >> Order DID >> Fax Numbers. You can select the desired region and a random number from the chosen area code will be assigned to you.

[[File:FaxorderDID2.jpg|700px|]]

You can also port in a number you currently own with another provider. This process can be started from the Customer Portal at DID Numbers >> DID Portability >> Porting Fax Numbers

[[File:FaxPortability.png|700px]]

== Send a Fax ==

To start using the service you will need to head to your Customer Portal >> DID Numbers >> Virtual Fax. From the Home Page you can select ‘Send Fax’. There you will see:
*Fax Number or Contact Name: This is where you will put the destination number. You can start typing a name or a number from your Phone book and it will become available.
*From Name: Here you will put the name to send in the Fax header.
*From Number: Select the Fax DID number from which you will send your Fax.
*File: Choose a file to send as a Fax. The file must be in pdf, txt, jpg, gif, png or tif
-'''IMPORTANT: The maximum file size is 25MB'''
*Station ID: This will be the station ID you set for the header of the Fax message. It could be a specified post if your location has several stations, such as Reception, Main Office, Accounting PC, etc.
*Send Email: If selected, an email will be sent to the specified address to confirm the Fax has been sent successfully or to advise of a failed attempt.

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

== My Faxes ==
In this section of the Virtual Fax menu you will be able to view your Inbound and Outbound Faxes. You may select a date range and choose the folder you would like to view. Click 'Get My Faxes' to view your selection. You can view the Status of each Fax and select from several Actions. You can select to View the Fax directly, Download the Fax, Email the Fax to an address of your choosing or alter the location of the Fax by moving it to another folder.

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

== My Folders ==

In the 'My Folders' section you can create folders by typing in the folder name of your choosing under 'New Folder' and clicking 'Create'.
You will have an overview of your Folders, see the date they were Created, the amount of Faxes in each Folder and be able to Edit the Folder or Delete it.
Any Faxes contained in a created folder will revert back to either the INBOX or SENT folder if the created folder is deleted.

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

== My Fax Numbers ==
In the 'My Fax Numbers' section you will see your Fax DID numbers and Description, the Options that have been enabled for each number, the Email address if one has been configured along with the URL if configured in the URL Callback section. You can Edit the number from the 'Actions' section or choose to Delete it. When editing you will have the option to set an Email Address to receive a notification when a new Fax is received (you can also select to have the PDF file attached in the email) and set a URL Callback (you can also enable URL Callback Retry).

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

== Email to Fax==

This feature allows you to send a Fax message using your email account.

How to send a Fax message using your email account:
*Use the email account you provided when enabling the Email to Fax service.
*Send the email to fax@voip.ms
*In the subject field type the Destination Fax Number (Example: 5148000000).
*Attach the document you wish to send to the email message. VoIP.ms supports the following formats: pdf, txt, jpg, gif, png, tif.
*Send the email.
**'''NOTE: Only the file attached will be faxed, the body of the e-mail won't be transmitted'''

'''Security Code and From Number:'''

If Security Code is enabled, you need to add a dot (.) and the Security Code after the Destination Fax Number (Example: 5148000000.Az09).

If you have more than one Fax number, you could change the From Number by adding a dot (.) and the From Number you'd like to use after the Destination Fax Number and the Security Code (Example: 5148000000.Az09.2268280000).

If you have not enabled the Security Code and want to change the From Number, you can add a dot (.) and the From Number after the Destination Fax Number (Example: 5148000000.2268280000).

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

Porting a Canadian Number

2016-06-28T19:33:11Z

Jeff:

You can port one or multiple numbers within one order. Please note that in order to port multiple numbers in one order, all numbers must have the same customer information with the losing provider. The fee remains the same at $10 per number. You can not mix toll-free numbers and local numbers on the same order. Same goes with Canadian and US Numbers. You'll need to start a new request if the information is somehow different.

'''Please note:''' The process usually takes 1 week.

In the following lines you will find all the necessary steps to port an existing Canadian number from another provider to VoIP.MS

Before starting the port process, it is very important to verify if your number can be ported to the VoIP.MS network, please confirm this at http://wiki.voip.ms/article/Porting_a_Number#Check_Availability

'''Step 1. '''

First you have to confirm the order type of your port process. In this case will be a Canadian number.

[[File:Lnpcanadastep1.JPG|600px]]

'''Step 2.'''

Now you should enter the number(s) you would like to port. If you have more than 1 number with the losing provider but you only want to port one or some of them, you should check on '''YES''' at the "'''Partial Port'''" section and then type the list of services and/or numbers that will remain with the current carrier and what is to be done with them (disconnect them, keep them as it is, etc.)
*Please note that some carriers do not allow this and demand that what is left with them must be either cancelled or ported, please verify this with them before you continue.
If you want to port all of them, you have to check on '''NO''' at "'''Partial Port'''" and you should enter all the numbers at "DID Number(s) to port" .

When this has been done, you will need to specify if the numbers are for a residence or a business use.

If you are going to port a mobile number you have to provide the PIN Number, IMEI and SIM Card if used. (If the number is not a cell phone you can leave these fields blank)

[[File:Step_2.jpg|600px]]

'''Step 3.'''

Now you should enter all the required information. It must be exactly as it appears on the CSR (Customer Service Record) of the losing carrier. Please note that your Customer Service Record can differ from your billing address. If you are not sure, please contact your current service provider to get that information since this is the most important information in order to avoid rejections from the losing provider and have a smooth port.

If you are planning to port a number under a different name than yours to a voip.ms account, you just have to fill the information as it appears on the losing provider´s CSR.
It does not matter if the name on the losing carrier's account does not match with the information at VoIP.ms.

Statement Name Field is for Business Numbers Only and please enter your Business Name or leave blank.

[[File:portstep3.JPG|600px]]

'''Step 4.'''

You must enter the Service Provider/Carrier Information. What is needed is your Customer Account number and Service Provider Name.

[[File:portstep4.JPG|600px]]

'''Step 5.'''

What is needed here is to upload a scan of your latest signed invoice from your losing provider. Please place your signature on a Blank part where there are no elements interfering with it. If the provider of the number does not provide an invoice, you could take a screen shot from the losing provider´s your Portal where it shows your number and account information.

If you would like to include any comment or additional information to this port. You can do it at the Notes/Comments field.

[[File:portstep5.JPG|600px]]

'''Step 6.'''

The system will show you the number(s) are going to be ported, the porting fee per number and the Total amount that will be debited from your balance. Please confirm that all information is correct.

After this, the port process will be completed and your request will be sent to the LNP Department on VoIP.ms

LNP Department will process your request and they will email you either requesting more information if needed or just to confirm everything is correct and will give you a FOC Date for your port as soon as the losing carrier confirms this information.

[[File:Portstep6can.JPG|600px]]

Please note that some carriers can take some hours to completely update their routes when a number has been ported and this could cause random situations with your number at your FOC Date.

Please contact the support department if a ported-in number is still unreachable from some providers after 24 hours since you were advised that the port had completed.

Thank you for your understanding.

[[category:guides]]

Cisco SPA525G

2016-04-08T18:16:16Z

Jeff:

[[File:525.jpg]]

The Cisco SPA525G 5-line IP Phone with Color Display is a full-featured VoIP (Voice
over Internet Protocol) phone that provide voice communication over an IP
network. It provides traditional features, such as call forwarding, redialing, speed
dialing, transferring calls, conference calling, and accessing voice mail. Calls can
be made or received with a handset, headset or speaker.

Your Cisco IP Phone provides a web interface for the phone user that allows you to
configure some features of your phone by using a web browser.

This article will guide you through the steps for basic configuration to make it work with VoIP.ms

----
 
'''Step 1'''
 
'''''Get the IP address of your phone'''''

a. Press Setup. 
b. Select to Status > Network Status. 
c. Scroll to view IP Address. This is the IP address of your phone. 

You should now have a number which is similar to 192.168.xxx.xxx

'''Step 2''' 
'''''Logging in to the Phone Web User Interface'''''

*On your PC, open a Web browser window. Your PC must be on the same subnetwork as the phone.
*Enter the IP address in the browser address bar.

You will now see this screen: 

[[File:525 1.gif|center]]
 
*Click on the '''"Admin Login"''' button near the top right side of the screen, then click on the '''"Ext 1"''' tab.

[[File:525 2.gif|center]]

'''Step 3'''
 
'''''Configure with your VoIP.ms account'''''

Find the following fields on the '''"Ext"''' tab and configure accordingly.

'''Nat Keep Alive:''' Yes 
'''Nat Mapping/Traversal:''' Yes 
'''Proxy:''' atlanta.voip.ms (one of the multiple VoIP.ms servers, you can choose the one closer to your location.) 
'''Register Expires:''' 300 
'''Display Name:''' John Smith (Replace with your name or company name) 
'''User ID:''' 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, e.g. 123456 or 123456_sub) 
'''Password:''' ******** (SIP Account Password) 
'''Use DNS SRV:''' NO 
'''DNS SRV Auto Prefix:''' NO 

[[File:cisco-spa525g.gif|center]]

*If a second extension is needed, click on '''"Ext 2"''' and repeat Step 3. Please make sure to increment the SIP port by one. For example, Ext 1 SIP port: 5060; Ext 2 SIP port: 5061. Make sure you also click on Phone Tab to route the Extensions to the proper lines.

'''Step 4'''
 

'''''Configure your dial plan'''''

This step can be considered optional however this is a dial plan that is optimized to work with VoIP.ms service.

Find the dial plan section of your line and enter the following string:

'''(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.)'''

'''If done properly, after completing all these steps your phone will now be ready to place and receive calls!'''

== '''Documentation:''' ==

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

[http://www.cisco.com/en/US/docs/voice_ip_comm/csbpipp/ip_phones/user/guide/525g_sip_user_guide_source/spa525_sip_user.pdf User´s Manual] 

keywords: SPA525

Server Realms

2016-02-16T18:38:11Z

Jeff:

==Some Devices and PBXs require the actual Realm for a server to be setup in its configurations.==

Here is a Listing of our Server Realms which can differ from the Domain Name for the server:

amsterdam.voip.ms:

realm = amsterdam.voip.ms''

atlanta.voip.ms:

realm = atlanta.voip.ms

atlanta2.voip.ms:

realm = atlanta2.voip.ms

chicago.voip.ms:

realm = chicago.voip.ms

chicago2.voip.ms:

realm = chicago2.voip.ms

chicago3.voip.ms:

realm = chicago3.voip.ms

chicago4.voip.ms:

realm = chicago4.voip.ms

dallas.voip.ms:

realm = dallasnew1.voip.ms

dallas2.voip.ms:

realm = dallasnew2.voip.ms

denver.voip.ms:

realm = denver.voip.ms

denver2.voip.ms:

realm = denver2.voip.ms

houston.voip.ms:

realm = houston1.voip.ms

houston2.voip.ms:

realm = houstonnew2.voip.ms

london.voip.ms:

realm = london1.voip.ms

losangeles.voip.ms:

realm = losangeles.voip.ms

losangeles2.voip.ms:

realm = losangeles2.voip.ms

melbourne.voip.ms:

realm = melbourne.voip.ms

montreal.voip.ms:

realm = montreal.voip.ms

montreal2.voip.ms:

realm = montreal2.voip.ms

montreal3.voip.ms:

realm = montreal3.voip.ms

montreal4.voip.ms:

realm = montreal4.voip.ms

newyork.voip.ms:

realm = newyork.voip.ms

newyork2.voip.ms:

realm = newyork2.voip.ms

newyork3.voip.ms:

realm = newyork3.voip.ms

newyork4.voip.ms:

realm = newyork4.voip.ms

paris.voip.ms:

realm = paris.voip.ms

sanjose.voip.ms:

realm = sanjose.voip.ms

sanjose2.voip.ms:

realm = sanjose.voip.ms

seattle.voip.ms:

realm = seattle.voip.ms

seattle2.voip.ms:

realm = seattle2.voip.ms

seattle3.voip.ms:

realm = seattle3.voip.ms

tampa.voip.ms:

realm = tampanew1.voip.ms

tampa2.voip.ms:

realm = tampanew2.voip.ms

toronto.voip.ms:

realm = toronto1.voip.ms

toronto2.voip.ms:

realm = toronto2.voip.ms

toronto3.voip.ms:

realm = toronto3.voip.ms

toronto4.voip.ms:

realm = toronto4.voip.ms

vancouver.voip.ms:

realm = vancouver.voip.ms

vancouver2.voip.ms:

realm = vancouver2.voip.ms

washington.voip.ms:

realm = washington.voip.ms

washington2.voip.ms:

realm = washington2.voip.ms

ws.voip.ms:

realm = ws.voip.ms

Cisco IOS

2016-02-16T18:32:19Z

Jeff: /* SIP Trunk (Username/Password Authentication) */

===SIP Trunk (Username/Password Authentication)===
For the configuration below to work, you must have DNS name lookups properly configured on your router.
The example below is based on IOS 15.1(3)T. Minor adjustments may be necessary for ealier IOS revisions.
This example uses newyork4.voip.ms as a primary route.

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.

configure terminal

voice service voip
ip address trusted list
ipv4 107.6.67.238 !'''Current IP address for newyork4.voip.ms at the time of this writing.'''
ip address trusted call-block cause not-in-cug
gcid
clid substitute name
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
e911
transport switch udp tcp
asserted-id ppi
localhost dns:'''dns.name.of.your.device'''
midcall-signaling passthru
no call service stop

sip-ua
credentials username '''your_account''' password 0 '''your_password''' realm '''newyork4.voip.ms'''
authentication username '''your_account''' password 0 '''your_password''' realm '''newyork4.voip.ms'''
registrar 1 dns:newyork4.voip.ms expires 300 !'''Pick your preferred server'''

!'''This dial peer will match all incoming calls for an specific DID'''
dial-peer voice 1 voip
huntstop
destination-pattern ########## !'''Switch the # with your DID Number'''
session protocol sipv2
session target ipv4:192.168.X.X !'''Your Call Manager IP Address'''
dtmf-relay cisco-rtp rtp-nte
codec g711ulaw
no vad

!'''This dial peer is for outgoing calls'''
dial-peer voice 2 voip
destination-pattern [2-9]..[2-9]......
session protocol sipv2
session target ipv4:107.6.67.238 !'''Your preferred server's IP address'''
no voice-class sip early-offer forced
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad

!'''This dial peer is for outgoing calls with the 1 prefix.'''
dial-peer voice 3 voip
destination-pattern 1[2-9]..[2-9]......
session protocol sipv2
session target ipv4:107.6.67.238 !'''Your preferred server's IP address'''
no voice-class sip early-offer forced
dtmf-relay rtp-nte cisco-rtp sip-kpml sip-notify
codec g711ulaw
no vad

!'''Incoming Dial-Peer'''
dial-peer voice 4 voip
session protocol sipv2
session target ipv4:107.6.67.238 !'''Your preferred server's IP address'''
incoming called-number .
dtmf-relay cisco-rtp rtp-nte
codec g711ulaw

Choosing Server

2016-02-16T18:31:52Z

Jeff:

[[File:Serverlocation3.png]]

= Choosing a Server =

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

=== IPs ===

*Amsterdam, NL ('''amsterdam.voip.ms''') 37.58.88.242
*Atlanta 1, GA ('''atlanta.voip.ms''') 174.34.146.162
*Atlanta 2, GA ('''atlanta2.voip.ms''') 72.9.246.170
*Chicago 1, IL ('''chicago.voip.ms''') 208.100.39.52
*Chicago 2, IL ('''chicago2.voip.ms''') 208.100.39.53
*Chicago 3, IL ('''chicago3.voip.ms''') 208.100.39.54
*Chicago 4, IL ('''chicago4.voip.ms''') 208.100.39.55
*Dallas, TX ('''dallas.voip.ms''') 158.85.149.162
*Dallas 2, TX ('''dallas2.voip.ms''') 158.85.149.163
*Denver 1, CO ('''denver.voip.ms''') 173.248.161.90
*Denver 2, CO ('''denver2.voip.ms''') 173.248.159.210
*Houston, TX ('''houston.voip.ms''') 173.193.85.18
*Houston 2, TX ('''houston2.voip.ms''') 173.193.85.19
*London, UK ('''london.voip.ms''') 159.8.157.212
*Los Angeles 1, CA ('''losangeles.voip.ms''') 96.44.149.186
*Los Angeles 2, CA ('''losangeles2.voip.ms''') 96.44.149.202
*Melbourne, AU ('''melbourne.voip.ms''') 168.1.73.84
*Montreal 1, QC ('''montreal.voip.ms''') 67.205.74.184
*Montreal 2, QC ('''montreal2.voip.ms''') 67.205.74.187
*Montreal 3, QC ('''montreal3.voip.ms''') 72.55.168.18
*Montreal 4, QC ('''montreal4.voip.ms''') 67.205.74.179
*New York 1, NY ('''newyork.voip.ms''') 74.63.41.218
*New York 2, NY ('''newyork2.voip.ms''') 107.6.67.236
*New York 3, NY ('''newyork3.voip.ms''') 107.6.67.237
*New York 4, NY ('''newyork4.voip.ms''') 107.6.67.238
*Paris, FR ('''paris.voip.ms''') 159.8.85.180
*San Jose, CA ('''sanjose.voip.ms''') 23.246.247.146
*San Jose 2, CA ('''sanjose2.voip.ms''') 23.246.247.147
*Seattle 1, WA ('''seattle.voip.ms''') 50.23.160.50
*Seattle 2, WA ('''seattle2.voip.ms''') 50.23.160.51
*Seattle 3, WA ('''seattle3.voip.ms''') 50.23.160.52
*Tampa, FL ('''tampa.voip.ms''') 162.254.144.173
*Tampa 2, FL ('''tampa2.voip.ms''') 162.254.144.176
*Toronto 1, ON ('''toronto.voip.ms''') 158.85.70.148
*Toronto 2, ON ('''toronto2.voip.ms''') 158.85.70.149
*Toronto 3, ON ('''toronto3.voip.ms''') 158.85.70.150
*Toronto 4, ON ('''toronto4.voip.ms''') 158.85.70.151
*Vancouver 1, BC ('''vancouver.voip.ms''') 162.213.157.82
*Vancouver 2, BC ('''vancouver2.voip.ms''') 162.213.157.117
*Washington 1, DC ('''washington.voip.ms''') 208.43.234.226
*Washington 2, DC ('''washington2.voip.ms''') 208.43.234.227

===Server Realms===

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

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

= What is a Ping? =

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

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

== How does the ping work? ==

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

==How to ping on a PC==

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

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

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

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

[[File:Ping.gif]]

==How to ping on a Mac Computer==

1- Click on Finder in the dock.

2- Click on Applications.

3- Click on Utilities.

4- Double-click on Network Utility. *

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

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

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

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

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

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

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

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

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

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

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

Output:

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

===Bash Script To Handle The Mac Ping Output Format (User Submitted) ===
pingVoipMS.sh script

<pre>
#!/bin/bash
# Ping several servers and display Latency, Jitter and Packet Loss
# Usage: [-c <count>] [-i <wait time>] [<server list file>]
#
# The optional text file should be formatted with one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
# If no args are supplied, this script will scrape a ping server list from voip.ms

DNLD_DIR="/tmp"; SERVER_LIST="voip_ping_hosts.txt"; PING_LIST="ping_result.txt"; USER_FILE=""
COUNT=3; INTERVAL=5

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

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

shift $(($OPTIND -1))

# Move the last arg (server list) to $1 and validate
[[ -n $1 && !(-f $1 && -r $1) ]] && { echo "\"$1\" file does not exist or is not readable"; exit 1; }
[[ -n $1 && -f $1 && -r $1 ]] && USER_FILE="$1"

if [[ -n $USER_FILE ]]
then
grep -v '^\s*#' $USER_FILE | awk NF > $DNLD_DIR/$SERVER_LIST
else
# N.B. The script looks for the html boldface tags inside a bracket
# If the website alters and the parse fails, manually create the list and
# supply as a script arg (or perhaps update the parsing to work again :)
curl --silent http://wiki.voip.ms/article/Choosing_Server | \
grep '([[:alpha:]]*[[:alnum:]]\.voip\.ms)' | \
tr "<>" " " | awk '{print $(NF-3)}' > $DNLD_DIR/$SERVER_LIST
fi

echo "PING will send $COUNT packet(s) with a wait of $INTERVAL sec(s) between each packet"
echo "Change the PING options by invoking this script with -c and/or -i, default \"-c 3 -i 5\""
echo "================================================================"
printf "%-20s %-18s %7s %8s %6s\n" "VoIP Server" "IP Address" "Latency" "Jitter" "Loss"
echo "================================================================"

while read myLn
do
ping -c $COUNT -i $INTERVAL -q $myLn | awk '\
/^PING / {myH=$2}
/^PING / {
IP = substr($3,2,15)
split(IP,myIP,")") }
/packet loss/ {myPL=$7}
/min\/avg\/max/ {
split($4,myS,"/")
printf("%-20s %-18s %7.3f %8.3f %6s\n",
myH, myIP[1], myS[2], myS[4], myPL) }
' | tee -a $DNLD_DIR/$PING_LIST
done < $DNLD_DIR/$SERVER_LIST

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

= Latency and it's importance =

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

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

[[category:guides]]

Server Realms

2016-02-16T18:27:39Z

Jeff: /* Some Devices and PBXs require the actual Realm for a server to be setup in its configurations. */

==Some Devices and PBXs require the actual Realm for a server to be setup in its configurations.==

Here is a Listing of our Server Realms which can differ from the Domain Name for the server:

amsterdam.voip.ms:

realm = amsterdam.voip.ms''

atlanta.voip.ms:

atlanta1.voip.ms:

realm = atlanta.voip.ms

atlanta2.voip.ms:

atlantanew2.voip.ms:

realm = atlanta2.voip.ms

chicago.voip.ms:

chicago1.voip.ms:

chicagonew1.voip.ms:

realm = chicago.voip.ms

chicago2.voip.ms:

chicagonew2.voip.ms:

realm = chicago2.voip.ms

chicago3.voip.ms:

chicagonew3.voip.ms:

realm = chicago3.voip.ms

chicago4.voip.ms:

chicagonew4.voip.ms:

realm = chicago4.voip.ms

dallas.voip.ms:

dallas1.voip.ms:

dallasnew1.voip.ms:

realm = dallasnew1.voip.ms

dallas2.voip.ms:

dallasnew2.voip.ms:

realm = dallasnew2.voip.ms

denver.voip.ms:

denver1.voip.ms:

realm = denver.voip.ms

denver2.voip.ms:

realm = denver2.voip.ms

houston.voip.ms:

houston1.voip.ms:

realm = houston1.voip.ms

houston2.voip.ms:

realm = houstonnew2.voip.ms

london.voip.ms:

london1.voip.ms:

realm = london1.voip.ms

losangeles.voip.ms:

losangeles1.voip.ms:

losangelesnew1.voip.ms:

realm = losangeles.voip.ms

losangeles2.voip.ms:

losangelesnew2.voip.ms:

realm = losangeles2.voip.ms

melbourne.voip.ms:

melbourne1.voip.ms:

realm = melbourne.voip.ms

montreal.voip.ms:

montreal1.voip.ms:

realm = montreal.voip.ms

montreal2.voip.ms:

realm = montreal2.voip.ms

montreal3.voip.ms:

realm = montreal3.voip.ms

montreal4.voip.ms:

realm = montreal4.voip.ms

newyork.voip.ms:

newyork1.voip.ms:

realm = newyork.voip.ms

newyork2.voip.ms:

realm = newyork2.voip.ms

newyork3.voip.ms:

realm = newyork3.voip.ms

newyork4.voip.ms:

realm = newyork4.voip.ms

paris.voip.ms:

paris1.voip.ms:

realm = paris.voip.ms

sanjose.voip.ms:

sanjose1.voip.ms:

realm = sanjose.voip.ms

sanjose2.voip.ms:

realm = sanjose.voip.ms

seattle.voip.ms:

seattle1.voip.ms:

seattlenew1.voip.ms:

realm = seattle.voip.ms

seattle2.voip.ms:

seattlenew2.voip.ms:

realm = seattle2.voip.ms

seattle3.voip.ms:

seattlenew3.voip.ms:

realm = seattle3.voip.ms

tampa.voip.ms:

tampa1.voip.ms:

tampanew1.voip.ms:

realm = tampanew1.voip.ms

tampa2.voip.ms:

tampanew2.voip.ms:

realm = tampanew2.voip.ms

toronto.voip.ms:

toronto1.voip.ms:

realm = toronto1.voip.ms

toronto2.voip.ms:

realm = toronto2.voip.ms

toronto3.voip.ms:

realm = toronto3.voip.ms

toronto4.voip.ms:

realm = toronto4.voip.ms

vancouver.voip.ms:

vancouver1.voip.ms:

realm = vancouver.voip.ms

vancouver2.voip.ms:

realm = vancouver2.voip.ms

washington.voip.ms:

washington1.voip.ms:

realm = washington.voip.ms

washington2.voip.ms:

realm = washington2.voip.ms

ws.voip.ms:

realm = ws.voip.ms

Server Realms

2016-02-16T18:23:56Z

Jeff:

==Some Devices and PBXs require the actual Realm for a server to be setup in its configurations.==

Here is a Listing of our Server Realms which can differ from the DNS Name for the server:

amsterdam.voip.ms:

realm = amsterdam.voip.ms''

atlanta.voip.ms:

atlanta1.voip.ms:

realm = atlanta.voip.ms

atlanta2.voip.ms:

atlantanew2.voip.ms:

realm = atlanta2.voip.ms

chicago.voip.ms:

chicago1.voip.ms:

chicagonew1.voip.ms:

realm = chicago.voip.ms

chicago2.voip.ms:

chicagonew2.voip.ms:

realm = chicago2.voip.ms

chicago3.voip.ms:

chicagonew3.voip.ms:

realm = chicago3.voip.ms

chicago4.voip.ms:

chicagonew4.voip.ms:

realm = chicago4.voip.ms

dallas.voip.ms:

dallas1.voip.ms:

dallasnew1.voip.ms:

realm = dallasnew1.voip.ms

dallas2.voip.ms:

dallasnew2.voip.ms:

realm = dallasnew2.voip.ms

denver.voip.ms:

denver1.voip.ms:

realm = denver.voip.ms

denver2.voip.ms:

realm = denver2.voip.ms

houston.voip.ms:

houston1.voip.ms:

realm = houston1.voip.ms

houston2.voip.ms:

realm = houstonnew2.voip.ms

london.voip.ms:

london1.voip.ms:

realm = london1.voip.ms

losangeles.voip.ms:

losangeles1.voip.ms:

losangelesnew1.voip.ms:

realm = losangeles.voip.ms

losangeles2.voip.ms:

losangelesnew2.voip.ms:

realm = losangeles2.voip.ms

melbourne.voip.ms:

melbourne1.voip.ms:

realm = melbourne.voip.ms

montreal.voip.ms:

montreal1.voip.ms:

realm = montreal.voip.ms

montreal2.voip.ms:

realm = montreal2.voip.ms

montreal3.voip.ms:

realm = montreal3.voip.ms

montreal4.voip.ms:

realm = montreal4.voip.ms

newyork.voip.ms:

newyork1.voip.ms:

realm = newyork.voip.ms

newyork2.voip.ms:

realm = newyork2.voip.ms

newyork3.voip.ms:

realm = newyork3.voip.ms

newyork4.voip.ms:

realm = newyork4.voip.ms

paris.voip.ms:

paris1.voip.ms:

realm = paris.voip.ms

sanjose.voip.ms:

sanjose1.voip.ms:

realm = sanjose.voip.ms

sanjose2.voip.ms:

realm = sanjose.voip.ms

seattle.voip.ms:

seattle1.voip.ms:

seattlenew1.voip.ms:

realm = seattle.voip.ms

seattle2.voip.ms:

seattlenew2.voip.ms:

realm = seattle2.voip.ms

seattle3.voip.ms:

seattlenew3.voip.ms:

realm = seattle3.voip.ms

tampa.voip.ms:

tampa1.voip.ms:

tampanew1.voip.ms:

realm = tampanew1.voip.ms

tampa2.voip.ms:

tampanew2.voip.ms:

realm = tampanew2.voip.ms

toronto.voip.ms:

toronto1.voip.ms:

realm = toronto1.voip.ms

toronto2.voip.ms:

realm = toronto2.voip.ms

toronto3.voip.ms:

realm = toronto3.voip.ms

toronto4.voip.ms:

realm = toronto4.voip.ms

vancouver.voip.ms:

vancouver1.voip.ms:

realm = vancouver.voip.ms

vancouver2.voip.ms:

realm = vancouver2.voip.ms

washington.voip.ms:

washington1.voip.ms:

realm = washington.voip.ms

washington2.voip.ms:

realm = washington2.voip.ms

ws.voip.ms:

realm = ws.voip.ms

Server Realms

2016-02-16T18:21:54Z

Jeff: Created page with "'''Some Devices and PBXs require the actual Realm for a server to be setup in its configurations.''' Here is a Listing of our Server Realms which can differ from the DNS Name fo..."

'''Some Devices and PBXs require the actual Realm for a server to be setup in its configurations.'''

Here is a Listing of our Server Realms which can differ from the DNS Name for the server:

amsterdam.voip.ms:

realm = amsterdam.voip.ms''

atlanta.voip.ms:

atlanta1.voip.ms:

realm = atlanta.voip.ms

atlanta2.voip.ms:

atlantanew2.voip.ms:

realm = atlanta2.voip.ms

chicago.voip.ms:

chicago1.voip.ms:

chicagonew1.voip.ms:

realm = chicago.voip.ms

chicago2.voip.ms:

chicagonew2.voip.ms:

realm = chicago2.voip.ms

chicago3.voip.ms:

chicagonew3.voip.ms:

realm = chicago3.voip.ms

chicago4.voip.ms:

chicagonew4.voip.ms:

realm = chicago4.voip.ms

dallas.voip.ms:

dallas1.voip.ms:

dallasnew1.voip.ms:

realm = dallasnew1.voip.ms

dallas2.voip.ms:

dallasnew2.voip.ms:

realm = dallasnew2.voip.ms

denver.voip.ms:

denver1.voip.ms:

realm = denver.voip.ms

denver2.voip.ms:

realm = denver2.voip.ms

houston.voip.ms:

houston1.voip.ms:

realm = houston1.voip.ms

houston2.voip.ms:

realm = houstonnew2.voip.ms

london.voip.ms:

london1.voip.ms:

realm = london1.voip.ms

losangeles.voip.ms:

losangeles1.voip.ms:

losangelesnew1.voip.ms:

realm = losangeles.voip.ms

losangeles2.voip.ms:

losangelesnew2.voip.ms:

realm = losangeles2.voip.ms

melbourne.voip.ms:

melbourne1.voip.ms:

realm = melbourne.voip.ms

montreal.voip.ms:

montreal1.voip.ms:

realm = montreal.voip.ms

montreal2.voip.ms:

realm = montreal2.voip.ms

montreal3.voip.ms:

realm = montreal3.voip.ms

montreal4.voip.ms:

realm = montreal4.voip.ms

newyork.voip.ms:

newyork1.voip.ms:

realm = newyork.voip.ms

newyork2.voip.ms:

realm = newyork2.voip.ms

newyork3.voip.ms:

realm = newyork3.voip.ms

newyork4.voip.ms:

realm = newyork4.voip.ms

paris.voip.ms:

paris1.voip.ms:

realm = paris.voip.ms

sanjose.voip.ms:

sanjose1.voip.ms:

realm = sanjose.voip.ms

sanjose2.voip.ms:

realm = sanjose.voip.ms

seattle.voip.ms:

seattle1.voip.ms:

seattlenew1.voip.ms:

realm = seattle.voip.ms

seattle2.voip.ms:

seattlenew2.voip.ms:

realm = seattle2.voip.ms

seattle3.voip.ms:

seattlenew3.voip.ms:

realm = seattle3.voip.ms

tampa.voip.ms:

tampa1.voip.ms:

tampanew1.voip.ms:

realm = tampanew1.voip.ms

tampa2.voip.ms:

tampanew2.voip.ms:

realm = tampanew2.voip.ms

toronto.voip.ms:

toronto1.voip.ms:

realm = toronto1.voip.ms

toronto2.voip.ms:

realm = toronto2.voip.ms

toronto3.voip.ms:

realm = toronto3.voip.ms

toronto4.voip.ms:

realm = toronto4.voip.ms

vancouver.voip.ms:

vancouver1.voip.ms:

realm = vancouver.voip.ms

vancouver2.voip.ms:

realm = vancouver2.voip.ms

washington.voip.ms:

washington1.voip.ms:

realm = washington.voip.ms

washington2.voip.ms:

realm = washington2.voip.ms

ws.voip.ms:

realm = ws.voip.ms

Troubleshooting Outgoing Calls

2016-02-15T23:14:06Z

Jeff: /* Unreachable Calls */

In this entry we are going talk about one of VoIP troubleshooting, when you try to call a number and you can not reach it:

=='''Unreachable Calls'''==

If you try to reach a number and it is Unreachable, here are some tips that you can do.

* Confirm if your account has enough balance, you can check this on '''Customer portal -> Finances -> Account balance'''. Only accounts with a balance over $0 are able to send and receive calls.

* Check if the account or sub account you are using to dial this number is registered at the moment, you can see this going to '''Customer portal -> Main menu -> Portal home -> Registration status'''. If unsure about your registration status you can also dial 4443 (free call) to reach our echo test, if you can reach it then you are registered.

* Make sure you are not blocking your CallerID. You can check in your [https://www.voip.ms/m/cdr.php CDR] if you see Anonymous for your outbound calls try Dialing *68, to disable CallerID Block, then Hangup and try another call.

If your account or sub account is not registered, you can check these links, according device or application you are using for dialing out the calls:

- For '''PBX´s servers''': http://wiki.voip.ms/article/PBXs

- For '''ATA devices''': http://wiki.voip.ms/article/Devices

- For '''Softphones''': http://wiki.voip.ms/article/Softphones

* Make sure that the number is being dialed correctly. Check Dial Mode on '''Customer Portal >> Main Menu >> Account Settings >> General.''' For Local numbers you can dial the 10 digits number or also try to use prefix 1. For international numbers you need to dial using prefixes '''00 or 011 + country code + number''' (This is if you have the default dialing mode, it can be changed between NANPA dialing mode or E164 dialing mode in your account´s restrictions, in the "General" tab).

* Verify if you are trying to reach a '''valid number''', per example, '''you can try to reach this number from a land line'''.

* It does not matter if you are trying to reach a USA/CAN or an international number, you should check if the country you are calling is allowed in your destinations, you can see this on '''customer portal -> main menu >> Account settings >> Account restrictions'''

* And, if this is an international number, make sure that your International calls are enabled for the main account or sub account being used. For your main account please check '''customer portal -> main menu -> account settings -> Account restrictions'''. For a sub account please go to '''customer portal -> Sub account -> manage sub account -> edit sub account -> international calls'''.

Important Note: Some destinations are not allowed by default, please check this on '''Customer portal -> Main menu -> Portal home -
>Allowed''' '''international destination -> click for details'''

* Try to reach this number using our 2 routes : Value and Premium. For Canadian numbers, you can change this value on '''Customer portal -> Main menu -> Account settings -> Account routing'''. For international numbers, if you are dialing it from your main account, then you can change this value on '''Customer portal -> Main menu -> Account settings -> Account routing''', or if you are dialing it from a sub account, then you can change this value on '''Customer portal -> Sub account -> Manage sub accounts -> Edit sub accounts -> International route.'''

Important Note: Some numbers can not be reached from Voip.ms system, per example international premium numbers, international toll free
numbers which are only reachable from the country they belong. Go to '''Customer Portal >> Rates >> Check Rates Online''', there you can
paste the number you are dialing to check the description.

* If you still experiencing issues, please contact Voip.ms technical support staff '''via ticket( sending an email to support@voip.ms ) or via live chat''', specifying in detail the number called, the result (long silence, busy, etc) and on the case of Canada, the route that was used (value/premium). If possible copy-paste the entries of your VoIP.ms Call detail record on the ticket. The more detail received by support, the more quickly they will be able to help.

[[category:guides]]

Sip Scanner Ghost Calls

2016-02-10T18:50:40Z

Jeff: Created page with "__TOC__ Some people from time to time experience calls on their IP phones from unknown numbers or extensions and when they pick up they just hear silence. This is unfortunately ..."

__TOC__

Some people from time to time experience calls on their IP phones from unknown numbers or extensions and when they pick up they just hear silence. This is unfortunately a well known problem in regular Telephone & VoIP and has nothing to do with the service provider. We call these types of calls SIP Scanner Ghost Calls and besides being extremely annoying they don’t pose any significant risk to your phones or network. Providing you make sure the firmware on your phone is up to date.

These calls are not coming from our service but they are generated by “port scans” performed by hackers trying to find a vulnerable phone network to gain access to. They do large series of automated tests against IP addresses on the internet, to find systems that respond. The good news is that there are several ways you can prevent these types of calls.

==Change Local SIP Port==

Changing the local SIP port on your phone will make it harder for the scanners to guess the way into your device. You can try and set this to something like 5080 or 42872. The place to do this is usually in the Line/EXT config page for that device and it will say either Sip Port or Local Sip Port in most cases. By Default the SIP Port is usually set to 5060.

==Use a Firewall==
Some firewalls are able to filter these port scans from legit traffic. Look in the manual for your router/firewall to see how to do this, or contact your internet provider and ask for their assistance.

==Change Your IP==
If you don’t have a specific reason to have a static IP address, you can ask your internet provider to assign you a new IP address. This may not be a permanent solution to the problem, but it can definitely stop the calls for some time.

==Only Allow Calls from VoIP.ms Servers==
Some IP phones can disable direct calls from other devices than a specific server. This means that the phone will reject all calls that are not coming from the VoIP.ms server. The setting’s location and name varies from phone to phone, so check your manual to see if your phone supports it.

== Here are a few models that have a resolution for this issue: ==

===Cisco/Linksys SPAxxx===
Please look under the Voice>> Line/EXT # 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]]
 

===Cisco/Linksys Pap2t===
Please look under the Voice>> Line/EXT # page in your Linksys 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:RestrictSourceIP.png|800px]]

===Grandstream GXP2200===
Advanced Settings -> Call Features -> Disable Direct IP Calls

===Grandstream GXP2160===
Accounts -> Account 1 -> SIP Settings -> Security Settings -> Accept Incoming SIP from Proxy Only

===Grandstream HT50X/HT70X===
To Prevent Direct IP calls to your device and only allow calls from our service please enable the following 2 options in your FXS Port Configuration Page.

'''Check SIP User ID for incoming INVITE''' - Default is No. Check the incoming SIP User ID in Request URI. If they don’t match, the call will be rejected. If this option is enabled, the device will not be able to make direct IP calls.

'''Allow Incoming SIP Messages from SIP Proxy Only''' - Default is No. Check the incoming SIP messages. If they don’t come from the SIP
proxy, they will be rejected. If this option is enabled, the device will not be able to make direct IP calls.

===Obi 1xx/2xx===
*You can just disable (by unchecking Enable) for SP2 and OBiTALK under your Voice Tab (If you are using our service as SP1).

*You can restrict which IP addresses that can connect to your OBi. Going to "Service Providers -> ITSP Profile A -> SIP -> X_AccessList" : voip.ms_ip_address. You can see the IP address of the server you are currently using from this link: [http://wiki.voip.ms/article/Choosing_Server#IPs Server's IPs]

*You can also change your Obi Firewall Setting X_InboundCallRoute to : {(?|x|xx|xxx|xxxx|xxxxx|xxxxxx|un@@.|anon@@.):}, ph
This will only allow 7 digit or greater numbers through.

*Another alternative: OBi Interface>> Voice Services>> SP1 Service>> X_InboundCallRoute: {>('Insert your AuthUserName here'):ph}, example:

{>('100000'):ph} where 100000 is replaced with your own six digit SIP account UserID or the sub-account registered with your device.

By default, OBi devices accept calls destined for any username. The above syntax rejects calls that are not intended for whatever you have configured as AuthUserName.

===Panasonic KX-TGP 500/550===
To turn off IP Dialing function on a TGP 500/550 you go to Line 1 > Enable SSAF (SIP Source Address Filter). That should stop the random dialing.

===Polycom Phones===
Try to utilize the Incoming Signaling Validation where you would be able to add security to the phone to validating incoming network signaling in the GUI.
All of this is described in the <requestValidation/> section of the Admin Guide matching your software version.

===Yealink===

Look for these settings on the Line config page.

Allow Direct IP Call - this means the phone will respond to calls coming in to it from any IP address, to any number. Sometimes used for internal intercom systems or basic phone testing without using a PBX. Set it to disabled. This setting is found in the "Features" setting tab, "General Information" page.

Accept SIP Trust Server Only - this is whether the phone accepts calls to the correct phone number but from a different place than it is Registered to. Sometimes needed for certain SIP providers but you want to set this to enabled wherever possible so the phone only accepts calls from your service provider. This setting is found either in the "Features" tab, "General Information" page or the "Account" tab depending on the phone model or firmware version.

You can also try to add below syntaxes to your cfg template(M7 template) and auto-provision it.

1. You can try this syntax in CFG template.
---------------------------------------------------------------------------
#!version:1.0.0.1

#The x of the parameter "account.x.sip_trust_ctrl " ranges from 1 to max accounts. For example, x ranges from 1 to 6 of T28.

account.x.sip_trust_ctrl=1
------------------------------------------------------------------------------------------

When you want to enable this sip trust control for account 1, fill 1 to “account.1.sip_trust_ctrl”.
Then SIP messages from other servers will refuse by the phone.

2. If not, you can disable the “Allow IP Call” in webpage or auto-provisioning and try again.

-------------------------------------------------------------------------------------------------
#!version:1.0.0.1

#Enable or disable the phone to dial the IP address directly; 0-Disabled, 1-Enabled (default);
features.direct_ip_call_enable = 0

-------------------------------------------------------------------------------------------------

OBi 100/110

2015-12-08T17:01:32Z

Jeff: /* Settings to avoid direct phone calls to your device in the middle of the night */

[[File:OBi110-ATA.jpg|none|200px|center]]

 ''The OBi100 is a single phone port ATA adapter that supports SIP VoIP services. The OBi100 is perfect for customers who do not have a traditional phone service, yet need a similar solution and want the savings and simplicity of using a VoIP service for all their calls. To start configuring your OBi100 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.

For an OBi202 please do the following to enable the GUI Web Interface:

Dial *** from the phone connected to the OBi202
Enter 0 For Advanced
Enter 30# Check Mark from
Press 1 to Enter a New Value
Press 1# to Enable
Press 1 to Save
Hang up

If done properly, the following window should appear on your screen:
[[File:ObiLogin.png|300px|thumb|left|Authentication Window - click to enlarge]]
 

Once you get the Authentication Required window you will be prompted for a User Name and Password, the default credentials are:

'''User Name:''' admin

'''Password:''' admin

After this, you should now be able to see the OBi Web interface.

Now on the left side of the screen please find the next options and follow the next steps:

===Disabling auto-provisioning===

'''**NOTE :''' You may use this guide to configure an OBi110 as well. This is the VoIP.ms recommended configuration versus using the Obihai configuration dashboard (more on this later on this page) and you may also not find all new VoIP.ms servers on the Obihai Dahsboard. In order to make sure there will be no conflicts between this Manual configuration and the Obihai dashboard, please perform the following steps to disable auto-provisioning:

*System Management -> Auto Provisioning -> Auto Firmware Update -> Method : Disabled
*System Management -> Auto Provisioning -> ITSP Provisioning -> Method : Disabled
*System Management -> Auto Provisioning -> OBiTALK Provisioning -> Method : Disabled
*Voice Services -> OBiTALK Service -> Enable : Unchecked

Please note you must remove the check mark from the "default" column, then under "Method" please use the ''''Drop Down Selection'''' and choose '''Disabled'''.

[[File:Step1.png|450px|thumb|left|Disabling Auto Provision - click to enlarge]]
 
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) 
:'''DigiMap''': Copy the line, including parenthesis, in the Digitmap field in the ITSP Profile and replace the "555" digits in the following lines by the area code of your choice:

::Dial Plan (recommended):
(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|911|011xx.|xx.|*xx.|***xxx|4xxx|(Mipd)|[^*]@@.)

Also, some clients have been successful by dialing '''**1 and *97''' for line 1 or '''**2 and *97''' for line 2.

:*If you need to set the dial plan back to Default, you can use this:

(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|011xx.|***xxx|xx.|(Mipd)|[^*]@@.)

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

====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.

== Configuration Using OBi Dashboard ==

Besides the Manual Configuration previously explained, Obihai also provides us with their own API dashboard where you can add your device, to complete the configuration in easy steps.
Add your device to the OBiTALK service in the OBi Dashboard [http://www.obitalk.com/obinet/]. Instructions for this are included with the OBi110 and are not discussed here.

After the OBi110 is added, edit the device. You can select '''Service Provider 1''' or '''Service Provider 2''' under the '''Configure Voice Services''' heading. This will take you to a page where you can select ''voip.ms''. Follow the instructions and once you are done the configuration will be downloaded to your Obi110.

== Features Star Codes ==
Please check this link to the Star Codes that are available to activate and deactivate some of the features on your device [http://www.obihai.com/docs/OBiFeatureStarCodes.pdf OBI feature star codes]
 

==Known Issues and Resolutions==
 
=== Settings to avoid direct phone calls to your device in the middle of the night ===
 
Some customers have reported receiving calls in the middle of the night coming from "100" or "101" as callerID. These calls are directly to your device and do not pass through our servers, so we cannot filter them. However, we have some suggestions:
*You can just disable (by unchecking Enable) for SP2 and OBiTALK under your Voice Tab (If you are using our service as SP1).

*You can restrict which IP addresses that can connect to your OBi. Going to "Service Providers -> ITSP Profile A -> SIP -> X_AccessList" : voip.ms_ip_address. You can see the IP address of the server you are currently using from this link: [http://wiki.voip.ms/article/Choosing_Server#IPs Server's IPs]

*You can also change your Obi Firewall Setting X_InboundCallRoute to : {(?|x|xx|xxx|xxxx|xxxxx|xxxxxx|un@@.|anon@@.):}, ph
This will only allow 7 digit or greater numbers through.

*Another alternative: OBi Interface>> Voice Services>> SP1 Service>> X_InboundCallRoute: {>('Insert your AuthUserName here'):ph}, example:

{>('100000'):ph} where 100000 is replaced with your own six digit SIP account UserID or the sub-account registered with your device.

By default, OBi devices accept calls destined for any username. The above syntax rejects calls that are not intended for whatever you have configured as AuthUserName.

 

===10 Second Delay Reaching voip.ms Voicemail Attendant when dialing *97 or *98===

The Obi 100, 110 and 202 devices have non-configurable 'short' and 'long' delays if a dialed sequence does not match a digitmap. So you may have a 10 second delay when you dial into your voip.ms voicemail because of the built-in 'long' delay. This can be resolved in a couple of ways. Simply dial a # sign after you dial *97 or *98. Or include literals in your digitmap under the Service Provider / ITSP profile A or B / General / digitmap. Here is an example digitmap with a *97 literal included:

(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|*xx.|'*97'|(Mipd)|[^*#]@@.)

The literal in the example is '*97'. You could also add a literal for '*98'.

Then when you dial *97, the device immediately sends it instead of waiting 10 seconds.

Read more on digitmaps under the topic Digit Map Configuration in the Obi Device Admin Guide.

=== Call Drops ===

If you experience random call drops while in the middle of a call or if the person you talk to remains silent for over a minute (60 seconds by default), OBi will hang up the call. Please go here and check and increase the following setting (Physical interface -> LINE port -> DetectFarEndLongSilence / SilenceTimeThreshold)

=== Enable Message Waiting Indicator MWI ===

To enable MWI please refer to the following section of the OBI web page:

Voice Services -> SP1 Service -> Calling Features ->

'''MWIEnable''' - Uncheck the box at the far right, to be able to check the box at the left, this enables the option.

'''X_VMWIEnable''' - Uncheck the box at the far right, to be able to check the box at the left, this enables the option.

'''MessageWaiting''' - Mark the checkbox

After these steps, the MWI should be active and working.

'''If you are trying to place an outbound call and get a recorded message ¨There is no service to complete your call¨ Please do the following to resolve this.'''
In Your OBi Device please go to Physical Interfaces >> PHONE Port which by default it is PSTN and it needs to be changed to Trunk Group 1

=== Using the OBi Network ===

You can use your OBi device to make calls directly to other OBi devices "''The OBi comes out of the box ready to make FREE calls to other OBi endpoints using the OBiTALK network. Dialing **9 + obi account number will use the OBiTALK feature and does not place calls to regular numbers nor use our network. ''" (you can get more information about [http://www.obihai.com/features-and-set-up here]), be aware that those calls will not pass through our network. If you need assistance with that feature, please contact [http://www.obihai.com/request-support OBI's support].
 
=== Can not dial *98 even if is on your DigiMap ===
By default your OBi device uses *98 as Blind transfer code. If you want to be able to dial *98 from your device, you should change this code. You can achieve this in the settings of your device at: ''Star Codes >> Star code profile (A/B)'', unmark the "default" box and change *98 for something else (like *99)

[[File:Step6.png|550px|thumb|left|Changing *98 default code - click to enlarge]]
 

=== Using the Phone book of your customer portal ===

If you plan on using the Phone Book in your Customer Portal and Speed Dial *75. Please log into your OBi and change the built-in speed dial code from *75 in the device to something else.

=== An additional note regarding outgoing calls===

In at least one instance it was necessary to specify a non-default outbound calling route in the OBi110 to be able to place calls using the voip.ms service. The default setting had the OBi110 attempting to place calls using the PSTN port on the device. The relevant setting is:

'''Physical Interfaces >> PHONE Port '''
*PrimaryLine: (Select from drop-down)

[[File:ObiPhoneport.JPG|550px|thumb|left|Changing Phone Port - click to enlarge]]
 

The default is PSTN. Select SP1 Service if you only have one SIP account configured on the device. Select Trunk Group 1 to have it attempt to place calls using SP1 first, then SP2. Additional Trunk groups can be configured under Voice Services >> Gateways and Trunk Groups.

Portions of this article have been taken from [http://www.toao.net/500-mangos-guide-to-configuring-an-obi100-obi110-and-obi202-ata Mango's Guide to Configuring an OBi ATA]. Used with permission.

[[Category:Analog Telephone Adapters]]

Cisco SPA112

2015-10-07T16:31:54Z

Jeff: /* Firmware Upgrade */

[[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 both customers of 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=1.4.0&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. '''

== Configuration Details ==

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

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 initialise; even a correctly configured and installed SPA112/122 will give no power to the 'phone or no dialtone until initialisation 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. Dialling ****110# will give one address, ****210# will give the other.

=== 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.

[[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: 
'''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 submit 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.)

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

You can create your own dial plan if you need it, referring to this entry [[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.

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 PAP2's 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 the 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=1.4.0&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

[[category:Analog Telephone Adapters]]

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

Cisco SPA112

2015-10-07T16:29:58Z

Jeff:

[[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 both customers of 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=1.4.0&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. '''

== Configuration Details ==

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

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 initialise; even a correctly configured and installed SPA112/122 will give no power to the 'phone or no dialtone until initialisation 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. Dialling ****110# will give one address, ****210# will give the other.

=== 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.

[[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: 
'''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 submit 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.)

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

You can create your own dial plan if you need it, referring to this entry [[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.

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 PAP2's 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 the VoIP.ms (or other provider's) web interface.

Updated firmware is available from the Cisco site [http://software.cisco.com/download/release.html?mdfid=283998771&softwareid=282463187&release=1.3.3&relind=AVAILABLE&rellifecycle=&reltype=latest] 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

[[category:Analog Telephone Adapters]]

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

Virtual Fax

2015-09-15T15:26:10Z

Jeff:

[[File:Faxhomelogo.png|center]]

The Virtual Fax feature is used for sending and receiving a Fax (facsimile) with the VoIP.ms service using a DID number specifically dedicated to Faxing. You may obtain such a number from your Customer Portal in the Fax Numbers section under the ''Order DID(s)'' of the ''DID Numbers'' menu.
Regular voice DID numbers are not compatible with the Virtual Fax feature.

__TOC__

== Important information to know about the Virtual Fax Service ==

* '''The Virtual Fax Service is in BETA version, that means it is not fully deployed. It is important to us that you report any issues with this service by sending an email to support@voip.ms so that the developers can get involved if necessary.'''
* '''The Virtual Fax Service is only available for U.S. and Canadian DID Numbers specifically acquired from the Fax Numbers ''Order DID'' section'''
* '''It is also possible to port your Voip.ms Voice DID Numbers and Numbers from other Providers into our Virtual Fax service, you can find this option under the ''DID Portability'' section. The porting fee is $15 per number.'''
* '''The Service can only be used to send Faxes to Canadian and U.S. Numbers at this time. We also cannot guarantee that International will be properly received.'''

== Cost and Rates ==

Setup Fee: $0.00 (Currently Free)

Monthly Fee: $1.99

Per Minute Fee: $0.0290 (2.9 Cents)

== Virtual Fax DID Number ==

Virtual Fax works specifically with Fax Numbers only acquired from the VoIP.ms Customer Portal or numbers ported in specifically as Fax enabled. There are '''local US and Canadian numbers''' available for order. You can order a Fax DID Number from your portal >> DID Numbers >> Order DID >> Fax Numbers. You can select the desired region and a random number from the chosen area code will be assigned to you.

[[File:FaxorderDID2.jpg|700px|]]

You can also port in a number you currently own with another provider. This process can be started from the Customer Portal at DID Numbers >> DID Portability >> Porting Fax Numbers

[[File:FaxPortability.png|700px]]

== Send a Fax ==

To start using the service you will need to head to your Customer Portal >> DID Numbers >> Virtual Fax. From the Home Page you can select ‘Send Fax’. There you will see:
*Fax Number or Contact Name: This is where you will put the destination number. You can start typing a name or a number from your Phone book and it will become available.
*From Name: Here you will put the name to send in the Fax header.
*From Number: Select the Fax DID number from which you will send your Fax.
*File: Choose a file to send as a Fax. The file must be in pdf, txt, jpg, gif, png or tif
*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:Sendafax7.jpg]]

== My Faxes ==
In this section of the Virtual Fax menu you will be able to view your Inbound and Outbound Faxes. You may select a date range and choose the folder you would like to view. Click 'Get My Faxes' to view your selection. You can view the Status of each Fax and select from several Actions. You can select to View the Fax directly, Download the Fax, Email the Fax to an address of your choosing or alter the location of the Fax by moving it to another folder.

[[File:Myfaxes4.jpg]]

== My Folders ==

In the 'My Folders' section you can create folders by typing in the folder name of your choosing under 'New Folder' and clicking 'Create'.
You will have an overview of your Folders, see the date they were Created, the amount of Faxes in each Folder and be able to Edit the Folder or Delete it.
Any Faxes contained in a created folder will revert back to either the INBOX or SENT folder if the created folder is deleted.

[[File:Faxmyfolders3.jpg]]

== 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:Myfaxnumbers3.jpg]]

== 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.

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:Faxemail.jpg|700px|]]

Dialing Rules and Patterns

2015-09-07T16:21:51Z

Jeff: /* Dialing Rules */

= Dialing Rules and Patterns =

This article explains the difference and usage between the Dialing Rules or Dial Plans (From the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk] outgoing settings) and the Dialing Patterns (From the [http://wiki.voip.ms/article/Trixbox#Outbound_Routes Outbound routes]) in the common asterisk distro.

==Dialing Rules==

The most common dialing rule that we can find in the '''[http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk] outgoing settings''' (either SIP or IAX) is the following:

(N Represents a Number 2-9 and X Represents Any Number)

'''1+NXXNXXXXXX'''

What it does is adding the "1" to any pattern like "NXXNXXXXXX"

''It is important to understand that the rules will apply as long as the pattern exists, if it doesn't exist the rule will never apply and the call will end in a typical " This call can not be placed as dialed".''

'''For example, if you want to dial 7 digits only:'''

'''1555+NXXXXXX'''

Replace the 555 digits in the previous line for the area code of your choice.

==Useful VoIP.ms Rules==

'''4443''' - For calling Echo Test to test call connectivity to our servers and call quality.

'''4747''' - For DTMF Testing.

'''***XXX''' - To test MOH (Music on Hold) Categories.

'''*xx''' - To access Voicemail Options with our service like *97 and *98.

'''0441+NXXNXXXXXX or 0331+NXXNXXXXXX''' - Used to manually dial a Premium (0441) or a Value (0331) Canadian Route.

'''011+.''' or '''00+.''' - For International Calling.

'''044+.''' and '''033+.''' - To Manually dial (044) Premium International Routes or (033) Value International Routes. Good for Testing a call via different routes on the go.

==Dialing Patterns==

The Dialing patterns can be found in the '''[http://wiki.voip.ms/article/Trixbox#Outbound_Routes Outbound route]''', whatever you dial from any [http://wiki.voip.ms/article/Trixbox#Extensions extension] must match a dialing pattern, the most common dialing pattern found here is the following:

'''NXXNXXXXXX'''

''The important thing to understand, is that the [http://wiki.voip.ms/article/Trixbox#Outbound_Routes outbound route] will select the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunks] it will use'', however if you have multiple [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunks] with the same patterns (which is commonly used), then you will have to select the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunks] priority (which is found at the top right of the outbound route screen, as a list of the [http://wiki.voip.ms/article/Trixbox#Outbound_Routes Outbound routes] names with arrows to move up and down as priority).

Now, what happens if you have multiple [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunks] and you need to force that one [http://wiki.voip.ms/article/Trixbox#Extensions extension] does come up from an specific [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk]? It is a simple play of rules and patterns.

=How to force one [http://wiki.voip.ms/article/Trixbox#Extensions extension] through a specific [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk]=

As has been explained, the [http://wiki.voip.ms/article/Trixbox#Extensions extension] does not "choose" on which [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk] to come out, this is done by the outbound route, what we need to do is to play with the patterns from the [http://wiki.voip.ms/article/Trixbox#Outbound_Routes Outbound routes] and dialing rules for the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunks] set.

What we can chose is the [http://wiki.voip.ms/article/Trixbox#Outbound_Routes outbound route] (which contains the specific [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk]).

How?

Lets say we have trunk1 and trunk2 for this example.
Lets say also we have outbound route1 and outbound route2 for this example.
Also lets say we have the extension1.

The trunk1 and trunk2 dialing rules will be the same = 1+NXXNXXXXXX

Now on the outbound route we can determine the specific pattern that will help us to "choose" either [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk] from when dialing from an specific [[http://wiki.voip.ms/article/Trixbox#Extensions extension]].

We can set to the outbound route1 the pattern:

'''NXXNXXXXXX'''

but also (this is the trick) we can add something like X|. (being X any number of your choice)

''' 2|.'''

This means any pattern with a "2" in front will be recognized by that route and use the specific [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk], the pattern also removes the 2 so this number is not sent and the rule in the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk] 1+ remains.

In this manner we dial a regular US/CAN number (10 digits) this way from the [http://wiki.voip.ms/article/Trixbox#Extensions extensions].

'''25626846308'''

''Note the 2 before the ten digits, this will be stripped out and substitute by 1 according to the dialing rules set in the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk]. By doing this we ensure the use of the trunk1 (which is being chosen in the outbound route).''

Now, in the outbound route2 we add the patterns:

'''NXXNXXXXXX'''

'''3|.'''

This way, when dialing within the [http://wiki.voip.ms/article/Trixbox#Extensions extension] we only need to add a 3 (that will match the specific route) and use the specific [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk].

'''like 35626846308'''

The number 3 will be removed and substitute by the "1" according to the dialing rules in the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk].

Additionally you can play with the dial rules on the devices that uses the [http://wiki.voip.ms/article/Trixbox#Extensions extension], so the 2 or 3 or number chosen is sent automatically without the need of dialing.

[[category:guides]]

Dialing Rules and Patterns

2015-09-07T16:19:46Z

Jeff: /* Useful VoIP.ms Rules */

= Dialing Rules and Patterns =

This article explains the difference and usage between the Dialing Rules or Dial Plans (From the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk] outgoing settings) and the Dialing Patterns (From the [http://wiki.voip.ms/article/Trixbox#Outbound_Routes Outbound routes]) in the common asterisk distro.

==Dialing Rules==

The most common dialing rule that we can find in the '''[http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk] outgoing settings''' (either SIP or IAX) is the following:

'''1+NXXNXXXXXX'''

What it does is adding the "1" to any pattern like "NXXNXXXXXX"

''It is important to understand that the rules will apply as long as the pattern exists, if it doesn't exist the rule will never apply and the call will end in a typical " This call can not be placed as dialed".''

'''For example, if you want to dial 7 digits only:'''

'''1555+NXXXXXX'''

Replace the 555 digits in the previous line for the area code of your choice.

==Useful VoIP.ms Rules==

'''4443''' - For calling Echo Test to test call connectivity to our servers and call quality.

'''4747''' - For DTMF Testing.

'''***XXX''' - To test MOH (Music on Hold) Categories.

'''*xx''' - To access Voicemail Options with our service like *97 and *98.

'''0441+NXXNXXXXXX or 0331+NXXNXXXXXX''' - Used to manually dial a Premium (0441) or a Value (0331) Canadian Route.

'''011+.''' or '''00+.''' - For International Calling.

'''044+.''' and '''033+.''' - To Manually dial (044) Premium International Routes or (033) Value International Routes. Good for Testing a call via different routes on the go.

==Dialing Patterns==

The Dialing patterns can be found in the '''[http://wiki.voip.ms/article/Trixbox#Outbound_Routes Outbound route]''', whatever you dial from any [http://wiki.voip.ms/article/Trixbox#Extensions extension] must match a dialing pattern, the most common dialing pattern found here is the following:

'''NXXNXXXXXX'''

''The important thing to understand, is that the [http://wiki.voip.ms/article/Trixbox#Outbound_Routes outbound route] will select the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunks] it will use'', however if you have multiple [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunks] with the same patterns (which is commonly used), then you will have to select the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunks] priority (which is found at the top right of the outbound route screen, as a list of the [http://wiki.voip.ms/article/Trixbox#Outbound_Routes Outbound routes] names with arrows to move up and down as priority).

Now, what happens if you have multiple [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunks] and you need to force that one [http://wiki.voip.ms/article/Trixbox#Extensions extension] does come up from an specific [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk]? It is a simple play of rules and patterns.

=How to force one [http://wiki.voip.ms/article/Trixbox#Extensions extension] through a specific [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk]=

As has been explained, the [http://wiki.voip.ms/article/Trixbox#Extensions extension] does not "choose" on which [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk] to come out, this is done by the outbound route, what we need to do is to play with the patterns from the [http://wiki.voip.ms/article/Trixbox#Outbound_Routes Outbound routes] and dialing rules for the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunks] set.

What we can chose is the [http://wiki.voip.ms/article/Trixbox#Outbound_Routes outbound route] (which contains the specific [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk]).

How?

Lets say we have trunk1 and trunk2 for this example.
Lets say also we have outbound route1 and outbound route2 for this example.
Also lets say we have the extension1.

The trunk1 and trunk2 dialing rules will be the same = 1+NXXNXXXXXX

Now on the outbound route we can determine the specific pattern that will help us to "choose" either [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk] from when dialing from an specific [[http://wiki.voip.ms/article/Trixbox#Extensions extension]].

We can set to the outbound route1 the pattern:

'''NXXNXXXXXX'''

but also (this is the trick) we can add something like X|. (being X any number of your choice)

''' 2|.'''

This means any pattern with a "2" in front will be recognized by that route and use the specific [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk], the pattern also removes the 2 so this number is not sent and the rule in the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk] 1+ remains.

In this manner we dial a regular US/CAN number (10 digits) this way from the [http://wiki.voip.ms/article/Trixbox#Extensions extensions].

'''25626846308'''

''Note the 2 before the ten digits, this will be stripped out and substitute by 1 according to the dialing rules set in the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk]. By doing this we ensure the use of the trunk1 (which is being chosen in the outbound route).''

Now, in the outbound route2 we add the patterns:

'''NXXNXXXXXX'''

'''3|.'''

This way, when dialing within the [http://wiki.voip.ms/article/Trixbox#Extensions extension] we only need to add a 3 (that will match the specific route) and use the specific [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk].

'''like 35626846308'''

The number 3 will be removed and substitute by the "1" according to the dialing rules in the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk].

Additionally you can play with the dial rules on the devices that uses the [http://wiki.voip.ms/article/Trixbox#Extensions extension], so the 2 or 3 or number chosen is sent automatically without the need of dialing.

[[category:guides]]

Asterisk IAX2

2015-07-17T18:05:16Z

Jeff: /* iax.conf */

===iax.conf===

<nowiki>
[general]
register => 100000:johnspassword@atlanta.voip.ms:4569

[voipms]
type=friend
username=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
secret=johnspassword ;your password
context=mycontext
host=atlanta.voip.ms ;(one of our multiple servers, you can choose the one closer to your location)
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support it
insecure=port,invite
requirecalltoken=no
</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(IAX2/voipms/${EXTEN})
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(IAX2/voipms/1${EXTEN})
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(IAX2/voipms/${EXTEN})
exten => _011.,n,Hangup()
exten => _00.,1,Dial(IAX2/voipms/${EXTEN})
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 IAX2

2015-07-17T17:48:34Z

Jeff: /* iax.conf */

===iax.conf===

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

[voipms]
type=friend
username=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
secret=johnspassword ;your password
context=mycontext
host=atlanta.voip.ms ;(one of our multiple servers, you can choose the one closer to your location)
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support it
insecure=port,invite
requirecalltoken=no
</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(IAX2/voipms/${EXTEN})
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(IAX2/voipms/1${EXTEN})
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(IAX2/voipms/${EXTEN})
exten => _011.,n,Hangup()
exten => _00.,1,Dial(IAX2/voipms/${EXTEN})
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>

OBi 100/110

2015-06-09T19:22:43Z

Jeff: /* General Section */

[[File:OBi110-ATA.jpg|none|200px|center]]

 ''The OBi100 is a single phone port ATA adapter that supports SIP VoIP services. The OBi100 is perfect for customers who do not have a traditional phone service, yet need a similar solution and want the savings and simplicity of using a VoIP service for all their calls. To start configuring your OBi100 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.

For an OBi202 please do the following to enable the GUI Web Interface:

Dial *** from the phone connected to the OBi202
Enter 0 For Advanced
Enter 30# Check Mark from
Press 1 to Enter a New Value
Press 1# to Enable
Press 1 to Save
Hang up

If done properly, the following window should appear on your screen:
[[File:ObiLogin.png|300px|thumb|left|Authentication Window - click to enlarge]]
 

Once you get the Authentication Required window you will be prompted for a User Name and Password, the default credentials are:

'''User Name:''' admin

'''Password:''' admin

After this, you should now be able to see the OBi Web interface.

Now on the left side of the screen please find the next options and follow the next steps:

===Disabling auto-provisioning===

'''**NOTE :''' You may use this guide to configure an OBi110 as well. This is the VoIP.ms recommended configuration versus using the Obihai configuration dashboard (more on this later on this page) and you may also not find all new VoIP.ms servers on the Obihai Dahsboard. In order to make sure there will be no conflicts between this Manual configuration and the Obihai dashboard, please perform the following steps to disable auto-provisioning:

*System Management -> Auto Provisioning -> Auto Firmware Update -> Method : Disabled
*System Management -> Auto Provisioning -> ITSP Provisioning -> Method : Disabled
*System Management -> Auto Provisioning -> OBiTALK Provisioning -> Method : Disabled
*Voice Services -> OBiTALK Service -> Enable : Unchecked

Please note you must remove the check mark from the "default" column, then under "Method" please use the ''''Drop Down Selection'''' and choose '''Disabled'''.

[[File:Step1.png|450px|thumb|left|Disabling Auto Provision - click to enlarge]]
 
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) 
:'''DigiMap''': Copy the line, including parenthesis, in the Digitmap field in the ITSP Profile and replace the "555" digits in the following lines by the area code of your choice:

::Dial Plan (recommended):
(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|911|011xx.|xx.|*xx.|***xxx|4xxx|(Mipd)|[^*]@@.)

Also, some clients have been successful by dialing '''**1 and *97''' for line 1 or '''**2 and *97''' for line 2.

:*If you need to set the dial plan back to Default, you can use this:

(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|011xx.|***xxx|xx.|(Mipd)|[^*]@@.)

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

====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.

== Configuration Using OBi Dashboard ==

Besides the Manual Configuration previously explained, Obihai also provides us with their own API dashboard where you can add your device, to complete the configuration in easy steps.
Add your device to the OBiTALK service in the OBi Dashboard [http://www.obitalk.com/obinet/]. Instructions for this are included with the OBi110 and are not discussed here.

After the OBi110 is added, edit the device. You can select '''Service Provider 1''' or '''Service Provider 2''' under the '''Configure Voice Services''' heading. This will take you to a page where you can select ''voip.ms''. Follow the instructions and once you are done the configuration will be downloaded to your Obi110.

== Features Star Codes ==
Please check this link to the Star Codes that are available to activate and deactivate some of the features on your device [http://www.obihai.com/docs/OBiFeatureStarCodes.pdf OBI feature star codes]
 

==Known Issues and Resolutions==
 
=== Settings to avoid direct phone calls to your device in the middle of the night ===
 
Some customers have reported receiving calls in the middle of the night coming from "100" or "101" as callerID. These calls are directly to your device and do not pass through our servers, so we cannot filter them. However, we have some suggestions:
*You can just disable (by unchecking Enable) for SP2 and OBiTALK under your Voice Tab (If you are using our service as SP1).

*You can restrict which IP addresses that can connect to your OBi. Going to "Voice Services -> ITSP Profile A -> SIP -> X_AccessList" : voip.ms_ip_address. You can see the IP address of the server you are currently using from this link: [http://wiki.voip.ms/article/FAQ#What_are_the_IP_addresses_of_VoIP.ms.C2.B4_servers_.3F Server's IPs]

*You can also change your Obi Firewall Setting X_InboundCallRoute to : {(?|x|xx|xxx|xxxx|xxxxx|xxxxxx|un@@.|anon@@.):}, ph
This will only allow 7 digit or greater numbers through.

*Another alternative: OBi Interface>> Voice Services>> SP1 Service>> X_InboundCallRoute: {>('Insert your AuthUserName here'):ph}, example:

{>('100000'):ph} where 100000 is replaced with your own six digit SIP account UserID or the sub-account registered with your device.

By default, OBi devices accept calls destined for any username. The above syntax rejects calls that are not intended for whatever you have configured as AuthUserName.

 

===10 Second Delay Reaching voip.ms Voicemail Attendant when dialing *97 or *98===

The Obi 100, 110 and 202 devices have non-configurable 'short' and 'long' delays if a dialed sequence does not match a digitmap. So you may have a 10 second delay when you dial into your voip.ms voicemail because of the built-in 'long' delay. This can be resolved in a couple of ways. Simply dial a # sign after you dial *97 or *98. Or include literals in your digitmap under the Service Provider / ITSP profile A or B / General / digitmap. Here is an example digitmap with a *97 literal included:

(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|*xx.|'*97'|(Mipd)|[^*#]@@.)

The literal in the example is '*97'. You could also add a literal for '*98'.

Then when you dial *97, the device immediately sends it instead of waiting 10 seconds.

Read more on digitmaps under the topic Digit Map Configuration in the Obi Device Admin Guide.

=== Call Drops ===

If you experience random call drops while in the middle of a call or if the person you talk to remains silent for over a minute (60 seconds by default), OBi will hang up the call. Please go here and check and increase the following setting (Physical interface -> LINE port -> DetectFarEndLongSilence / SilenceTimeThreshold)

=== Enable Message Waiting Indicator MWI ===

To enable MWI please refer to the following section of the OBI web page:

Voice Services -> SP1 Service -> Calling Features ->

'''MWIEnable''' - Uncheck the box at the far right, to be able to check the box at the left, this enables the option.

'''X_VMWIEnable''' - Uncheck the box at the far right, to be able to check the box at the left, this enables the option.

'''MessageWaiting''' - Mark the checkbox

After these steps, the MWI should be active and working.

'''If you are trying to place an outbound call and get a recorded message ¨There is no service to complete your call¨ Please do the following to resolve this.'''
In Your OBi Device please go to Physical Interfaces >> PHONE Port which by default it is PSTN and it needs to be changed to Trunk Group 1

=== Using the OBi Network ===

You can use your OBi device to make calls directly to other OBi devices "''The OBi comes out of the box ready to make FREE calls to other OBi endpoints using the OBiTALK network. Dialing **9 + obi account number will use the OBiTALK feature and does not place calls to regular numbers nor use our network. ''" (you can get more information about [http://www.obihai.com/features-and-set-up here]), be aware that those calls will not pass through our network. If you need assistance with that feature, please contact [http://www.obihai.com/request-support OBI's support].
 
=== Can not dial *98 even if is on your DigiMap ===
By default your OBi device uses *98 as Blind transfer code. If you want to be able to dial *98 from your device, you should change this code. You can achieve this in the settings of your device at: ''Star Codes >> Star code profile (A/B)'', unmark the "default" box and change *98 for something else (like *99)

[[File:Step6.png|550px|thumb|left|Changing *98 default code - click to enlarge]]
 

=== Using the Phone book of your customer portal ===

If you plan on using the Phone Book in your Customer Portal and Speed Dial *75. Please log into your OBi and change the built-in speed dial code from *75 in the device to something else.

=== An additional note regarding outgoing calls===

In at least one instance it was necessary to specify a non-default outbound calling route in the OBi110 to be able to place calls using the voip.ms service. The default setting had the OBi110 attempting to place calls using the PSTN port on the device. The relevant setting is:

'''Physical Interfaces >> PHONE Port '''
*PrimaryLine: (Select from drop-down)

[[File:ObiPhoneport.JPG|550px|thumb|left|Changing Phone Port - click to enlarge]]
 

The default is PSTN. Select SP1 Service if you only have one SIP account configured on the device. Select Trunk Group 1 to have it attempt to place calls using SP1 first, then SP2. Additional Trunk groups can be configured under Voice Services >> Gateways and Trunk Groups.

Portions of this article have been taken from [http://www.toao.net/500-mangos-guide-to-configuring-an-obi100-obi110-and-obi202-ata Mango's Guide to Configuring an OBi ATA]. Used with permission.

[[Category:Analog Telephone Adapters]]

OBi 100/110

2015-06-09T18:55:47Z

Jeff: /* General Section */

[[File:OBi110-ATA.jpg|none|200px|center]]

 ''The OBi100 is a single phone port ATA adapter that supports SIP VoIP services. The OBi100 is perfect for customers who do not have a traditional phone service, yet need a similar solution and want the savings and simplicity of using a VoIP service for all their calls. To start configuring your OBi100 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.

For an OBi202 please do the following to enable the GUI Web Interface:

Dial *** from the phone connected to the OBi202
Enter 0 For Advanced
Enter 30# Check Mark from
Press 1 to Enter a New Value
Press 1# to Enable
Press 1 to Save
Hang up

If done properly, the following window should appear on your screen:
[[File:ObiLogin.png|300px|thumb|left|Authentication Window - click to enlarge]]
 

Once you get the Authentication Required window you will be prompted for a User Name and Password, the default credentials are:

'''User Name:''' admin

'''Password:''' admin

After this, you should now be able to see the OBi Web interface.

Now on the left side of the screen please find the next options and follow the next steps:

===Disabling auto-provisioning===

'''**NOTE :''' You may use this guide to configure an OBi110 as well. This is the VoIP.ms recommended configuration versus using the Obihai configuration dashboard (more on this later on this page) and you may also not find all new VoIP.ms servers on the Obihai Dahsboard. In order to make sure there will be no conflicts between this Manual configuration and the Obihai dashboard, please perform the following steps to disable auto-provisioning:

*System Management -> Auto Provisioning -> Auto Firmware Update -> Method : Disabled
*System Management -> Auto Provisioning -> ITSP Provisioning -> Method : Disabled
*System Management -> Auto Provisioning -> OBiTALK Provisioning -> Method : Disabled
*Voice Services -> OBiTALK Service -> Enable : Unchecked

Please note you must remove the check mark from the "default" column, then under "Method" please use the ''''Drop Down Selection'''' and choose '''Disabled'''.

[[File:Step1.png|450px|thumb|left|Disabling Auto Provision - click to enlarge]]
 
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) 
:'''DigiMap''': Copy the line, including parenthesis, in the Digitmap field in the ITSP Profile and replace the "555" digits in the following lines by the area code of your choice:

::Dial Plan (recommended):
(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|911|011xx.|xx.|*xx.|**xxx|4xxx|(Mipd)|[^*]@@.)

Also, some clients have been successful by dialing '''**1 and *97''' for line 1 or '''**2 and *97''' for line 2.

:*If you need to set the dial plan back to Default, you can use this:

(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|011xx.|**xxx|xx.|(Mipd)|[^*]@@.)

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

====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.

== Configuration Using OBi Dashboard ==

Besides the Manual Configuration previously explained, Obihai also provides us with their own API dashboard where you can add your device, to complete the configuration in easy steps.
Add your device to the OBiTALK service in the OBi Dashboard [http://www.obitalk.com/obinet/]. Instructions for this are included with the OBi110 and are not discussed here.

After the OBi110 is added, edit the device. You can select '''Service Provider 1''' or '''Service Provider 2''' under the '''Configure Voice Services''' heading. This will take you to a page where you can select ''voip.ms''. Follow the instructions and once you are done the configuration will be downloaded to your Obi110.

== Features Star Codes ==
Please check this link to the Star Codes that are available to activate and deactivate some of the features on your device [http://www.obihai.com/docs/OBiFeatureStarCodes.pdf OBI feature star codes]
 

==Known Issues and Resolutions==
 
=== Settings to avoid direct phone calls to your device in the middle of the night ===
 
Some customers have reported receiving calls in the middle of the night coming from "100" or "101" as callerID. These calls are directly to your device and do not pass through our servers, so we cannot filter them. However, we have some suggestions:
*You can just disable (by unchecking Enable) for SP2 and OBiTALK under your Voice Tab (If you are using our service as SP1).

*You can restrict which IP addresses that can connect to your OBi. Going to "Voice Services -> ITSP Profile A -> SIP -> X_AccessList" : voip.ms_ip_address. You can see the IP address of the server you are currently using from this link: [http://wiki.voip.ms/article/FAQ#What_are_the_IP_addresses_of_VoIP.ms.C2.B4_servers_.3F Server's IPs]

*You can also change your Obi Firewall Setting X_InboundCallRoute to : {(?|x|xx|xxx|xxxx|xxxxx|xxxxxx|un@@.|anon@@.):}, ph
This will only allow 7 digit or greater numbers through.

*Another alternative: OBi Interface>> Voice Services>> SP1 Service>> X_InboundCallRoute: {>('Insert your AuthUserName here'):ph}, example:

{>('100000'):ph} where 100000 is replaced with your own six digit SIP account UserID or the sub-account registered with your device.

By default, OBi devices accept calls destined for any username. The above syntax rejects calls that are not intended for whatever you have configured as AuthUserName.

 

===10 Second Delay Reaching voip.ms Voicemail Attendant when dialing *97 or *98===

The Obi 100, 110 and 202 devices have non-configurable 'short' and 'long' delays if a dialed sequence does not match a digitmap. So you may have a 10 second delay when you dial into your voip.ms voicemail because of the built-in 'long' delay. This can be resolved in a couple of ways. Simply dial a # sign after you dial *97 or *98. Or include literals in your digitmap under the Service Provider / ITSP profile A or B / General / digitmap. Here is an example digitmap with a *97 literal included:

(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|*xx.|'*97'|(Mipd)|[^*#]@@.)

The literal in the example is '*97'. You could also add a literal for '*98'.

Then when you dial *97, the device immediately sends it instead of waiting 10 seconds.

Read more on digitmaps under the topic Digit Map Configuration in the Obi Device Admin Guide.

=== Call Drops ===

If you experience random call drops while in the middle of a call or if the person you talk to remains silent for over a minute (60 seconds by default), OBi will hang up the call. Please go here and check and increase the following setting (Physical interface -> LINE port -> DetectFarEndLongSilence / SilenceTimeThreshold)

=== Enable Message Waiting Indicator MWI ===

To enable MWI please refer to the following section of the OBI web page:

Voice Services -> SP1 Service -> Calling Features ->

'''MWIEnable''' - Uncheck the box at the far right, to be able to check the box at the left, this enables the option.

'''X_VMWIEnable''' - Uncheck the box at the far right, to be able to check the box at the left, this enables the option.

'''MessageWaiting''' - Mark the checkbox

After these steps, the MWI should be active and working.

'''If you are trying to place an outbound call and get a recorded message ¨There is no service to complete your call¨ Please do the following to resolve this.'''
In Your OBi Device please go to Physical Interfaces >> PHONE Port which by default it is PSTN and it needs to be changed to Trunk Group 1

=== Using the OBi Network ===

You can use your OBi device to make calls directly to other OBi devices "''The OBi comes out of the box ready to make FREE calls to other OBi endpoints using the OBiTALK network. Dialing **9 + obi account number will use the OBiTALK feature and does not place calls to regular numbers nor use our network. ''" (you can get more information about [http://www.obihai.com/features-and-set-up here]), be aware that those calls will not pass through our network. If you need assistance with that feature, please contact [http://www.obihai.com/request-support OBI's support].
 
=== Can not dial *98 even if is on your DigiMap ===
By default your OBi device uses *98 as Blind transfer code. If you want to be able to dial *98 from your device, you should change this code. You can achieve this in the settings of your device at: ''Star Codes >> Star code profile (A/B)'', unmark the "default" box and change *98 for something else (like *99)

[[File:Step6.png|550px|thumb|left|Changing *98 default code - click to enlarge]]
 

=== Using the Phone book of your customer portal ===

If you plan on using the Phone Book in your Customer Portal and Speed Dial *75. Please log into your OBi and change the built-in speed dial code from *75 in the device to something else.

=== An additional note regarding outgoing calls===

In at least one instance it was necessary to specify a non-default outbound calling route in the OBi110 to be able to place calls using the voip.ms service. The default setting had the OBi110 attempting to place calls using the PSTN port on the device. The relevant setting is:

'''Physical Interfaces >> PHONE Port '''
*PrimaryLine: (Select from drop-down)

[[File:ObiPhoneport.JPG|550px|thumb|left|Changing Phone Port - click to enlarge]]
 

The default is PSTN. Select SP1 Service if you only have one SIP account configured on the device. Select Trunk Group 1 to have it attempt to place calls using SP1 first, then SP2. Additional Trunk groups can be configured under Voice Services >> Gateways and Trunk Groups.

Portions of this article have been taken from [http://www.toao.net/500-mangos-guide-to-configuring-an-obi100-obi110-and-obi202-ata Mango's Guide to Configuring an OBi ATA]. Used with permission.

[[Category:Analog Telephone Adapters]]

Cisco SPA525G

2015-04-10T19:45:00Z

Jeff:

[[File:525.jpg]]

The Cisco SPA525G 5-line IP Phone with Color Display is a full-featured VoIP (Voice
over Internet Protocol) phone that provide voice communication over an IP
network. It provides traditional features, such as call forwarding, redialing, speed
dialing, transferring calls, conference calling, and accessing voice mail. Calls can
be made or received with a handset, headset or speaker.

Your Cisco IP Phone provides a web interface for the phone user that allows you to
configure some features of your phone by using a web browser.

This article will guide you through the steps for basic configuration to make it work with VoIP.ms

----
 
'''Step 1'''
 
'''''Get the IP address of your phone'''''

a. Press Setup. 
b. Select to Status > Network Status. 
c. Scroll to view IP Address. This is the IP address of your phone. 

You should now have a number which is similar to 192.168.xxx.xxx

'''Step 2''' 
'''''Logging in to the Phone Web User Interface'''''

*On your PC, open a Web browser window. Your PC must be on the same subnetwork as the phone.
*Enter the IP address in the browser address bar.

You will now see this screen: 

[[File:525 1.gif|center]]
 
*Click on the '''"Admin Login"''' button near the top right side of the screen, then click on the '''"Ext 1"''' tab.

[[File:525 2.gif|center]]

'''Step 3'''
 
'''''Configure with your VoIP.ms account'''''

Find the following fields on the '''"Ext"''' tab and configure accordingly.

'''Nat Keep Alive:''' Yes 
'''Nat Mapping/Traversal:''' Yes 
'''Proxy:''' atlanta.voip.ms (one of the multiple VoIP.ms servers, you can choose the one closer to your location.) 
'''Register Expires:''' 300 
'''Display Name:''' John Smith (Replace with your name or company name) 
'''User ID:''' 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, e.g. 123456 or 123456_sub) 
'''Password:''' ******** (SIP Account Password) 
'''Use DNS SRV:''' NO 
'''DNS SRV Auto Prefix:''' NO 

[[File:cisco-spa525g.gif|center]]

*If a second extension is needed, click on '''"Ext 2"''' and repeat Step 3. Please make sure to increment the SIP port by one. For example, Ext 1 SIP port: 5060; Ext 2 SIP port: 5061.

'''Step 4'''
 

'''''Configure your dial plan'''''

This step can be considered optional however this is a dial plan that is optimized to work with VoIP.ms service.

Find the dial plan section of your line and enter the following string:

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

'''If done properly, after completing all these steps your phone will now be ready to place and receive calls!'''

== '''Documentation:''' ==

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

[http://www.cisco.com/en/US/docs/voice_ip_comm/csbpipp/ip_phones/user/guide/525g_sip_user_guide_source/spa525_sip_user.pdf User´s Manual] 

keywords: SPA525

Cisco SPA112

2015-04-10T19:43:21Z

Jeff: /* Configuring the Quick Setup screen */

[[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 both customers of VoIP.ms and other providers.'''
: '''Make sure to install the latest firmware from [http://software.cisco.com/download/release.html?mdfid=283998793&softwareid=282463187&release=1.3.3&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. '''

== Configuration Details ==

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

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 initialise; even a correctly configured and installed SPA112/122 will give no power to the 'phone or no dialtone until initialisation 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. Dialling ****110# will give one address, ****210# will give the other.

=== 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|***xx|*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.

[[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: 
'''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 submit 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|<:1'''555'''>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|[3468]11|822|0|00|[2-9]xxxxxx|4xxx|**275x.|xxxxxxxxxxxx.)

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

You can create your own dial plan if you need it, referring to this entry [[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.

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 PAP2's 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 the VoIP.ms (or other provider's) web interface.

Updated firmware is available from the Cisco site [http://software.cisco.com/download/release.html?mdfid=283998771&softwareid=282463187&release=1.3.3&relind=AVAILABLE&rellifecycle=&reltype=latest] 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

[[category:Analog Telephone Adapters]]

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

Dialing Rules and Patterns

2015-04-09T18:46:42Z

Jeff:

= Dialing Rules and Patterns =

This article explains the difference and usage between the Dialing Rules or Dial Plans (From the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk] outgoing settings) and the Dialing Patterns (From the [http://wiki.voip.ms/article/Trixbox#Outbound_Routes Outbound routes]) in the common asterisk distro.

==Dialing Rules==

The most common dialing rule that we can find in the '''[http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk] outgoing settings''' (either SIP or IAX) is the following:

'''1+NXXNXXXXXX'''

What it does is adding the "1" to any pattern like "NXXNXXXXXX"

''It is important to understand that the rules will apply as long as the pattern exists, if it doesn't exist the rule will never apply and the call will end in a typical " This call can not be placed as dialed".''

'''For example, if you want to dial 7 digits only:'''

'''1555+NXXXXXX'''

Replace the 555 digits in the previous line for the area code of your choice.

==Useful VoIP.ms Rules==

'''4443''' - For calling Echo Test to test call connectivity to our servers and call quality.

'''4747''' - For DTMF Testing.

'''***XX''' - To test MOH (Music on Hold) Categories.

'''*xx''' - To access Voicemail Options with our service like *97 and *98.

'''0441+NXXNXXXXXX or 0331+NXXNXXXXXX''' - Used to manually dial a Premium (0441) or a Value (0331) Canadian Route.

'''011+.''' or '''00+.''' - For International Calling.

'''044+.''' and '''033+.''' - To Manually dial (044) Premium International Routes or (033) Value International Routes. Good for Testing a call via different routes on the go.

==Dialing Patterns==

The Dialing patterns can be found in the '''[http://wiki.voip.ms/article/Trixbox#Outbound_Routes Outbound route]''', whatever you dial from any [http://wiki.voip.ms/article/Trixbox#Extensions extension] must match a dialing pattern, the most common dialing pattern found here is the following:

'''NXXNXXXXXX'''

''The important thing to understand, is that the [http://wiki.voip.ms/article/Trixbox#Outbound_Routes outbound route] will select the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunks] it will use'', however if you have multiple [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunks] with the same patterns (which is commonly used), then you will have to select the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunks] priority (which is found at the top right of the outbound route screen, as a list of the [http://wiki.voip.ms/article/Trixbox#Outbound_Routes Outbound routes] names with arrows to move up and down as priority).

Now, what happens if you have multiple [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunks] and you need to force that one [http://wiki.voip.ms/article/Trixbox#Extensions extension] does come up from an specific [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk]? It is a simple play of rules and patterns.

=How to force one [http://wiki.voip.ms/article/Trixbox#Extensions extension] through a specific [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk]=

As has been explained, the [http://wiki.voip.ms/article/Trixbox#Extensions extension] does not "choose" on which [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk] to come out, this is done by the outbound route, what we need to do is to play with the patterns from the [http://wiki.voip.ms/article/Trixbox#Outbound_Routes Outbound routes] and dialing rules for the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunks] set.

What we can chose is the [http://wiki.voip.ms/article/Trixbox#Outbound_Routes outbound route] (which contains the specific [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk]).

How?

Lets say we have trunk1 and trunk2 for this example.
Lets say also we have outbound route1 and outbound route2 for this example.
Also lets say we have the extension1.

The trunk1 and trunk2 dialing rules will be the same = 1+NXXNXXXXXX

Now on the outbound route we can determine the specific pattern that will help us to "choose" either [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk] from when dialing from an specific [[http://wiki.voip.ms/article/Trixbox#Extensions extension]].

We can set to the outbound route1 the pattern:

'''NXXNXXXXXX'''

but also (this is the trick) we can add something like X|. (being X any number of your choice)

''' 2|.'''

This means any pattern with a "2" in front will be recognized by that route and use the specific [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk], the pattern also removes the 2 so this number is not sent and the rule in the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk] 1+ remains.

In this manner we dial a regular US/CAN number (10 digits) this way from the [http://wiki.voip.ms/article/Trixbox#Extensions extensions].

'''25626846308'''

''Note the 2 before the ten digits, this will be stripped out and substitute by 1 according to the dialing rules set in the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk]. By doing this we ensure the use of the trunk1 (which is being chosen in the outbound route).''

Now, in the outbound route2 we add the patterns:

'''NXXNXXXXXX'''

'''3|.'''

This way, when dialing within the [http://wiki.voip.ms/article/Trixbox#Extensions extension] we only need to add a 3 (that will match the specific route) and use the specific [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk].

'''like 35626846308'''

The number 3 will be removed and substitute by the "1" according to the dialing rules in the [http://wiki.voip.ms/article/Trixbox#Trunk_Configuration trunk].

Additionally you can play with the dial rules on the devices that uses the [http://wiki.voip.ms/article/Trixbox#Extensions extension], so the 2 or 3 or number chosen is sent automatically without the need of dialing.

[[category:guides]]

Cisco SPA2100 Phone Adapter

2015-04-09T17:28:21Z

Jeff: /* Configuration Details */

[[File:Sipura2100.jpg|300px|thumb|left|Cisco SPA2100 Phone Adapter]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Haga clic aquí para Español: [[Cisco SPA2100 (Español)]]

==Configuration Details==

'''Step 1'''

The first step is to find out what IP Address your adapter is currently using. To do this, pick a phone connected on Line 1 and do the following:

'''Dial: **** (That is 4 asterisks)''' 
'''Once this is done, dial: 110# (110 followed by a square)'''

The system should now playback the IP Address your device has been assigned. 
(Example: 192.168.1.2)

'''Step 2'''

'''Using your favorite web browser from a computer on the same network, point the address to the IP address of your adapter.''' 
<nowiki>(example http://192.168.1.2)</nowiki> Replace 192.168.1.2 by the IP Address your device is currently using.

'''Step 3'''

You should now see the web interface of your Linksys/Sipura.

'''click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View".'''

'''Step 4'''

'''Under the LINE 1 Tab, Find the following fields and fill them with the following information'''

'''Nat Keep Alive:''' Yes 
'''Nat Mapping/Traversal:''' Yes 
'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms servers) 
'''Register Expires:''' 180 
'''Proxy Fallback Intvl:''' 180 
'''Display Name:''' John Smith (Replace with your name or company name) 
'''User ID:''' 100000 (Replace with your VoIP.ms username) 
'''Password:''' ******** (Type in the account password) 
'''Use DNS SRV:''' NO 
'''DNS SRV Auto Prefix:''' NO

'''Step 5 (Optional)'''

Optionally, you can configure your adapter with a better dial plan, allowing faster dialing of 10 digits number (Local US/Canada) and also enable 7 digits dialing in one area code of your choice.

'''At the bottom of Line 1 TAB, you will find a field called Dial Plan'''

Replace the 555 digits in the following line by the area code of your choice and copy the line, including parenthesis, in the Dialplan field in Line 1 Tab at the bottom of the page:

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

'''Step 6'''

Click on the "Save Settings " button at the bottom of the form.

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

== How to avoid the long delay to hear the ringtone ==

If you ever experience some delay to hear the ringtone when you make outgoing calls with your SPA. Changing the SPA's Interdigit Long Timer value can help resolve the issue. Follow the next steps in order to change that setting:

'''Note''': However before changing that option, test if calling the number with an # at the end of the number works(e.g. 5554441234#).
If that doesn't work you need to contact the support staff in voip.ms.

*1- First access the SPA's web interface.
*2- Click on the '''Admin Login''' and then click on the '''Voice''' tab.
*3- Click on the '''Regional''' tab and look for the '''Control Timer Values (sec)''' section.
*4- Enter the desire value in the '''Interdigit Long Timer''' field (for example lower this value to 4).

[[File:Ctrl timer values.jpg|800px]]

'''Note''': The image correspond to the PAP2 device. If your device doesn't have this setting, contact voip.ms support.

== See also ==
* [[Dial Plan for Linksys ATAs]]
* You can check the most commonly used Star Codes from [[Cisco/Linksys Star Codes]]

[[Category:Analog Telephone Adapters]]

Cisco Linksys SPA942 NA

2015-04-09T17:18:15Z

Jeff: /* Configuration Details */

[[File:SPA942-0.jpg|300px|thumb|left|Cisco Linksys SPA942 NA]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Haga clic aquí para Español: [[Cisco Linksys SPA942 NA (Español)]]

==Configuration Details==

'''Step 1'''

The first step is to find out what IP Address your adapter is currently using. To do this, pick a phone connected on Line 1 and do the following:

'''Dial: **** (That is 4 asterisks)''' 
'''Once this is done, dial: 110# (110 followed by a square)'''

The system should now playback the IP Address your device has been assigned. 
(Example: 192.168.1.2)

Note: Some multi-line Cisco IP 'phones do provide a [setup] key (icon is one piece of paper with one corner folded). On these models, pressing [setup] then selecting 'Network' from the menu provides its IP without dialling any codes. The remainder of the configuration process remains the same.

'''Step 2'''

'''Using your favorite web browser from a computer on the same network, point the address to the IP address of your adapter.''' 
<nowiki>(example http://192.168.1.2)</nowiki> Replace 192.168.1.2 by the IP Address your device is currently using.

'''Step 3'''

You should now see the web interface of your Linksys/Sipura.

'''click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View".'''

'''Step 4'''

'''Under the LINE 1 Tab, Find the following fields and fill them with the following information'''

'''Nat Keep Alive:''' Yes 
'''Nat Mapping/Traversal:''' Yes 
'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms servers) 
'''Register Expires:''' 180 
'''Proxy Fallback Intvl:''' 180 
'''Display Name:''' John Smith (Replace with your name or company name) 
'''User ID:''' 100000 (Replace with your VoIP.ms username) 
'''Password:''' ******** (Type in the account password) 
'''Use DNS SRV:''' NO 
'''DNS SRV Auto Prefix:''' NO

'''Step 5 (Optional)'''

Optionally, you can configure your adapter with a better dial plan, allowing faster dialing of 10 digits number (Local US/Canada) and also enable 7 digits dialing in one area code of your choice.

'''At the bottom of Line 1 TAB, you will find a field called Dial Plan'''

Replace the 555 digits in the following line by the area code of your choice and copy the line, including parenthesis, in the Dialplan field in Line 1 Tab at the bottom of the page:

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

'''Step 6'''

Click on the "Save Settings " button at the bottom of the form.

== See also ==
* [[Dial Plan for Linksys ATAs|Dial Plan for Cisco and Linksys devices]]; the same rules apply to dial plans for Cisco's IP 'phones as to Linksys ATA's.

* You can check the most commonly used Star Codes from [[Cisco/Linksys Star Codes]]
[[Category: SIP phones]]

Cisco SPA2102 Phone Adapter with Router

2015-04-09T17:17:11Z

Jeff: /* Configuration Details */

[[File:Spa2102b.jpg|300px|thumb|left|Cisco SPA2102 Phone Adapter with Router]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Details==

''This is a guide for the Initial Configuration of the SPA2102/3102.''

'''Step 1'''

*Power off your network devices, including your modem and PC.
*Connect an Ethernet Cable from the Ethernet port of the SPA to the Ethernet port of the PC.
*Connect an Ethernet Cable from the Internet port of the SPA to the LAN/Ethernet port of the Modem.
*Connect your regular handset phone, to the Line port of the SPA2102.
*Then power up modem, then the SPA2102 and then the PC.
*Launch a web browser from the PC and enter "'''http://192.168.0.1/advanced'''" in the URL address bar field.

After these steps you should now have access to the Web Interface page of the SPA2102 to start with the initial configuration.

'''NOTE: If this is not working or the Browser can't find the page, you may also need to enable the Administration web service. '''
'''Dial 7932#, then when prompted press 1 to enable and 1 to confirm''' '''

You can also check this information from the Linksys Quick Guide: http://wiki.voip.ms/files/linksys-spa-2102-user-guide.pdf

'''Step 2'''

You should now see the web interface of your Linksys/Sipura.

'''click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View".'''

'''Step 3'''

'''Under the LINE 1 Tab, Find the following fields and fill them with the following information'''

'''Nat Keep Alive:''' Yes 
'''Nat Mapping/Traversal:''' Yes 
'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms servers) 
'''Register Expires:''' 180 
'''Proxy Fallback Intvl:''' 180 
'''Display Name:''' John Smith (Replace with your name or company name) 
'''User ID:''' 100000 (Replace with your VoIP.ms username) 
'''Password:''' ******** (Type in the account password) 
'''Use DNS SRV:''' NO 
'''DNS SRV Auto Prefix:''' NO

'''Step 3.5'''

'''On the SIP tab, Under NAT Support Parameters, set the following.'''

'''Handle VIA received:''' Yes 
'''Substitute VIA Addr:''' Yes 

'''Step 4 (Optional)'''

Optionally, you can configure your adapter with a better dial plan, allowing faster dialing of 10 digits number (Local US/Canada) and also enable 7 digits dialing in one area code of your choice.

'''At the bottom of Line 1 TAB, you will find a field called Dial Plan'''

Replace the 555 digits in the following line by the area code of your choice and copy the line, including parenthesis, in the Dialplan field in Line 1 Tab at the bottom of the page:

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

[[File:SPA2102Line1.png]]

'''Step 5'''

Click on the "Save Settings " button at the bottom of the form.

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

== How to avoid the long delay to hear the ringtone ==

If you ever experience some delay to hear the ringtone when you make outgoing calls with your SPA. Changing the SPA's Interdigit Long Timer value can help resolve the issue. Follow the next steps in order to change that setting:

'''Note''': However before changing that option, test if calling the number with an # at the end of the number works(e.g. 5554441234#).
If that doesn't work you need to contact the support staff in voip.ms.

*1- First access the SPA's web interface.
*2- Click on the '''Admin Login''' and then click on the '''Voice''' tab.
*3- Click on the '''Regional''' tab and look for the '''Control Timer Values (sec)''' section.
*4- Enter the desire value in the '''Interdigit Long Timer''' field (for example lower this value to 4).

[[File:Ctrl timer values.jpg|800px]]

'''Note''': The image correspond to the PAP2 device. If your device doesn't have this setting, contact voip.ms support.

'''Additionally, whenever the registration is dropped and its having problems to gain it back, you can try this simple trick:'''
'''On Advanced view, go to the Line tab, look for the SIP port setting, and change it to another, i.e. if you have port 5060 change to 5080, if it is 5061 you can change to 5081.'''

== Known Issues ==

If you are receiving phone calls at strange times and with a strange CallerID and they do not show up in your CDR then these calls could be Direct IP Device Calls trying to connect to you. To remove these types of calls please follow these directions.

Please Login to your Cisco Adapter and from Admin Login > Advanced > Voice > System > System Configuration > Restricted Access Domain

In this field put the servers you connect to: I.E. montreal.voip.ms, toronto.voip.ms
Please remember you did this in case you change VoIP.ms servers.

Once the Restricted Access Domain field is populated with Point-of-Presence (POP) servers, any server specified in Proxy field (Voice > LINE1> Proxy & Registration) will not be able to establish registration UNLESS the same server is included in the Restricted Access Domain field.

== Factory Reset Procedure ==

Step 1: Connect a phone into TEL1 port

Step 2: On the handset dial digits **** (4 Stars)

Step 3: Dial 73738# then press 1 to confirm

== See also ==
* [[Dial Plan for Linksys ATAs]]
* You can check the most commonly used Star Codes from [[Cisco/Linksys Star Codes]]

[[Category:Analog Telephone Adapters]]

[http://tinyurl.com/lf7f3bo PDF Owner`s Manual]

Cisco SPA504G Phone

2015-04-09T17:15:03Z

Jeff: /* Configuration Details */

[[File:Cisco_spa504g.jpg|300px|thumb|left|Cisco SPA504G Phone]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Details==

'''Step 1'''

The first step is to find out what IP Address your adapter is currently using. To do this, pick a phone connected on Line 1 and do the following:

'''Dial: **** (That is 4 asterisks)''' 
'''Once this is done, dial: 110# (110 followed by a square)'''

The system should now playback the IP Address your device has been assigned. 
(Example: 192.168.1.2)

Note: Some versions of Cisco IP 'phone (typically multi-line devices from the SPA30x and SPA50x series) provide a [setup] key (icon is one piece of paper with one corner folded). On these models, pressing [setup] then selecting 'Network' from the menu provides its IP without dialing any codes.

'''Step 2'''

'''Using your favorite web browser from a computer on the same network, point the address to the IP address of your adapter.''' 
<nowiki>(example http://192.168.1.2)</nowiki> Replace 192.168.1.2 by the IP Address your device is currently using.

'''Step 3'''

You should now see the web interface of your Linksys/Sipura.

'''click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View".'''

'''Step 4'''

'''Under the LINE 1 Tab, Find the following fields and fill them with the following information'''

'''Nat Keep Alive:''' Yes 
'''Nat Mapping/Traversal:''' Yes 
'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms servers) 
'''Register Expires:''' 300 
'''Proxy Fallback Intvl:''' 300 
'''Display Name:''' John Smith (Replace with your name or company name) 
'''User ID:''' 100000 (Replace with your VoIP.ms username) 
'''Password:''' ******** (Type in the account password) 
'''Use DNS SRV:''' NO 
'''DNS SRV Auto Prefix:''' NO

(On multi-line 'phones like the SPA30x/50x this tab is labelled 'Ext1', 'Ext2', 'Ext3'. The initial defaults assign all individual line keys to 'Ext1' settings. Labels are specified on the 'Phone' tab)

'''Step 5 (Optional)'''

Optionally, you can configure your adapter with a better dial plan, allowing faster dialing of 10 digits number (Local US/Canada) and also enable 7 digits dialing in one area code of your choice.

'''At the bottom of Line 1 TAB, you will find a field called Dial Plan'''

Replace the 555 digits in the following line by the area code of your choice and copy the line, including parenthesis, in the Dialplan field in Line 1 Tab (or Ext1 tab) at the bottom of the page:

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

'''Step 6'''

Click on the "Save Settings " button at the bottom of the form. The IP 'phone may take up to 35 seconds to re-initialise.

== See also ==
* [[Dial Plan for Linksys ATAs|Dial Plan for Cisco and Linksys devices]]; the same rules apply to dial plans for Cisco's IP 'phones as to Linksys ATA's. Multiline IP 'phones are easier to configure as the dialplan-translated number is displayed on the LCD when completing the call but the syntax is identical.

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

[[Category: SIP phones]]

Cisco Linksys SPA942 NA (Español)

2015-04-09T17:13:27Z

Jeff: /* Configuration Details */

[[File:SPA942-0.jpg|300px|thumb|left|Cisco Linksys SPA942 NA]]
<div style="width:100%;overflow:hidden;clear:both"></div>
==Configuration Details==

'''Paso 1'''

El primer paso es encontrar cual es la dirección IP que el adaptador esta usando. Para hacer esto, tome el teléfono conectado a la linea 1 y haga lo siguiente:

'''Marque: **** (4 Asteriscos)''' 
'''Una vez hecho esto, marque: 110# (110 seguido del símbolo #)'''

Deberá escuchar una grabación que dicta la dirección IP del adaptador. 
(Ejemplo: 192.168.1.2)

'''Paso 2'''

'''Usando su navegador favorito desde una computadora en la misma Red, dirija la dirección de una pagina hacia la dirección IP del adaptador.''' 
<nowiki>(ejemplo http://192.168.1.2)</nowiki> Reemplace 192.168.1.2 por la dirección IP de su adaptador.

'''Paso 3'''

Usted ahora debe estar viendo la pagina Web del Linksys/Sipura.

'''Haga clic en el link "Admin", cuando cargue la página, haga clic nuevamente en el link "Advanced View".'''

'''Paso 4'''

'''En la pestaña de LINE 1, encuentre los siguientes campos y llenelos con la información correcta a continuación:'''

'''Proxy:''' atlanta.voip.ms (Atlanta es solo un ejemplo, aquí puede usar cualquiera de los servidores) 
'''Display Name:''' John Smith (Su nombre o el de su compañia) 
'''User ID:''' 100000 (Su número de usuario de VoIP.ms, 10000 es solo un ejemplo) 
'''Password:''' ******** (La contraseña de su cuenta.) 
'''Register Expires:''' 180 
'''Proxy Fallback Intvl:''' 180 
'''Nat Keep Alive:''' Yes 
'''Nat Mapping/Traversal:''' Yes

'''Paso 5 (Opcional)'''

Opcionalmente, puede configurar su adaptador con un plan de marcado mas personalizado, haciendo posible un marcado de los números mas rápido (Local US/Canada) y también habilitar el marcado de 7 dígitos para el área de su preferencia.

'''Al final de la Linea 1, usted vera el campo llamado "Dial Plan"'''

Reemplace los dígitos 555 en la siguiente linea por el código de área de su elección y copie la linea completa incluyendo paréntesis en el campo del "Dial Plan":

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

'''Paso 6'''

Haga clic en "Save Settings " al final de la pagina.
[[category:teléfonos IP]]

Cisco Linksys PAP2T Español

2015-04-09T17:12:12Z

Jeff: /* Detalles de la Configuración */

[[File:Pap2t.jpg|300px|thumb|left|Cisco Linksys PAP2T]]
<div style="width:100%;overflow:hidden;clear:both"></div>
==Detalles de la Configuración==

'''Paso 1'''

El primer paso es encontrar cual es la dirección IP que el adaptador esta usando. Para hacer esto, tome el teléfono conectado a la linea 1 y haga lo siguiente:

'''Marque: **** (4 Asteriscos)''' 
'''Una vez hecho esto, marque: 110# (110 seguido del símbolo #)'''

Deberá escuchar una grabación que dicta la dirección IP del adaptador. 
(Ejemplo: 192.168.1.2)

'''Paso 2'''

'''Usando su navegador favorito desde una computadora en la misma Red, dirija la dirección de una pagina hacia la dirección IP del adaptador.''' 
<nowiki>(ejemplo http://192.168.1.2)</nowiki> Reemplace 192.168.1.2 por la dirección IP de su adaptador.

'''Paso 3'''

Usted ahora debe estar viendo la pagina Web del Linksys/Sipura.

'''Haga clic en el link "Admin", cuando cargue la página, haga clic nuevamente en el link "Advanced View".'''

'''Paso 4'''

'''En la pestaña de LINE 1, encuentre los siguientes campos y llenelos con la información correcta a continuación:'''

'''Proxy:''' atlanta.voip.ms (Atlanta es solo un ejemplo, aquí puede usar cualquiera de los servidores) 
'''Display Name:''' John Smith (Su nombre o el de su compañia) 
'''User ID:''' 100000 (Su número de usuario de VoIP.ms, 10000 es solo un ejemplo) 
'''Password:''' ******** (La contraseña de su cuenta.) 
'''Register Expires:''' 180 
'''Proxy Fallback Intvl:''' 180 
'''Nat Keep Alive:''' Yes 
'''Nat Mapping/Traversal:''' Yes

'''Paso 5 (Opcional)'''

Adicionalmente, para ahorrar ancho de banda, usted puede configurar la Line 1 con codec preferido, "Preferred Codec", para G729a y asegúrese de que la opción "Use Pref Codec Only" este en NO.

'''Paso 6 (Opcional)'''

Opcionalmente, puede configurar su adaptador con un plan de marcado mas personalizado, haciendo posible un marcado de los números mas rápido (Local US/Canada) y también habilitar el marcado de 7 dígitos para el área de su preferencia.

'''Al final de la Linea 1, usted vera el campo llamado "Dial Plan"'''

Reemplace los dígitos 555 en la siguiente linea por el código de área de su elección y copie la linea completa incluyendo paréntesis en el campo del "Dial Plan":

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

'''Paso 7'''

Haga clic en "Save Settings " al final de la pagina.

'''Paso 8'''

Cambie a la pestaña de '''SIP''' y baje hasta el campo de '''RTP Parameters''' y escriba lo siguiente:

'''RTP Packet Size''': 0.020

=='''Capturas de Pantalla'''==

[[File:PAP2T001.JPG|PAP2 Configuration Screen|frame|left]]

[[File:PAP2T002.JPG|PAP2 Configuration Screen|frame|left]]

[[File:PAP2T003.JPG|PAP2 Configuration Screen|frame|left]]

[[File:Pap2sip.jpg|PAP2 Configuration Screen|frame|left]]

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

== Como prevenir el retraso en el tono, después de marcar ==

Si usted experimenta algún retraso en escuchar el tono de marcado después de hacer una llamada con sus PAP2, es posible modificar un valor de la configuración llamado, '''"Interdigit Long Timer"'''. Siga los siguientes pasos para lograr este cambio:

'''NOTA''': Sin embargo, antes de cambiar esta opción, pruebe si marcar un número, seguido del botón de #, hace alguna diferencia.
Por ejemplo: 5551231234#, si esto no es así, este cambio no tendra efecto y le sugiero contactar al soporte de VoIP.ms

*1- Primero accese a la interface Web del PAP2.
*2- Haga clic en el link '''Admin Login''' y luego clic en '''(switch to advanced view)'''

[[File:Pap2admlog.jpg|800px]]

*3- Haga clic en la pestaña '''Regional''' y busque la sección de '''Control Timer Values (sec)'''.

[[File:Regional.jpg|800px]]

*4- Ingrese el valor deseado en el campo de '''Interdigit Long Timer''' (por ejemplo, reduzca el valor a 4).

[[File:Ctrl timer values.jpg|800px]]
[[category:adaptador de teléfono analógico]]

Cisco Linksys PAP2 (Español)

2015-04-09T17:11:09Z

Jeff: /* Configuration Details */

[[File:Pap2.jpg|300px|thumb|left|Cisco Linksys PAP2]]
<div style="width:100%;overflow:hidden;clear:both"></div>
==Configuration Details==

'''Paso 1'''

El primer paso es encontrar cual es la dirección IP que el adaptador esta usando. Para hacer esto, tome el teléfono conectado a la linea 1 y haga lo siguiente:

'''Marque: **** (4 Asteriscos)''' 
'''Una vez hecho esto, marque: 110# (110 seguido del símbolo #)'''

Deberá escuchar una grabación que dicta la dirección IP del adaptador. 
(Ejemplo: 192.168.1.2)

'''Paso 2'''

'''Usando su navegador favorito desde una computadora en la misma Red, dirija la dirección de una pagina hacia la dirección IP del adaptador.''' 
<nowiki>(ejemplo http://192.168.1.2)</nowiki> Reemplace 192.168.1.2 por la dirección IP de su adaptador.

'''Paso 3'''

Usted ahora debe estar viendo la pagina Web del Linksys/Sipura.

'''Haga clic en el link "Admin", cuando cargue la página, haga clic nuevamente en el link "Advanced View".'''

'''Paso 4'''

'''En la pestaña de LINE 1, encuentre los siguientes campos y llenelos con la información correcta a continuación:'''

'''Proxy:''' atlanta.voip.ms (Atlanta es solo un ejemplo, aquí puede usar cualquiera de los servidores) 
'''Display Name:''' John Smith (Su nombre o el de su compañia) 
'''User ID:''' 100000 (Su número de usuario de VoIP.ms, 10000 es solo un ejemplo) 
'''Password:''' ******** (La contraseña de su cuenta.) 
'''Register Expires:''' 180 
'''Proxy Fallback Intvl:''' 180 
'''Nat Keep Alive:''' Yes 
'''Nat Mapping/Traversal:''' Yes

'''Paso 5 (Opcional)'''

Opcionalmente, puede configurar su adaptador con un plan de marcado mas personalizado, haciendo posible un marcado de los números mas rápido (Local US/Canada) y también habilitar el marcado de 7 dígitos para el área de su preferencia.

'''Al final de la Linea 1, usted vera el campo llamado "Dial Plan"'''

Reemplace los dígitos 555 en la siguiente linea por el código de área de su elección y copie la linea completa incluyendo paréntesis en el campo del "Dial Plan":

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

'''Paso 6'''

Haga clic en "Save Settings " al final de la pagina.

'''Paso 7'''

Cambie a la pestaña de '''SIP''' y baje hasta el campo de '''RTP Parameters''' y escriba lo siguiente:

'''RTP Packet Size''': 0.020

=='''Capturas de Pantalla'''==

[[File:PAP2T001.JPG|PAP2 Configuration Screen|frame|left]]

[[File:PAP2T002.JPG|PAP2 Configuration Screen|frame|left]]

[[File:PAP2T003.JPG|PAP2 Configuration Screen|frame|left]]

[[File:Pap2sip.jpg|PAP2 Configuration Screen|frame|left]]

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

== Como prevenir el retraso en el tono, después de marcar ==

Si usted experimenta algún retraso en escuchar el tono de marcado después de hacer una llamada con sus PAP2, es posible modificar un valor de la configuración llamado, '''"Interdigit Long Timer"'''. Siga los siguientes pasos para lograr este cambio:

'''NOTA''': Sin embargo, antes de cambiar esta opción, pruebe si marcar un número, seguido del botón de #, hace alguna diferencia.
Por ejemplo: 5551231234#, si esto no es así, este cambio no tendra efecto y le sugiero contactar al soporte de VoIP.ms

*1- Primero accese a la interface Web del PAP2.
*2- Haga clic en el link '''Admin Login''' y luego clic en '''(switch to advanced view)'''

[[File:Pap2admlog.jpg|800px]]

*3- Haga clic en la pestaña '''Regional''' y busque la sección de '''Control Timer Values (sec)'''.

[[File:Regional.jpg|800px]]

*4- Ingrese el valor deseado en el campo de '''Interdigit Long Timer''' (por ejemplo, reduzca el valor a 4).

[[File:Ctrl timer values.jpg|800px]]
[[category:adaptador de teléfono analógico]]

Dial Plan para ATA Linksys

2015-04-09T17:10:09Z

Jeff:

El dial plan en la sección de ejemplos de configuración para ATA Linksys (como [[Cisco Linksys PAP2|PAP2]], [[Cisco Linksys PAP2T|PAP2T]] y [[Cisco SPA2100 Phone Adapter|SPA2100]]), debe de funcionar sin ningun problema con VoIP.ms

'''Dial plan recomendado por VoIP.ms ''':

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

'''Nota''': Reemplace 555 por su código de área. De esta forma solo tendrá que marcar 7 números para su código de área.

Esta guía ha sido creada para ayudarle a entender su Dial Plan y como personalizarlo de acuerdo a sus preferencias.
Tenga en cuenta que personalizar su Dial plan es '''opcional'''.

__TOC__

== ¿ Qué es un Dial Plan? ==

El Dial plan es una cadena de caracteres que determina cómo se interpretan los digitos que oprime en el teclado de su adaptador ATA, también determina si el número marcado puede ser aceptado o rechazado. De esta forma usted puede utilizar el dial plan para facilitar el marcado y bloquear cierto tipo de llamadas, ya sean de larga distancia o internacionales.

'''Nota''': Tenga en cuenta que también puede bloquear las llamadas internacionales de su cuenta VoIP.ms

== Secuencia de Dígitos ==

Un dial plan contiene una serie de dígitos secuenciales separados por el caracter | y todo el contenido de la secuencia esta encerrado en paréntesis. Cada vez que usted presiona una tecla, su adaptador ATA va a tratar de compararla con la secuencia de dígitos de su dial plan.

{| class="wikitable"
|-
! Secuencia
! Función de los Dígitos
|-
| style="text-align:center;"|0 1 2 3 4 5 6 7 8 9 * #
| Puede utilizar cualquiera de estos caracteres para representar una tecla presionada.
|-
| style="text-align:center;"| x
| Esto representa cualquier caracter en el teléfono.
|-
| style="text-align:center;"|[secuencia]
| Puede utilizar caracteres entre [ ] para crear una lista de dígitos aceptados. Por ejemplo, si utiliza [1-5] esto permitirá al usuario oprimir cualquier dígito entre 1 y 5. También puede crear una lista usando números con otros caracteres, por ejemplo [35-8*], esto permitirá introducir 3,5,6,7,8, *
|-
| style="text-align:center;"|. (punto)
| Puede utilizar un punto "." para aceptar cero o mas veces un determinado número. Por ejemplo, '''01.''' permite introducir 0, 01, 011, etc.
|-
| style="text-align:center;"|<marcado:sustituído>
| Esto es usado para substituir una secuencia, puede indicar que ciertos números marcados sean remplazados por otros caracteres. Los dígitos marcados pueden ser ninguno o más caracteres. Por ejemplo, con esta secuencia '''<:1555>xxxxxxx''' si usted marca un número de 7 dígitos, el número 1555 será agregado al principio de la secuencia. Si presiona 6782345, el sistema enviará automáticamente 15556782345.
|-
| style="text-align:center;"|, (coma)
| Esto puede ser usado entre dígitos para reproducir un tono de marcado después de introducir una secuencia de caracteres. Por ejemplo, con esta secuencia '''9, 1x.''' se escuchará tono de marcado después de presionar 9 y el sonido continuará hasta presionar 1.
|-
| style="text-align:center;"|! (signo de exclamación)
| Puede usar este caracter para prohibir el marcado de una secuencia. Por ejemplo, con la secuencia '''1900xxxxxxx!''' el sistema rechazará cualquier secuencia que comience con 1900.
|-
| style="text-align:center;"|S0 or L0
| Esto sobrescribirá el parámetro de "Short inter-digit timer" ó "Long inter-digit timer" a 0 segundos. Este parámetro es utilizado para indicar el número de segundos que tardará en escucharse el tono de marcado después de introducir el último dígito marcado.
|-
| style="text-align:center;"|P# (donde # es el la duración de la pausa en segundos)
| Esto proporcionará una pausa según el número de segundos que se haya especificado.
|}

== Ejemplos ==

Aquí se presentan algunos ejemplos de secuencias que puede agregar a su dial plan:

* Para marcar cualquier número internacional sin usar el prefijo 011.
<:011> [2-9]xxxxxxxx.

También puede lograr esto si seleciona el modo de marcado(Dial Mode) E164 en su cuenta [[Account Settings]]

* Para bloquear una llamada a un área específica, reemplace el 555 por el código de área que desee.
<:1> 555 xxxxxxx !

* La siguiente secuencia, le permite marcar los números de su libreta de direcciones [[Phone book]] utilizando el marcado rápido. Por ejemplo, si usted marca 20# el sistema enviará *7520.
<:*75>xx< # : >

== Referencias ==
[http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf Cisco - Administration Guide SPA2102, SPA3102, SPA8000, SPA8800, PAP2T]

[[category:Guías]]

Cisco Linksys PAP2T

2015-04-09T17:09:19Z

Jeff: /* Configuration Details */

[[File:Pap2t.jpg|300px|thumb|left|Cisco Linksys PAP2T]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Haga clic aquí para la versión en Español:: [[Cisco Linksys PAP2T (Español)]]
__TOC__

'''If you are running the PAP2t with a tomato firmware router, this may represent a persistent issue after'''
'''rebooting the device, generally it won't register back as usual, in order to fix this you only need to set'''
'''Tomato UDP Unreplied timeout down to 10 (from default of 30).'''

==Configuration Details==

'''Step 1'''

The first step is to find out what IP Address your adapter is currently using. To do this, pick a phone connected on Line 1 and do the following:

'''Dial: **** (That is 4 asterisks)''' 
'''Once this is done, dial: 110# (110 followed by a square)'''

The system should now playback the IP Address your device has been assigned. 
(Example: 192.168.1.2)

'''Step 2'''

'''Using your favorite web browser from a computer on the same network, point the address to the IP address of your adapter.''' 
<nowiki>(example http://192.168.1.2)</nowiki> Replace 192.168.1.2 by the IP Address your device is currently using.

'''Step 3'''

You should now see the web interface of your Linksys/Sipura.

'''At the top of the page below the tabs, click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View".'''

'''Step 4'''

'''Under the LINE 1 Tab, Find the following fields and fill them with the following information'''

'''Nat Keep Alive:''' Yes 
'''Nat Mapping/Traversal:''' Yes 
'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms servers) 
'''Register Expires:''' 180 
'''Proxy Fallback Intvl:''' 180 
'''Display Name:''' John Smith (Replace with your name or company name) 
'''User ID:''' 100000 (Replace with your VoIP.ms username) 
'''Password:''' ******** (Type in the account password) 
'''Use DNS SRV:''' NO 
'''DNS SRV Auto Prefix:''' NO

'''Step 5'''

You can configure your adapter with a better dial plan, allowing faster dialing of 10 digits number (Local US/Canada) and also enable 7 digits dialing in one area code of your choice.

'''At the bottom of Line 1 TAB, you will find a field called Dial Plan'''

Replace the 555 digits in the following line by the area code of your choice and copy the line, including parenthesis, in the Dialplan field in Line 1 Tab at the bottom of the page:

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

[[Dial Plan for Linksys ATAs|Click here for more information on Linksys dial plans]]

'''Step 6'''

Click on the "Save Settings " button at the bottom of the form.

'''Configuration Screens'''

[[File:PAP2T001.JPG|PAP2T Configuration Screen|frame|left]]

[[File:PAP2T002.JPG|PAP2T Configuration Screen|frame|left]]

[[File:PAP2T003.JPG|PAP2T Configuration Screen|frame|left]]

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

'''Customer Submitted Information:''' 
For North America: 
Found this link on configuring the PAP2-NA hardware to work better in North America and specifically with VOIP.ms. 
Read the article called: [http://www.toao.net/25-linksys-ata-configuration Configure your Linksys VoIP ATA the right way!]

To upgrade a firmware version from a Windows system, the PAP2T-NA documentation nor Cisco's web site does not say how to do this.
Go to [http://www.cisco.com/en/US/products/ps10029/index.html Cisco PAP2T Internet Phone Adapter with 2 VoIP Ports]. Click on the link "Download Firmware", which downloads the .zip file. Run the .exe file, then enter the IP number of the ATA device (called “SPA” in the program). It then upgrades the device. This is provided for your information: the author is not saying you have to upgrade the firmware.

== How to avoid the long delay to hear the ringtone ==

If you ever experience some delay to hear the ringtone when you make outgoing calls with your PAP2T. Changing the PAP2T's Interdigit Long Timer value can help resolve the issue. Follow the next steps in order to change that setting:

'''Note''': However before changing that option, test if calling the number with an # at the end of the number works(e.g. 5554441234#).
If that doesn't work you need to contact the support staff in voip.ms.

*1- First access the PAP2T's web interface.
*2- Click on the '''Admin Login''' and then click on '''(switch to advanced view)'''

[[File:Pap2admlog.jpg|800px]]

*3- Click on the '''Regional''' tab and look for the '''Control Timer Values (sec)''' section.

[[File:Regional.jpg|800px]]

*4- Enter the desire value in the '''Interdigit Long Timer''' field (for example lower this value to 4).

[[File:Ctrl timer values.jpg|800px]]

== '''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 PAP2's 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]]

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 Linksys 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.

==Linksys PAP2T ATA Adapter Reset Procedure==

Sometimes it will be very helpful to reset your linksys ATA adapter to factory default settings.

*Connect a telephone to line 1 of the PAP2T unit and power it on.
*Disconnect your PAP2T adapter from the internet connection(unplug the Ethernet cable from the PAP2T hardware unit). Resetting with internet connection may mess up the unit making it completely useless.
*Dial ****, and wait for the Interactive Voice Menu (IVM) to get activated.
*Type in the following number including the # symbol.

73738#

(This number spells RESET.)

*Confirm this by pressing 1.

Your linksys ATA unit will now go back to it factory default settings.

'''Note:''' There are some PAP2/PAP2T devices in circulation which were originally 'locked' to one provider and subsequently unlocked by end users. Do *not* use the RESET$ command with these boxes.

== See also ==
* [[Dial Plan for Linksys ATAs]]
* You can check the most commonly used Star Codes from [[Cisco/Linksys Star Codes]]

[[Category:Analog Telephone Adapters]]

Cisco Linksys PAP2

2015-04-09T17:08:04Z

Jeff: /* Configuration Details */

[[File:Pap2.jpg|300px|thumb|left|Cisco Linksys PAP2]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Haga clic aquí para Español: [[Cisco Linksys PAP2 (Español)]]

==Configuration Details==

'''Step 1'''

The first step is to find out what IP Address your adapter is currently using. To do this, pick a phone connected on Line 1 and do the following:

'''Dial: **** (That is 4 asterisks)''' 
'''Once this is done, dial: 110# (110 followed by a square)'''

The system should now playback the IP Address your device has been assigned. 
(Example: 192.168.1.2)

'''Step 2'''

'''Using your favorite web browser from a computer on the same network, point the address to the IP address of your adapter.''' 
<nowiki>(example http://192.168.1.2)</nowiki> Replace 192.168.1.2 by the IP Address your device is currently using.

'''Step 3'''

You should now see the web interface of your Linksys/Sipura.

'''click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View".'''

'''Step 4'''

'''Under the LINE 1 Tab, Find the following fields and fill them with the following information'''

'''Nat Keep Alive:''' Yes 
'''Nat Mapping/Traversal:''' Yes 
'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms servers) 
'''Register Expires:''' 180 
'''Proxy Fallback Intvl:''' 180 
'''Display Name:''' John Smith (Replace with your name or company name) 
'''User ID:''' 100000 (Replace with your VoIP.ms username) 
'''Password:''' ******** (Type in the account password) 
'''Use DNS SRV:''' NO 
'''DNS SRV Auto Prefix:''' NO

'''Step 5 '''

You can configure your adapter with a better dial plan, allowing faster dialing of 10 digits number (Local US/Canada) and also enable 7 digits dialing in one area code of your choice. Here is the one we recommend:

'''At the bottom of Line 1 TAB, you will find a field called Dial Plan'''

Replace the 555 digits in the following line by the area code of your choice and copy the line, including parenthesis, in the Dialplan field in Line 1 Tab at the bottom of the page:

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

[[Dial Plan for Linksys ATAs|Click here for more information on Linksys dial plans]]

'''Step 6'''

Click on the "Save Settings " button at the bottom of the form.

'''Step 7'''

Switch to the '''SIP''' tab and scroll down to '''RTP Parameters''' and set the follow setting:

'''RTP Packet Size''': 0.020

=='''Configuration Screens'''==

[[File:PAP2T001.JPG|PAP2 Configuration Screen|frame|left]]

[[File:PAP2T002.JPG|PAP2 Configuration Screen|frame|left]]

[[File:PAP2T003.JPG|PAP2 Configuration Screen|frame|left]]

[[File:Pap2sip.jpg|PAP2 Configuration Screen|frame|left]]

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

'''Customer Submitted Information:''' 
For North America: 
Found this link on configuring the PAP2-NA hardware to work better in North America and specifically with VOIP.ms. 
Read the article called: [http://www.toao.net/25-linksys-ata-configuration Configure your Linksys VoIP ATA the right way!]

== How to avoid the long delay to hear the ringtone ==

If you ever experience some delay to hear the ringtone when you make outgoing calls with your PAP2. Changing the PAP2's Interdigit Long Timer value can help resolve the issue. Follow the next steps in order to change that setting:

'''Note''': However before changing that option, test if calling the number with an # at the end of the number works(e.g. 5554441234#). If that doesn't work you need to contact the support staff in voip.ms.

*1- First access the PAP2's web interface.
*2- Click on the '''Admin Login''' and then click on '''(switch to advanced view)'''

[[File:Pap2admlog.jpg|800px]]

*3- Click on the '''Regional''' tab and look for the '''Control Timer Values (sec)''' section.

[[File:Regional.jpg|800px]]

*4- Enter the desire value in the '''Interdigit Long Timer''' field (for example lower this value to 4).

[[File:Ctrl timer values.jpg|800px]]

'''Note:''' There are some PAP2/PAP2T devices in circulation which were originally 'locked' to one provider and subsequently unlocked by end users. Do *not* use the RESET# command with these boxes as you may be locked out of your device's settings by doing so.

== '''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 PAP2's 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]]

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

== See also ==
* [[Dial Plan for Linksys ATAs]]
* You can check the most commonly used Star Codes from [[Cisco/Linksys Star Codes]]

[[Category:Analog Telephone Adapters]]

Cisco Linksys SPA942 NA

2015-04-09T17:01:55Z

Jeff: /* Configuration Details */

Dial Plan for Linksys ATAs

2015-04-09T16:57:55Z

Jeff:

The basic dial plan provided in the configuration samples for the Linksys ATA devices (like [[Cisco Linksys PAP2|PAP2]], [[Cisco Linksys PAP2T|PAP2T]] and [[Cisco SPA2100 Phone Adapter|SPA2100]]) should work with VoIP.ms without any issue.

'''VoIP.ms Recommended Dial Plan''':

:(911S0|310xxxx|<:1'''555'''>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|***xx|*xx.|[3468]11|822|0|00|4xxx|**275*x.|xxxxxxxxxxxx.)

'''Note''': For 7 digit dialing, replace 555 by the area code of your choice.

This guide has been created to help you learn about Dial Plans. They can be customized according to your preferences. Please note that customizing your dial plan is entirely optional.
 

__TOC__

== What is a Dial Plan? ==

The dial plan is a string of characters that determines how entered phone digits are interpreted and transmitted by your ATA Device/Phone. It also determines whether to accept, or reject, a call. A dial plan thus facilitates dialing, and also the blocking, of certain types of calls, such as long distance or international.

'''Please Note''': International calls can also be blocked within your VoIP.ms account.

== Digit Sequence ==

A dial plan contains a series of digit sequences, separated by the | character, entirely enclosed within parentheses. Each time a phone button is pressed, your ATA device will attempt to match the digit sequence in your dial plan.

{| class="wikitable"
|-
! Digit Sequence
! Function
|-
| style="text-align:center;"|0 1 2 3 4 5 6 7 8 9 * #
| You can use any of these characters to represent a pressed phone digit.
|-
| style="text-align:center;"| x
| Any phone digit.
|-
| style="text-align:center;"|[sequence]
| You can enter characters between brackets to create a list of acceptable digits. For example, if you enter the range [1-5], the user may only press the digits from 1 to 5. You can also use individual numbers, and certain other characters, in combination. For example [35-8*] allows the user to press 3, 5, 6, 7, 8 or *.
|-
| style="text-align:center;"|. (period)
| You can use a period to accept zero or more entries of a give digit. For example, '''01.''' allows the user to enter 0, 01, 011 and so on.
|-
| style="text-align:center;"|<dialed:substituted>
| This is used for sequence substitution, you can use this to indicate that certain numbers dialed are replaced by other characters. The ''dialed'' digits can be zero or more characters. For example with this sequence '''<:1555>xxxxxxx''' if the user dial a 7 digit number, the number 1555 is added to the beginning of the sequence. If the user press 6782345, the system transmits 15556782345.
|-
| style="text-align:center;"|, (comma)
| This can be used between digits to play an “outside line” dial tone after a user-entered sequence. For example, with this sequence '''9, 1x.''' an “outside line” dial tone is sounded after the user presses 9, and the tone continues until the user presses 1
|-
| style="text-align:center;"|! (exclamation point)
| You can use this character to prohibit a dial sequence. For example with the sequence '''1900xxxxxxx!''' the system reject any sequence that starts with 1900.
|-
| style="text-align:center;"|S0 or L0
| Overrides the Short or Long inter-digit timer to 0 seconds. For example: '''<:1555>[2-9]xxxxxxS2''' indicates, on a seven-digit local call, first wait two seconds to see if any more digits are dialed - after the delay expires, prefix the number with local area code +1-555 and send it '''1[2-9]xx[2-9]xxxxxxS0''' indicates, if +1-areacode-number are dialed as eleven digits, send them immediately as-is as there is no need to wait for additional dialed digits.
|-
| style="text-align:center;"|P# (where # is the duration of the pause in seconds)
| Pauses # seconds.
|}

== Examples ==

Some examples of dial plan digit sequences:

* To dial any international number without using the 011 prefix.
<:011> [2-9]xxxxxxxx.

You can also accomplish this if you set the Dialing Mode to E164 in your [[Account Settings]]

* To block a call to a specific area code (replace 555 with the area code you want)
<:1> 555 xxxxxxx !

* The next sequence, allows you to dial your [[Phone book]] entries using an speed dial like the POTS provider's. For example if you dial 20# the system will send *7520
<:*75>xx< # : >

== See also ==
ATA's
* [[Cisco SPA2100 Phone Adapter]]
* [[Cisco SPA2102 Phone Adapter with Router]]
* [[Cisco Linksys PAP2]]
* [[Cisco Linksys PAP2T]]

IP phones
* [[Cisco SPA504G Phone|Cisco SPA300/500-series 'phones]]
* [[Cisco Linksys SPA942 NA]]

Networking devices
* [[Cisco WRP400]]

[[Category:Guides]]

== References ==
* [http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf Cisco - Administration Guide: SPA2102, SPA3102, SPA8000, SPA8800, PAP2T analogue telephone adapters]
* [http://www.cisco.com/en/US/docs/voice_ip_comm/csbpipp/ip_phones/administration/guide/spa500_admin.pdf Cisco - Administration Guide: Cisco SPA300/SPA500 series and Cisco Wireless-G IP phones]

[[Category: SIP devices]]

Polycom SoundPoint IP 501

2015-03-25T15:32:21Z

Jeff: /* Configuring Voicemail Messages */

[[File:Polycom501.jpg|258px|thumb|left|Polycom SoundPoint IP 501]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Details==

'''Get the IP'''

Start the Phone and it will get an IP from the DHCP Server, then it will display the IP on the Screen.

'''Configure Settings'''

* Enter the IP address on your web browser (PC and Device should be on the same network)
* The browser should load the configuration page for the IP 501
* Click on Sip Link
* Enter User and Password (default values are user: Polycom, pass: 456)
* Enter the Following values:

[[File:Poly501.gif|Settings]]

* Click on Submit. This will save settings and reboot unit.
* After the Device has rebooted, click now on Line Link and set your Identification values:

[[File:Poly501b.gif|Settings]]

* Click on Submit. This will save settings and reboot unit.
* After the Device has rebooted, your Device will be ready to be used with VoIP.ms

''NOTE'': If you have issues reaching an internal extension, make sure to add the following string in your dial plan: [1-9]xx

==Configuring Voicemail Messages==

Please go to the Message Center Section and put the following values.

Subscriber: [blank]

Callback Mode: Contact

Callback Contact: *97

[[category:SIP phones]]

Please check this link to get rid of the MWI Audible Sound http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio

Virtual Fax

2015-03-03T18:32:43Z

Jeff: /* Important information to know about the Virtual Fax Service */

[[File:Faxhomelogo.png|center]]

<div style="text-align:center; font-size: 150%;color:red;"><ins>'''This Feature Is Currently Available For Beta Customers</ins> </div>

<div style="text-align:center; font-family:Arial;"><ins>''' You can sign up for Beta testing by clicking [https://www.voip.ms/m/beta.php Here]'''</ins> </div>

The Virtual Fax feature is used for sending and receiving a Fax (facsimile) with the VoIP.ms service using a DID number specifically dedicated to Faxing. You may obtain such a number from your Customer Portal in the Fax Numbers section under the ''Order DID(s)'' of the ''DID Numbers'' menu.
Regular voice DID numbers are not compatible with the Virtual Fax feature.

__TOC__

== Important information to know about the Virtual Fax Service ==

* '''The Virtual Fax Service is a Beta release, the features are there and working, issues have been identified and are being or have been fixed. But still it is important to note that we recommend to not be dependent solely on this feature for your critical needs (or mass faxing needs as there is a daily limit) while it is still in the Beta phase. It is important to us that you report any issues with this service by sending an email to support@voip.ms so that the developers can get involved if necessary.'''
* '''The Virtual Fax Service is only available for U.S. and Canadian DID Numbers specifically acquired from the Fax Numbers ''Order DID'' section. Porting in your own number for Fax is not possible at this time, this is being worked on however there is no ETA as of yet.'''
* '''The Service can only be used to send Messages to Canadian and U.S. Numbers at this time. We also cannot guarantee that International will be properly received.'''
* '''There are currently limits for the Fax Beta Program. At this time the limits are : Customers can only purchase 2 numbers and can only send 20 faxes per day.'''

== Virtual Fax DID Number ==

Virtual Fax works specifically with Fax Numbers only acquired from the VoIP.ms Customer Portal for the moment. 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]]

== 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:
*Contact: This is where you will put the destination number.
*From Name: Here you will put the name to send in the Fax header.
*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.
*From Number: Select the Fax DID number from which you will send your Fax.
*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: Choose a file to send as a Fax. The file must be in pdf, txt, jpg, gif, png or tif

[[File:Sendafax2.png]]

== My Faxes ==
In this section of the Virtual Fax menu you will be able to view your Inbound and Outbound Faxes. You may select a date range and choose the folder you would like to view. Click 'Get My Faxes' to view your selection. You can view the Status of each Fax and select from several Actions. You can select to View the Fax directly, Download the Fax, Email the Fax to an address of your choosing or alter the location of the Fax by moving it to another folder.

[[File:Myfaxes2.png]]

== My Folders ==

In the 'My Folders' section you can create folders by typing in the folder name of your choosing under 'New Folder' and clicking 'Create'.
You will have an overview of your Folders, see the date they were Created, the amount of Faxes in each Folder and be able to Edit the Folder or Delete it.
Any Faxes contained in a created folder will revert back to either the INBOX or SENT folder if the created folder is deleted.

[[File:Faxmyfolders2.png]]

== 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:Myfaxnumbers2.png]]

Grandstream HandyTone 702 - HT702

2015-02-20T16:40:34Z

Jeff:

== GENERAL INFORMATION ==

The Grandstream HandyTone 702/7XX series is the latest in the HandyTone line. The 702 is a reliable, inexpensive telephone adapter which works with the VoIP.ms service when placed after your broadband internet router.

'''Websites:'''
[http://www.grandstream.com/index.php/products/ip-voice-telephony/consumer-analog-telephone-adaptors/ht702_704 HT702 Product Page]

'''Help / Support:'''
[http://www.grandstream.com/support/firmware Grandstream Support]

[[File:Ht702.jpg‎]]

== Configuring the Handytone 702 ==

These instructions are based on HandyTone 702 software version 1.0.0.18; if you are running a different software version some menus and settings may be different.
 
These instructions are also based on using the HandyTone in its factory default configuration, which obtains a dynamic IP address automatically from your router using DHCP. For information on configuring your HandyTone with a Static IP Address, please refer to the HandyTone user´s manual.

Each step is important in assuring that your device works properly.

:''We recommend that you read each step through in its entirety before performing the action indicated in the step.''

'''Step 1''' 

Connect your HandyTone to your router with the supplied Ethernet network cable.

Now connect your phone to the HandyTone. Plugging the cable into the correct FXS Port that you configure.

Finally plug the supplied power cable into the HandyTone.
 

'''Step 2''' 

: Wait 60 seconds after plugging your HT702 in.

Pick up the phone connected to the HT702 and dial the * key on your phone 3 times.

Please have a pen and paper ready. You will hear a message - "Enter a menu option", then enter 0 2 on your phone. You will now hear a message giving you the IP address of your HT702 such as - "192.168.001.010" and write this number down.

Open a web browser on your computer such as Chrome or Firefox and enter the IP address you heard in step 4 as the address (I.E. where you would normally enter www.voip.ms).

Please note: Some browsers will require you to remove leading zero's ( 0 's ) in the IP address. For example if you heard "192.168.001.010" you should change this to "192.168.1.10".

The Interface has a timeout so please make changes quickly or save/update your settings every couple of minutes.
 

'''Step 3''' 

: You should now see a page that looks like this: 
[[File:Grandstream_702_01.png‎]]

Enter the password for the HT702 in the password field. The default administrator password for the HT702 is "admin" (without quotes).

After entering the password you should see a screen that looks similar to the one below:

[[File:Grandstream_702_02.png‎]]
 

'''Step 4''' 

Now, click on FXS PORT1 and configure your settings accordingly (as shown below):

Please use the same server in the Failover SIP Server as your Primary SIP Server or leave the Failover SIP Server field Blank.

[[File:GrandstreamConfigPage.png‎]]

Please use the following settings to configure your VoIP.ms account:
'''
Configuration Page Settings:'''

Primary SIP Server: servername.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server VoIP Servers])

* You can also find this information by logging into your [https://www.voip.ms/m/accountinfo.php Customer portal].

Failover SIP Server: (Please leave this Blank)

Outbound Proxy: servername.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server VoIP Servers])
* You can also find this information by logging into your [https://www.voip.ms/m/accountinfo.php Customer portal].

NAT Traversal (STUN): No, but send keep-alive

SIP User ID: (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)

Authenticate ID: (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)

Authenticate Password: ****** (Use the SIP account password - By default this is the same as the Customer Portal)

DNS Mode: SRV

SIP Registration: Yes

Unregister On Reboot: Yes

Outgoing Call Without Registration: Yes

Register Expiration: 3

Allow Incoming SIP Messages from SIP Proxy Only: Yes

Preferred DTMF method: In-audio, RFC2833

Enable Call Features: No

Dial Plan: {[x*]+}

Preferred Vocoder: PCMU, G729, PCMA
 

'''Step 5''' 

: Once you have configured the settings above, click the Update button and then the Reboot button to save the configurations.

Your HT702 will power cycle after you click the reboot button. Please wait at least 30 seconds for the unit to finish power cycling. If you see that the Phone 1 LED (or phone 2 LED, depending on which FXS port you've configured our service for) is a solid green color, then your unit is configured and ready to make calls.
 

'''Step 6''' 

: That's it! You can now make a phone call.

1 + the area code and number for calls to the US & Canada

Or

011 + the country code, area code, and number for calls worldwide (you may also use 00 instead of 011).
 

== Preventing Direct IP calls like 100 & 1000 ==

To Prevent Direct IP calls to your device and only allow calls from our service please enable the following 2 options in your FXS Port Configuration Page.

'''Check SIP User ID for incoming INVITE''' - Default is No. Check the incoming SIP User ID in Request URI. If they don’t match, the call will be rejected. If this option is enabled, the device will not be able to make direct IP calls.

'''Allow Incoming SIP Messages from SIP Proxy Only''' - Default is No. Check the incoming SIP messages. If they don’t come from the SIP
proxy, they will be rejected. If this option is enabled, the device will not be able to make direct IP calls.

== Auto Provisioning ==

Some newer models of the HT702 now have Auto Provisioning and will delete the changes you make in setting up the device to use our service. Please go to your Graphical user interface and go to 'Advanced Settings' tab and look for "Firmware Upgrade and Provisioning" and disable it.

''Keywords: Handytone 701, HT701, Grandstream''

[[Category:Analog Telephone Adapters]]