Marrying SIP to Google Voice

[Lost Trunk]

This might explain why it's not answering

See this message:

http://archives.free.net.ph/message/20101005.155907.49a50b6a.en.html

Two points from that message:

Apparently part of the problem (at least for Digium's David Vossel, the person who posted that message) is that when Google Voice calls via Gtalk, it prompts you with, "Press 1 to accept the call, press 2 to send to voicemail." But there is apparently no way to send touch tones back through the Gtalk channel, at least not that he's discovered yet.

And also, Mr. Vossel writes: "I'm currently using the svn/asterisk/team/dvossel/gtalk_fixup branch if anyone is interested in messing with this."

So it appears someone at Digium is working on this, but without much success so far.

Which begs the question of how on earth the FreeSWITCH people managed to get it working...

 

[Lost Trunk]

Just one more note, for those of you who are whole lot smarter than I am (I hate iptables, it almost appears that it was designed so that mere mortal couldn't understand it, but maybe that's just me). Anyway it appears that if you do know how to use iptables, there is a way to redirect sip packets so you can see what they are doing. This thread was actually written in a forum that discussed the Tomato firmware, which can be installed as replacement firmware on many popular routers, but the tip there might be useful in other circumstances:

http://www.linksysinfo.org/forums/showthread.php?p=368398#post368398

Here's the current contents of the thread (at the time I'm posting this), in case it gets moved or taken down for some reason:

premudriy wrote:

Hello everyone,


I'm writing a program that deals with SIP packets. I have a MagicJack phone running on my old laptop, but I am developing on another computer.

I need somehow to redirect/retransmit packets which go to the laptop so that they would also go to my computer on which I am developing.

I'm guessing that I can do it with a firewall rule, but my knowledge with iptables is limited. Is this even possible?

If so, then what firewall rule will take all the packets and retransmit/duplicate/forward them to an additional IP on the network?
​

rhester72 replied:

modprobe ipt_ROUTE
iptables -A PREROUTING -t mangle -p <protocol> --dport <port> -j ROUTE --gw <target> --tee

where <protocol> is either udp or tcp (I'm guessing udp in your case), <port> is the port number (I seem to recall SIP uses 10000?), and <target> is the IP address you want to duplicate the packets to.

You can safely put the above in your Firewall script.

Rodney
​

My thought is that it may be possible to examine the packets going to and from Google Voice to crack open its secrets and get it working with Asterisk, although as I said, this would need to be done by someone a WHOLE lot smarter than I.

 

[malcolmd]

Howdy,

Do an update from the above-mentioned branch

http://svn.asterisk.org/svn/asterisk/team/dvossel/gtalk_fixup/

today and your results should be different.

Toss something like:

exten => s,1,Answer()
exten => s,n,Wait(2)
exten => s,n,SendDTMF(1)

in front of your Dial, and you won't have to hear the "press 1 to accept" message.

Cheers

 

[Lost Trunk]

A blog post you may find interesting

This quote is from an article entitled "Asterisk 1.8.0 RC3 and Google Voice" in the PSU VoIP blog:

New features in Asterisk 1.8 may drag me out of the comfortable stability that is version 1.4.

Release candidate 3 of version 1.8.0 was announced last week, and it includes an updated chan_gtalk module that enables communications with the Gmail web client and its protocol/codecs. It's neither standard SIP nor XMPP-Jingle, which I hope is yet to come, but the updated module is the last link necessary to allow directly placing and receiving calls through Google Voice. From the Google Voice web interface, you can specify Gmail Chat as one of your phones to ring on an incoming call. And this comment from the Asterisk code repository shows how to make outbound Google Voice calls in one line from the dialplan.
​

Just trying to help contribute to the knowledge base on this… but so far it's looking like the best bet for making outgoing Google Voice calls is to use either FreeSWITCH or Asterisk 1.8 (and even then we don't know how well it will work, and of course FreeSWITCH isn't supported at all by PiaF). Oh, and here's two other articles from that same blog (the last one added well after this post was originally written) that may or may not be relevant:
More Asterisk and XMPP (Jabber) integration
[URL="http://www.personal.psu.edu/wcs131/blogs/psuvoip/2010/10/using_freeswitch_to_add_google.html"]Using FreeSWITCH to add Google Voice to Asterisk
[/URL]

 

[wardmundy]

Latest PIAF Purple payload includes the option for 1.8 RC3.

 

[malcolmd]

The PSU blogger is on the right track, but isn't wholly accurate.

1.8.0-rc3 is nearly there, Google Talk calling should be fine, but Google Voice isn't addressed in that release. It *is* addressed in the gtalk_fixup team branch; and, after some cleanup, will find its way into a future release candidate of 1.8.

We've discovered a few tips along the way, that we'll be sharing at Astricon. David Vossel will be there speaking, not specifically about Google and Asterisk integration, but either he or I will be happy to answer questions.

Cheers.

 

[malcolmd]

As a side note, it's really great to see the PIAF community excited about using Asterisk with Google Talk/Voice.

 

[wardmundy]

And it's great to see you excited about PIAF.

In other 1.8 news, looks like app-swift has been patched to work with Asterisk 1.8 which means all the Nerd Vittles TTS apps should be working shortly.

 

[MichiganTelephone]

Just to let you know, you might not have all the time in the world to find some other way to make calls via Google Voice if you are using a Gizmo5 account to receive the callbacks, because some users are finding this no longer works for them. See my blog post at http://michigantelephone.wordpress.com/2010/10/15/is-time-running-out-for-gizmo5/ for additional information.

I hope this is just a programming error at Google and that they will fix it, but that may not be the case. It may be that Gizmo5 is slowly headed toward the graveyard of formerly-loved but forgotten services. :angelb:

EDIT: It's looking like this is an issue that only affects those who have Gizmo5 numbers in the 747-0XX-XXXX or 747-1XX-XXXX range (in other words, they don't fit the normal PSTN pattern, but are perfectly valid Gizmo5 numbers). Not totally confirmed yet, but for now it's looking like that's behind this problem - see http://www.google.com/support/forum/p/voice/thread?tid=352a5dfbba6a9388&hl=en for additional discussion.

 

[Lost Trunk]

Another article for your reading pleasure:

Asterisk 1.8 and Google Voice

Be aware that this article seems to assume installation on a Ubuntu or Debian system rather than CentOS (CentOS uses yum rather than apt-get during installations).