100% Free Outside and Inside Calling with LinuxMCE

[brent2009]

....Well at least until the end of this year. I have successfully set this up on LinuxMCE 8.10 final using Google Voice.  I have found out that Asterisk support for this is basically zero at this time so I am using Yate as a gateway to do this. Inbound and outbound calls are working %100 of the time with this configuration.   Note: I am sure yate can be successfully installed on linuxmce and configured. I had trouble with this even though it seems to install successfully. Right now it is running on a separate windows virtual server.

*I understand that I may not be the first to do this with asterisk but as far as I can tell this is the only up to date documentation out there that actually works and the only one that seems to work well with linuxmce.

FreePBX/asterisk configuration (linuxmce web admin/advanced/phone setup)

The next three screenshots show the configuration for the trunk.
Outbound caller ID I have set as my google voice number (with area code) in this format: xxxxxxxxxx



The IP address and port below are the yate gateway (windows server which by the way is on a virtual machine as is my linuxmce core).  (the main configuration that tells asterisk where to route outgoing calls)





Outbound Route
Showing route name (can be anything), dial patterns, and the selected trunk (created above).





Setup an inbound route. The only requirements here are to give a description and choose an extension. My extension is actually a Cisco 7970 phone!





Click General settings and scroll all the way to the bottom and change “Allow Anonymous Inbound SIP Calls” to yes and click Submit Changes. (Note: To me this is a very little security risk, linuxmce is behind a second firewall and because no ports are forwarded to linuxmce from the second firewall the only device that will ever be allowed to call is anything within my local network and in this case it will only be the yate gateway.)  You can of course set this up to just allow the IP of the yate gw but this is the easiest and quickest way to get things working.



Be sure to click "Apply Configuration Changes" when done.



Yate Configs

Yate configuration was followed exactly with a couple exceptions via the following link:
http://yate.null.ro/pmwiki/index.php?n=Main.ConnectingToGoogleVoice

The only tweaks to the tutorial above were related to the regexroute.conf file.
1)   You do not need to do anything with regfile.conf

2)    Do not add the following line:
${username}^$=-;error=noauth

3)   Do not add the following under contexts as show in the guide. Modify it as shown below and add it under [default] (change user to the asterisk ext (ex: 201) and the IP to your linuxmce IP address)

${in_line}GoogleVoice=sip/sip:user@linuxmceip:5060;called=user;jingle_version=0;jingle_flags=noping;dtmfmethod=rfc2833

Enjoy free calling! I have some additional tweaks and troubleshooting info I would like to add.  This was possible because of the great support from the guys at yate.  A big thanks to them, mainly Paul C for his help with some of the yate configuration.

[DragonK]

Hi brent,

Pls add this to the wiki so others can benefit of it too.

Karel

[l3mce]

Outstanding job Brent2009!

[tschak909]

However, with what you've done here, it will not pass correctly into LinuxMCE's custom dialing context, you need to do more work and not just duct tape it on.

-Thom.

[Marie.O]

Oh, and I think foxi is already working on Google Voice support in 1004 - look at http://svn.linuxmce.org/trac.cgi/ticket/1332 - maybe people can test stuff that's available.

[l3mce]

As a non-asterisk user, I didn't even read the post... just saw that it wasn't a feature request.

As possy said, there is a ticket open for this. I would be happy to help you do this the right way. I simply know nothing bout asterisk... however I do have a decent grasp of the system.

[brent2009]

As I understand Asterisk doesn't support a google voice configuration at all. Until Asterisk supports this there will be no "right" way to do this.  I don't understand why this wouldn't pass "correctly" into linuxmce's custom dialing context either?  LinuxMCE really is still handling everything as it normally would. I think of Yate as being a voip provider in this case. Am I wrong?

[l3mce]

From the ticket linked...
http://svn.linuxmce.org/trac.cgi/ticket/1332

As Asterisk 1.8.1.1 and higher has full support for direct use of Google's Talk services, thus enabling use of the Google Voice service as a free VoIP service, I'd like to see this built into the available Providers.

[brent2009]

This used to be asterisk's solution which has been broken for quite some time:

https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google

Note that Google's changes to their Google Voice system have rendered much of these capabilities inoperable, in many cases, or unreliable, in most cases.

[JaseP]

It's possible to integrate Google Voice calling with Asterisk using an Obi110. The Obi110 is similar to the SPA3102, developed by (much of) the same team. This is something I am going to be working on for my own system. What has to be done is to set up one of the Obi110's two configurable services for Google Voice, the other for Asterisk. I'm way out of my element with phone setups, so it's going to take me some time to do... If there's anyone familiar with asterisk who has a spare $45 (cost of the Obi110) and a little time, I'd recommend you beat me to the punch.

[brent2009]

It's possible to integrate Google Voice calling with Asterisk using an Obi110. The Obi110 is similar to the SPA3102, developed by (much of) the same team. This is something I am going to be working on for my own system. What has to be done is to set up one of the Obi110's two configurable services for Google Voice, the other for Asterisk. I'm way out of my element with phone setups, so it's going to take me some time to do... If there's anyone familiar with asterisk who has a spare $45 (cost of the Obi110) and a little time, I'd recommend you beat me to the punch.

Or you could just do it without buying any additional hardware as shown above. Instead of using Obi110 as a gateway you are using yate as the gateway.

[JaseP]

Yeah, but the Obi110 is an ATA, like the SPA3102, and provides a POTS to VOIP bridge. And it's cheaper than the average price you can get the SPA3102 for...

[tschak909]

Let's try to see if we can get this stuff plug and play.

-Thom

[JaseP]

I'd like that. Is there a device template for the SPA3102 that I could attempt to modify into an Obi110 template?

[JaseP]

Found the generic SIP softphone stuff in the Wiki and Web Admin... .still trying to wrap my noodle around SIP, Asterisk, and the Obi110 setup. I'm a complete newbie when it comes to telephony. Fake it 'til you make it, I guess... When I have something definitive, or need a rescue floatation device, I'll let you know.

PS: I don't think this will ever become truly plug and play, as VOIP setups are different from one provider to the next, as are ATAs, and desired setups will very from person to person. For instance, I want the Obi110 to handle most external connections, and Asterisk mainly for internal call routing, voice mail, and the system making calls out. That way I can configure Google Voice, externally, when I'm not home, based in where I or the family are going to be. Others will want LinuxMCE's Asterisk setup to handle mostly everything, but the physical VOIP to POTS bridge. However, I do think that mostly Plug-and-Play, with some walk-throughs might be a desired result.