Monday, 22 July 2013

OBi202 - Configure Google Voice, Voip.ms, CallCentric, FreePhoneLine.ca Simultaneously


In this article I will be illustrating how to manually configure the Obihai OBi202 VoIP ATA with four (4) completely independent and separate VoIP service providers. If I include the default OBiTalk VoIP network, the OBi202 can make and receive calls directly over five (5) independent VoIP networks, all simultaneously configured and functional in one little box.

The four VoIP services that I will be manually configuring into the OBi202 are:  Google Voice, Voip.ms, CallCentric, and FreePhoneLine.ca.

Because I'm located in Canada, and testing the OBi202 from an Internet service provider located in Canada, I will be presenting the Google Voice portion of this article primarily for the benefit of Canadians who would like to know how to configure their own Obi202 VoIP adapter for service with Google Voice, and enjoy free USA/Canada phone calling without needing to be tied to a PC or laptop to make those calls. Unfortunately, the one thing we Canadians can't do that our American friends have been doing for quite some time is to obtain a GV phone number for incoming calls.  But, we can at least still make outgoing calls over the GV network for free (at least until the end of 2013).



As well, I will be presenting a completely "manual" method of configuring the OBi202 with all four VoIP providers mentioned above.  My procedures that follow will not require any linking or direct connection to the OBiTalk administration portal (at www.obitalk.com) and will not require remote administration via the OBiTalk network.  All my administration of the OBi202 will be via my local LAN network.  This article is about how I do it manually, the DIY way.

So let's get started...


UPDATE: (Nov. 3, 2013)
As of May 15, 2014 Obihai ATA's will no longer be able to make and receive calls via Google Voice.


To achieve my goal of quick and reliable setup, I will recommend that the ATA be initially "factory reset" to it's default values (this way, we're starting on the same page, literally).  Then, I will point out only the essential settings that "must be edited" in order to connect and achieve service registration with Google Voice, Voip.ms, CallCentric, and finally FreePhoneLine.ca.  As well, I have recommended a few additional optional settings that will generally improve VoIP service registration reliability.

I have personally configured my OBi202 with these four services exactly as I'm illustrating in this article.  And, I'm assuming that if they all work for me, hopefully they will work just as well for you, too.  (Bearing in mind that I'm in Canada and Google doesn't issue GV phone numbers for Canadian accounts - thus, we Canadians can't receive incoming calls via a GV phone number - but, we can still make GV outbound calls, just fine.)

                                                    ---------------------------------
IMPORTANT NOTE:
My procedures outlined here are based on the premise that your OBi202 is NOT, and will not be, linked to the OBiTalk web portal at www.obitalk.com using their "Add Device" feature.  If you "add your device" to the OBiTalk portal, the portal will takeover administration of your OBi device by remotely administering your OBi202 via the portal (which is fine, if that is what you want to do, but this tutorial is not based on that premise). Any local administration settings you do locally to your OBi202, via your web browser via your LAN side administration, will be over-ridden by the OBiTalk remote administration portal periodically, or upon device reboot.  Thus, to proceed with my procedures here, and if your OBi202 is/was previously configured via OBiTalk portal, then I recommend you "UNLINK" and remove your OBi202 from OBiTalk portal (remove from "Add Device" list) first, and then followed by a factory reset to ensure you are starting with default configuration values.

Remember, the premise of my tutorials are to allow you to locally and manually configure your OBi202 without interaction with the OBiTalk remote administration portal at www.obitalk.com.  (My goal is to show you how to DIY without any 3rd party interaction.)

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

Because the OBi202 shares many features of the OBi100, I'm going to suggest that it may be worth while for you to review some of my previous articles about configuring the OBi100 with Google Voice, Voip.ms, and CallCentric.
                                                    ---------------------------------


In order to proceed with the quick and easy setup procedures, a few assumptions must be made and prerequisites met:

Assumptions and Prerequisites
  • You need to know how to login to the OBi202 -  I explain how:  HERE.
  • The OBi202 has not been locked-down by a VSP (VoIP Service Provider).
  • The OBi202 is not linked to OBiTalk Portal via "Add Device" method on OBiTalk website.
  • Make sure the OBi202 is directly Ethernet connected to your ISP supplied Internet broadband Modem.
  • By default, the OBi202 will be operating in NAT-Router Mode.  If you plan to use it behind another NAT-Router, best performance may be obtained if you change it to operate in Bridge Mode.
  • You have an analog telephone plugged into the ATA Phone-1 port, or Phone-2 port.  (Actually, I'm testing with two separate phones; one connected to each port.)
  • You know how to perform a Factory Reset (Reset Configuration to Defaults) - I explain how:  HERE.
  • Once you know how to perform a Factory Reset, I recommend that you do so, before proceeding.
    (This is an important step to be sure that what works for me will work for you, too.)
  • Your OBi202 is running the latest Firmware version.  If not, I explain how to check/update:  HERE.
    At time of this article, I was running Firmware/Software version:  3.0.1 (Build: 3932).
  • Your OBi202 is powered-up and ready to be configured.
  • Test that your OBi ATA is working properly by placing an ECHO TEST call to the Obihai Echo Test Server, by dialing **9-222-222-222 from your telephone connected to the OBi202.  (All OBi devices can magically do this without any configuration changes from the factory defaults.)
  • You have setup a Gmail account and have your login username and password handy.
  • You will need to validate your Gmail "Call Phone" feature by making a phone call from your PC to a landline/mobile number.  Once you have done that successfully, you are ready to configure your OBi ATA for Google Voice access.


    NOTES:    It may be advisable to setup a separate Gmail account just for use with your OBi ATA.
    As well, some people feel more secure setting up their Gmail (or, Google Apps, which I use) accounts to use "2-Step Verification".  And, if you use 2-step verification, alternate login devices such as with your OBi202 will require that you setup an "Application Specific Password" for account login from that alternate device.  While, I use 2-step verification (and have been for years now), you are not required to do so in order to use Google Voice with your OBi202 ATA.  I'm just letting you know that the OBi202 will work with Google Voice even if your account is protected with 2-step verification AND an application specific password. (If 2-Step Verification is enabled on your Gmail account, then you will be required to use an Application Specific Password to login with the OBi202 ATA)

  • To configure the OBi202 with Voip.ms, you should already have opened an account with Voip.ms and know your SIP username and password.  I myself created a Voip.ms Sub-Account just for testing my OBi202.  (Although, I have had an account with Voip.ms for many years.)
  • To configure the OBi202 with CallCentric, you should have opened a CallCentric account and know your CallCentric account # and SIP password.  CallCentric now allows for the creation of Sub-Accounts.  I created a Sub-Account just for testing my OBi202.  (I have also had accounts with CallCentric for many years, too.)
  • To configure the OBi202 with Fongo's FreePhoneLine.ca, you will already need to have an account with them.  And, to login with a BYOD VoIP ATA like the OBi100 or OBi202, you will be required to have already purchased their "VoIP Unlock Key".  For a one-time $50 fee, Canadians with FreePhoneLine.ca account will be allowed to make unlimited free calls across Canada to any landline or mobile phone located in the Fongo/FreePhoneLine.ca free calling zones (All major Canadian cities).  Currently, there is no time limit as to how long the unlock key allows unlimited access to your Fongo/FreePhoneLine.ca SIP account.

    Have your FreePhoneLine.ca VoIP Unlock Key credentials handy for device configuration.

Now that we have all the assumptions and prerequisites out of the way, all that remains is to configure the non-default essentials required to successfully login/register the OBi202 with your Gmail (or Google APP's) Google Voice account, Voip.ms, CallCentric, and FreePhoneLine.ca VoIP proxy servers.


The Plan of Attack
Each service provider to be configured will be paired-up between an ITSP Profile(X) and a specified Voice Service SP(X).  I have paired each service as follows:

  1. Google Voice:             ITSP Profile A and SP1     (default primary line, or  **1 )
  2. Voip.ms:                     ITSP Profile B and SP2               ( **2 )
  3. CallCentric:                 ITSP Profile C and SP3               ( **3 )
  4. FreePhoneLine.ca:      ITSP Profile D and SP4               ( **4 )
  5. ObiTalk Network:       Enabled by default                       ( **9 )
By default, SP1 is the Primary Line for outbound calling.  By default, any number you dial will be routed through the SP1 configured service provider, unless you specifically specify another line to be used.  (i.e. precede number dialing with **2,  **3,  **4,  or **9)

So, lets get started:
  • Login to the OBi202 configuration utility using your PC web browser.


**1
Configuring Google Voice as Profile A and SP1
  • Click on the Service Providers menu option, located in the left-hand navigation column.
  • Click on "ITSP Profile A" menu option
  • Click on "General" menu option.
  • Set the Signaling Protocol to:  Google Voice  (first "uncheck" the Default check box next to it)
  • Click the Submit button at the bottom of the window.

  • When a pop-up dialog window appears, click the OK button to proceed.


  • You will then be greeted with the Configuration Update Successful screen.

    Once you clicked OK, you could reboot, however, the previous submission will temporarily be saved until we finish making more changes, and then we will Reboot, once finished all configuration changes at the end of this article.
  • Next, click on Voice Services menu item in the left navigation window pane.

  • Then, click on "SP1 Service" menu item, located under Voice Services.

  • Locate the SIP Credentials section about midway down th SP1 Service page.
  • In the text box labeled AuthUserName, enter your Gmail or Google APP's User Name.  This is your account email address.  (i.e.  yourname@gmail.com  or,  yourname@yourGoogleAPPSdomain.com if using Google Apps like I do).
  • In the text box labeled AuthPassword, enter your Gmail or Google APP's password.

NOTE:
If you are using 2-Step-Verification, you will need to enter your Application Specific Password that you manually generate in your Google Account Settings Security section.  I myself am using an application specific password generated specifically for my OBi202 login.

  • After entering your login credentials, click the Submit button at the bottom of the SP1 Service page:
  • Next, click on the OK button to proceed with submitting the changes:

  • Once again, you will be greeted with the Configuration Update Successful window screen.

    After you clicked OK, you could reboot to initialize the new settings.  However, the previous submissions will temporarily be saved until we finish making more changes, and then we will Reboot at the end of all changes.



**2
Configuring Voip.ms as ITSP Profile B and SP2

  • Click on the ITSP Profile B sub-menu option to expand its sub-menu list.
  • Click on "General" sub-menu option, under ITSP Profile B.
  • Observe that the Signaling Protocol is set to:  SIP  (as it should be, by default).
  • Just for your own reference purposes, you can fill in the "Name" field with "Voip.ms", or whatever you want to label this ITSP configuration profile.  (optional)

The OBi202 default Digit Map will only partially accommodate Voip.ms dialing needs.

This is the Default Digit Map already set in the OBi202:
(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*#]@@.)


  • I'm recommending the following Digit Map for use with Voip.ms:
(*xxx.S2|**275*x.|911|4xxx|1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*#]@@.)


  • Clear the existing default OBi202 Digit Map.  (First, un-check its associated default check box.)
  • Copy and past my recommended digit map in place of the default digit map. 


Optional Settings:   STUNEnable and STUNServer
You may want to Enable STUN Server support and specify a STUN server to assist with VoIP NAT traversal issues.  This is an optional setting, but is often very useful if you experience incoming call issues.

I typically will specify stun.counterpath.net or stun.3cx.com as the STUN server name to use.
  • click the Submit button at the bottom of the ITSP Profile B page:
  • Next, click on the OK button to proceed with submitting the changes:

  • Once again, you will be greeted with the Configuration Update Successful window screen.

    After you clicked OK, you could reboot to initialize the new settings.  However, the previous submission will temporarily be saved until we finish making more changes, and then we will Reboot at the end of all changes.
  • Now, click on the SIP sub-menu item, also listed under ITSP Profile B.
    Here, you will be setting the Proxy and Registrar server values for Voip.ms.


While in ITSP Profile BSIP settings window, edit the following fields with city.voip.ms as the Proxy and Registrar server:
Proxy Server:                toronto.voip.ms      (choose one of  ~13 server cities)
Registrar Server:          toronto.voip.ms      (should match the Proxy Server selected)

Note:
I'm using the Seattle server.  Users of Voip.ms will typically be selecting a proxy server nearest their own location, or which ever one provides the lowest ping route latency.  While I'm located in British Columbia, Canada, I could be using the Toronto server.  But, I get much lower latency by connecting via the Seattle servers. (However, I do wish Voip.ms could commission a Vancouver, BC server, too.  Or, perhaps a Calgary, AB server.... :)


    Note:
    If you purchased a DID phone number with Voip.ms, that DID phone number will be associated with one of these POP Proxy servers in your Voip.ms account settings.  To receive your incoming DID phone calls reliably, be sure that the DID POP server setting matches the server you program into your VoIP ATA. 

    • When finished editing the server values, scroll down to the bottom of the page and click the
      Submit button.

      When the "Are you sure you want to submit changes?" dialog box pop's up, click its OK button to proceed.   
    • Once again, you will be greeted with the Configuration Update Successful window screen.
     (Again, you do not need to reboot until we are finished with all editing, later.)
    • Now, click on the Voice Services menu option, making sure its options list is expanded.
    • Next, click on SP2 Service sub-menu option.
    • While in the SP2 Service settings page, change the Service Provider Profile setting from A to B:

      X_ServProvProfile:   B
      This setting will correlate the ITSP Profile B settings to the SP2 service provider settings for our service configuration with Voip.ms.
    • Optional Settings:   X_KeepAliveEnable
      Enabling Keep Alive can be useful for reducing the chances of loss of incoming calls and audio. Enabling Keep Alive essentially triggers periodic querying of the proxy/registrar servers to keep the incoming VoIP RTP ports open on the ATA for incoming calls and audio.


    • Scroll down to the SIP Credentials section of SP2, and set the following login values:

      AuthUserName:     xxxxxx_xxxx   (Your Voip.ms Main Account, or Sub-Account Number)
      AuthPassword:        xxxxxxxxxx           (Your Voip.ms SIP password - minimum 6 characters)

    • Next, scroll down to the "Calling Features" section near the bottom portion of  SP2 page.

    Optional Voice Mail Notification Settings
    If you have setup and enabled Voice Mail with your Voip.ms account, you may want to Enable Voice Mail Waiting Notification by clicking the following check-boxes.
    • MWIEnable                 (checked)        
    • MWIEnable2               (checked)
    • X_VMWIEnable          (checked)      
    • X_VMWIEnable2        (checked) 



    • When finished editing the SP2 page for Voip.ms, scroll down to the bottom of the page and click the Submit button.
    • Next, click on the OK button to proceed with submitting the changes:

    (You do not need to reboot, just yet, until finished submitting all changes.)



    **3

    Configuring CallCentric as ITSP Profile C and SP3

    • Click on the ITSP Profile C sub-menu option to expand its sub-menu list.
    • Click on "General" sub-menu option, under ITSP Profile C.
    • Observe that the Signaling Protocol is set to:  SIP  (as it should be, by default).
    • Just for your own reference purposes, you can fill in the "Name" field with "CallCentric", or whatever you want to label this ITSP configuration profile.  (optional)
    While you may get-by with the OBi202 default Digit Map, you will only have limited dialing functionality with CallCentric's service dialing options.  CallCentric recommends the following modified Digit Map for use with the OBi202:

    (*xx.|**275*x.|[3469]11|1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*#]@@.)

    • Clear the existing default OBi202 Digit Map.  (First, un-check its associated default check box.)
    • Copy and past the above recommended digit map in place of the default digit map. 


    Optional Settings:    STUNEnable and STUNServer
    You may want to Enable STUN Server support and specify a STUN server to assist with VoIP NAT traversal issues.  This is an optional setting, but is often very useful if you experience incoming call issues.
    I typically will specify stun.counterpath.net or stun.3cx.com as the STUN server name to use.

    • Click the Submit button at the bottom of the window, before proceeding to the next step.
      When the "Are you sure ..." dialog box pop's up, click its OK button to proceed.
      (You do not need to reboot, just yet.  You can wait until finished submitting all changes.)
    • Now, click on the SIP sub-menu item, also listed under ITSP Profile C.
    • Here, you will be setting the proxy and registration server values for CallCentric.

      The server settings and values are as follows:
    • ProxyServer:               callcentric.com
      RegistrarServer:         callcentric.com
      UserAgentDomain:     callcentric.com
      Outbound Proxy:         callcentric.com

    • After editing the proxy servers, etc.,  scroll down near the bottom of  SIP page of ITSP Profile C page.

    CallCentric Optional Settings:
    CallCentric recommends enabling Proxy Server Redundancy and Secondary Registration to assist with server fallback reliability and fail-over.

    CallCentric is somewhat unique in the way they distribute their VoIP servers.  All registrations are done just using their main domain name of callcentric.com. (No specific sub-domain names for each SIP server like many other VoIP services do.)   But, in the background they utilize an array of servers which are dynamically load-balanced for all incoming SIP traffic with the help of DNS lookup.
    • To facilitate proxy and registration redundancy, check and enable the following settings:
      X_ProxyServerRedundancy    (check to enable)
      X_SecondaryRegistration        (check to enable)



    • When finished editing the server settings, scroll down to the bottom of the page and click the Submit button.
    • Next, click on the OK button to proceed with submitting the changes:

    • Once again, you will also be greeted with the Configuration Update Successful window screen.

      After you clicked OK, you could reboot to initialize the new settings.  However, the previous submission will temporarily be saved until we finish making more changes, and then we will Reboot at the end of all changes.  


    • Now, click on the Voice Services menu option, making sure its sub-menu options list is expanded.
    • Next, click on SP3 Service sub-menu option.
    • Change the Service Provider Profile settings from A to C:
      X_ServProvProfile:     C

    • Optional Settings:   X_KeepAliveEnable
      Enabling Keep Alive can be useful for reducing the chances of loss of incoming calls and audio. Enabling Keep Alive essentially triggers periodic querying of the proxy/registrar servers to keep the incoming VoIP RTP ports open on the ATA for incoming calls and audio.


    • Now, in the SIP Credentials section, set the following login values:

      AuthUserName:     1777xxxxxxx   (11-digit account number, or 14-digit Sub-account number.)
      AuthPassword:     xxxxxxx       (Your CallCentric SIP password - minumum 6 characters)
    • Scroll down the  SP3 page to the section labeled Calling Features.

    Optional Voice Mail Notification Settings
    • If you have setup and enabled Voice Mail with your CallCentric account, you may want to Enable Voice Mail Waiting Notification by clicking the following check-boxes (found under the "Calling Features" section)
    • MWIEnable                 (checked)        
    • MWIEnable2               (checked)
    • X_VMWIEnable          (checked)      
    • X_VMWIEnable2        (checked) 

    • When finished editing the CallCentric SP3 settings, scroll down to the bottom of the page and click the Submit button.
    • Next, click on the OK button to proceed with submitting the changes:

    • Once again, you will also be greeted with the Configuration Update Successful window screen.
    (You do not need to reboot, just yet, until finished submitting all changes.)




    **4

    Configuring FreePhoneLine.ca as ITSP Profile D and SP4

    For Canadians who previously used the Dell Voice and Fongo SIP VoIP Unlock Key to access Canada wide free calling, those services have now been transitioned over to their sister service FreePhoneLine.ca.

    I originally started with a Dell Voice - Fongo account and purchased their "VoIP Unlock Key".  The following outlined settings are working just fine for me on my OBi202 with FreePhoneLine.ca configuration.

    • Click on the Service ProviderITSP Profile D sub-menu option to expand its sub-menu list.
    • Click on "General" sub-menu option, under ITSP Profile D.
    • Observe that the Signaling Protocol is set to:  SIP  (as it should be, by default).
    • Just for your own reference purposes, you can fill in the "Name" field with "FreePhoneLine-Fongo", or whatever you want to label this ITSP configuration profile.  (optional)

    The OBi202 default Digit Map will only partially accommodate FreePhoneLine dialing needs.


    This is the Default Digit Map already set in the OBi202:
    (1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*#]@@.)

    • I'm recommending the following Digit Map for use with FreePhoneLine.ca:  (*xx|911|1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*#]@@.)
      This digit map will allow you to also access Voice Mail (*98) and 911 emergency dialing.
    • Clear the existing default OBi202 Digit Map.  (First, un-check its associated default check box.)
    • Copy and past the above recommended digit map in place of the default digit map. 


    Optional Settings:    STUNEnable and STUNServer
    You may want to Enable STUN Server support and specify a STUN server to assist with VoIP NAT traversal issues.  This is an optional setting, but is often very useful if you experience incoming call issues.

    I typically will specify stun.counterpath.net or stun.3cx.com as the STUN server name to use.

    • When finished editing this page, click the Submit button at the bottom of the window, before proceeding to the next step.

      When the "Are you sure ..." dialog box pop's up, click its OK button to proceed.
      (You do not need to reboot, just yet, until finished submitting all changes.)
    • Now, click on the SIP sub-menu item, also listed under ITSP Profile D.
      Here, you will be setting the Proxy and Registrar server values for FreePhoneLine.ca.


    While in ITSP Profile DSIP settings window, edit the following fields with voip.freephoneline.ca, or voip2.freephoneline.ca as the Proxy and Registrar server: 

    • Proxy Server:                voip.freephoneline.ca      (or, voip2.freephoneline.ca )
    • Registrar Server:          voip.freephoneline.ca      (should match Proxy Server used)


    The OBi ATA's default to a Registration Period of only 60 seconds.  Typically, many industry standard ATA's will default to a registration period of 3600 seconds.  Fongo/FreePhonLine.ca is recommending 3600 seconds.  However, many VoIP users have found 3600 seconds to not always be frequent enough to maintain reliable registration.  Thus, I often tend to compromise somewhere around 360 seconds registration intervals.


    • When finished editing the server values, scroll down to the bottom of the page and click the
      Submit button.   (However, do not reboot just yet)

      When the "Are you sure ..." dialog box pop's up, click its OK button to proceed.

      (You do not need to reboot, just yet, until finished submitting all changes.)
    • Now, click on the Voice Services menu option, making sure its options list is expanded.
    • Next, click on SP4 Service sub-menu option.
    • Change the following PROFILE setting from A to D:
      X_ServProvProfile:     D
    • Optional Settings:   X_KeepAliveEnable
      Enabling Keep Alive can be useful for reducing the chances of loss of incoming calls and audio. Enabling Keep Alive essentially triggers periodic querying of the proxy/registrar servers to keep the incoming VoIP RTP ports open on the ATA for incoming calls and audio.


    • Now, in the SIP Credentials section of SP4, set the following login values:

      AuthUserName:   xxxxxxxxxxx (Your 11-digit  DV, Fongo, or FPL account phone #)
      AuthPassword:      xxxxxxxxxx   (Your DV, Fongo, or FreePhoneLine.ca  SIP password )
    • Scroll down to the bottom portion of  SP4 page.

    Optional Voice Mail Notification Settings
    I have never been able to get the Fongo/FreePhoneLine.ca voice mail notifications to work, thus I am not bothering to enable Message Waiting MWI settings for this account.
    (Message waiting works just fine for Voip.ms and CallCentric accounts, but not Fongo/FPL).



    • When finished editing, click the Submit button at the bottom of the page.
    • Next, click on the OK button to proceed with submitting the changes:



    Now, to finalize all setting changes, we are ready to Reboot the OBi202.

    This is the final step to complete our  OBi202 configuration settings for service with Google Voice, Voip.ms, CallCentric, and FreePhoneLine.ca VoIP accounts.   
    • Click the REBOOT button located in the window upper-right corner.


    Rebooting your OBi202 ATA only takes about 30 seconds to complete.

    * * *
    I highly recommend that you go back and review all changed settings and that they have been set correctly.  With this many accounts and configuration changes....... it's very easy to make a mistake.
    * * * * *

    If you had all your SIP account usernames and passwords handy for GV, Voip.ms, CallCentric, and FreePhoneLine.ca, it is possible to completely configure all these accounts into the OBi202 in about 30 minutes, or so.  I verified this information by first performing a Factory Reset and then following the above outlined procedures.  It's all working for me without a hitch.

    Obviously, there are numerous additional settings that are possible for tweaking the configuration and performance of your ATA.  But, configuring and enabling the bare bones basics was the theme of this article.


    Testing The OBi202 For Outbound Calling
    I have provided some phone numbers for testing your manually multiconfigured OBi202.

    By default, SP1 is the Primary Line for outbound calling.  By default, any number you dial will be routed through the Profile-A and SP1 configured service provider, unless you specifically specify another line to be used.  (i.e. precede number dialing with access code **2, **3, **4, or **9)

    1. Google Voice:             ITSP Profile A and SP1     (default primary line, or  **1 )
    2. Voip.ms:                     ITSP Profile B and SP2               ( **2 )
    3. CallCentric:                 ITSP Profile C and SP3               ( **3 )
    4. FreePhoneLine.ca:      ITSP Profile D and SP4               ( **4 )
    5. ObiTalk Network:       Enabled by default                       ( **9 )

    **1
    Testing Outbound Calls Using Google Voice In Canada

    By default, any outbound phone number you dial will automatically be routed via your Google Voice account.

    All Phone numbers listed below are Free Calls via your Google Voice connected OBi202 phone:
    (At least free through 2013...)


    Canadian Weather Lines:

    • Victoria, BC     wx:        250-363-6717
    • Calgary, AB     wx:        403-299-7878
    • Edmonton, AB  wx:       780-468-4940
    • Regina, SK       wx:       306-780-5744
    • Winnipeg, MB   wx       204-983-2050
    • Montreal, QC    wx:      514-283-4006

    Other Test Numbers:

    • National Research Council, Canada - Talking Clock:    613-745-1576
    • Google411 (decommissioned; but, still fun to call):    800GOOG411  (800-466-4411)
    • iNum Echo Test:
       1.  Dial the Vancouver, BC  iNum gateway number:  778-786-3497
       2.  At the iNum IVR voice prompt, enter the iNum Echo Test number: 883-5100-0000-0091

    NOTE:
    The Canadian numbers listed above are also free calls from a Fongo or FreePhoneLine.ca account.

    NOTE on Google Chat Caller-ID associated with outbound calls:
    Because, here in Canada we cannot get a Google Voice inbound PSTN style DID phone number, you need to know that all your outbound Google Voice initiated calls will show a California Caller-ID as the Caller-ID associated with your outbound phone calls.  Currently, my outbound Google Voice calls are tagged as coming from:  Escondido, Ca - 1-760-705-8888.  
    Here is Google's online-help related confirmation on this Caller-ID number:  Here



    **2

    Testing Calls via Voip.ms

    Precede all outbound calls intended for routing through your Voip.ms account with:  **2, followed by the number you wish to call.

    All numbers listed below are Free Calls, via Voip.ms and your OBi202 ATA phone:
    • Voip.ms Echo Test number:   4443       e.g., dial as:  **24443
    • Voip.ms DTMF Test:   4747        e.g., dial as:  **24747
      Once connected, press any digit on your phone and the service will "talk-back" the digit pressed.
    • Checks your Voip.ms calling credits balance:   *225     e.g.,  dial as:   **2*225
      This feature must explicitly be enabled for each sub-account used.  (Disabled, by default on sub-accounts.)
    • Call your Voip.ms Voice Mail box (if you enabled it via your Voip.ms dashboard):   *97
             e.g., dial as:   **2*97
    • iNum network Echo Test:   011-883-5100-0000-0091#
      e.g.,  dial as:     **2011883510000000091#
    • iNum network Caller-ID Talk-Back:   011-883-5100-0000-0093#
      e.g.,  dial as:     **2011883510000000093#
    • SIP Broker peering network Test Announcement:    **275*-0111-88888#
      e.g.,   dial as:     **2**275*011188888#
    • e911 activation validation Test:   1-555-555-0911     e.g.,  dial as:     **215555550911
      This is not a 911 call.  It only checks the status of your e911 service activation with Voip.ms.

    Voip.ms Service Notes:
    1. 1800 numbers may not be free via Voip.ms. You may need calling credits. (Depends on your Voip.ms account Toll Free Routing preferences settings.  ("Value" vs. "Premium" routing)
    2. Calling to PSTN landlines and mobiles requires Voip.ms calling credits.
    3. Voip.ms also provides for free in-network calling between all their account holders by dialing any other  account DID phone number, or SIP URI (via their Phone Book feature).  Or, directly dial one of your own sub-account extension numbers.
    4. If you populated your Voip.ms online Phone Book with phone numbers and SIP URI's for quick dialing, you will need to append the dialed sequence with the # key.  e.g.  **2*75xx#
    5. e911 service is available with Voip.ms- but must be activated with a paid e911subscription.




    **3
    Testing Calls via CallCentric
    Precede all outbound calls intended for routing through your CallCentric account with:  **3,  followed by the number you wish to call.


    All numbers listed below are Free Calls, via CallCentric and your OBi202 ATA connected phone:
    • CallCentric's Test number:   1-777-000-0001    e.g.,  dial as:    **317770000001
    • Call your CallCentric Voice Mail box (if you enabled it via your CallCentric dashboard):   *123#     e.g.,  dial as:     **3*123#
    • Free voice activated information service; sponsored by 24-7 Information Line:   1-777-123-4567     e.g.,  dial as:     **317771234567
    • Free Directory Assistance (advertiser sponsored):    411        e.g.,  dial as:   **3411
    • iNum network Echo Test:   011-883-5100-0000-0091#   e.g. dialed as:   **3011883510000000091#
    • SIP Broker peering network Test Announcement:    **275*-0111-88888#
      e.g.,  dialed as:   **3**275*011188888#

    Notes:
    1. 1800 numbers are not free calls via CallCentric pay-per-minute accounts. You will need calling credits if using pay-per-minute service like I do.
    2. Calling to PSTN landlines and mobiles requires CallCentric calling credits.
    3. CallCentric also provides for free in-network calling between all their account holders by dialing any other 1777xxxxxxx SIP account number, or sub-account extension.
    4. If you populated your CallCentric online Phone Book with phone numbers and SIP URI's for quick dialing, you may need to follow the *75xx phone book access code (and, assuming you used CallCentric's recommended Digit Map above) with the # send key.    i.e., dial :   **3*75xx #
    5. The CallCentric recommended Digit Map settings may not be optimal for some number sequence dialings.  If you experience excessive connect delays (~10 seconds), try following your dialed number with the pound key (#) to speed things up.
    6. e911 service is available with CallCentric - but must be activated with a paid DID incoming account. 


    **4


    All of the phone numbers I listed for test calling from the Google Voice account are Canadian phone numbers and are thus free phone calls via a Fongo or FreePhoneLine.ca account.  Use those same numbers for testing your Fongo-FreePhoneLine.ca calls.   Just precede each number with the FreePhoneLine.ca line code of **4.

    For example:
    • Check your Voice Mail:  *98     e.g.,  dial as:  **4*98
    • National Research Council, Canada - Talking Clock:    613-745-1576
      Dial as:   **46137451576


    I have now covered the bare-bones essentials of "manually" configuring your OBi202 ATA with Google Voice, Voip.ms, CallCentric, and FreePhoneLine.ca in one box and placing a few test calls from each service provider configured.

    Enjoy!