I’m looking to add a Google Voice trunk that’s built into the new Asterisk 1.8. I use FreePBX in my system and was wondering if there’s any method to add it within the FreePBX framework. I really appreciate any help.
How to use Google Voice for free calls on an Asterisk 1.8+/FreePBX 2.8 system (the easy way)
That in part inspired me to come up with a vision for a Google Voice trunk module, which you can see here. Unfortunately I am not a coder, so if anyone wants that you’ll have to talk to the guys who write modules.
billsimon, thanks for trying to understand 
I’ve made some progress since my last post but still have some issues.
My main goal is to call GTalk users from my IP hardphones and call them using digits only. So, I already have 2 trunks like you mentioned and can route to them by prefixes.
Now I need to find a way to store gtalk addresses somewhere within the FreePBX and pass them to the custom trunk I created. I almost found a workaround with the phonebook but faced the fact that dots are not allowed even in the “name” field.
At the moment the only working option for me is creating an extension of a custom type for each gtalk contact, with the dial string like “gtalk/asterisk/[email protected]”, but I don’t like this approach.
Thanks for reading!
I’m having a hard time understanding what you are trying to do.
The only limitation I have encountered when dealing with GTalk in FreePBX is that FreePBX doesn’t permit alphabet characters in routing. So you can’t (for example) match on a word used as a dial string and then route to your custom GTalk trunk accordingly.
I configured a workaround but it’s not great. Just make a numeric dialing prefix for GTalk such that when you want to make a GTalk call, you start with the numeric prefix, e.g. 048.
In the outbound route, configure 048 as the prefix and . (dot) as the match string. The outbound route will strip off the 048 and send the string to the GTalk trunk.
On my softphone then I just enter 048username to dial [email protected], who is in my GTalk buddy list.
By the way the GTalk trunk has to be different from the Google Voice trunk because Google Voice forces the domain name @voice.google.com to be sent. So you have to set up two separate custom trunks that route over the same GTalk connection if you want to do both Talk and Voice connections.
That’s great tutorial!
But I’ still looking for a nice and easy way of calling other GTalk users from FreePBX. Anybody have a solution already?
It will be nice to use Asterisk Phonebook to store addresses but I have nothing against configuring individual “extensions” for such users. What I’m trying to avoid is adding those users as separate Dial strings within the config file.
Thanks!
Edit:
currently I have to use the extensions of a custom type for that purpose, don’t mess with Custom Extensions
I also made some custom scripting for a single gtalk user - i.e. checking it’s online status before dialing, sending a custom message before the call, etc. Now I want to pass the variable addresses to that section of the custom dialplan in a nice way.
Thanks for all the help. Between the great blog posts by billsimon at http://www.personal.psu.edu/wcs131/blogs/psuvoip/ and a little bit of researching, I think I also have everything work. The only problems I’m having right now is that when I make calls, I can hear the call, but I can’t speak. I’m using a snom 870 over IPv6. I know it doesn’t have NTP support in IPv6, but should that make a difference?
I’ve got IPv6 at work and at home (tunneled) and would be glad to try out the IPv6 transport in Asterisk 1.8. If you know someone who can make it happen on a RSCloud server, that’d be great.
That said, this thread being about Google Voice, the gtalk module (and probably Google Talk/Voice itself) doesn’t run over IPv6 yet. Google has been releasing some of their services over IPv6 to those who qualify: http://www.google.com/intl/en/ipv6/ Don’t think Talk/Voice are offered yet.
I actually work at one of the Google IPv6 partner institutions. We have a aaaa dns record for them and we got all the stuff they offer over v6. I connected the v6 person that we have and, unfortuently, you’re right, no Google Voice/Talk over v6. I’'ll still let you know how everything else goes. Thanks for all the help
I saw your blog and already subscribed to it. I also really enjoyed the Rackspace cloud piece, it’s a great idea. Great commercial opportunities too.
The project I’m working on has moved to an IPv6 only environment. I was looking at the Asterisk IPv6 port but it’s really old. Do you have any experience with Asterisk and IPv6? Do you know if the current SIP configurations allow for incoming IPv6 connections?
Take a look at the Asterisk 1.8 release announcement for the latest on IPv6 support.
Sorry about that. Missed the new IPv6 SIP channel feature. I’ll see what I can get up and running. If you’re interested in the v6 part, I also know some employees at Rackspace and can see what there v6 support is and if you can work that in to your guide at all.
Is anyone else using Asterisk with v6 out there? I know some SIP phones (snom on newer firmware, current Cisco 7900 line) support v6, but I’ve found that there’s often a big difference in the v6 world between saying it works and actually having it working. I’m running on a production v6 network which is looking to upgrade it’s communications system in the next few months. Any other unified communications systems (Cisco, NEC, etc.) that anyone has using v6?
Thanks for the mention. Indeed, I wrote up a how-to on my blog. See http://www.personal.psu.edu/wcs131/blogs/psuvoip/2010/11/adding_google_voice_to_freepbx.html
I would be glad for any corrections or improvements to this how-to.
Since no one else has answered and since I am too tired to write a long reply, I’ll just send you a few possibly pertinent links:
http://pbxinaflash.com/forum/showthread.php?t=8494
http://wiki.asterisk.org/wiki/display/AST/Calling+using+Google
Also keep an eye on this blog over the next few days/weeks, I think he’s about to do a writeup on the subject (and I don’t mean the stopgap that required running both Asterisk and FreeSWITCH on the same server): http://www.personal.psu.edu/wcs131/blogs/psuvoip/
Bottom line: Yes there is a method, but nobody’s written the definitive “cookbook” article quite yet, though my guess is it’s coming real soon now.