Marrying SIP to Google Voice

[isaacl]

A friend just showed me that if you make a call using the GV plugin in Gmail, then check the Google Talk logs (on Windows XP, the logs are in an archived file in C:\Documents and Settings\<user name>\Local Settings\Application Data\Google\Google Talk Plugin, something like gtalkplugin-c32521054.log.bz2), then the SIP info seems to be in there.
I'm not really sure what's what in there, but if anyone has any ideas, I'd love to be able to hook up a SIP device directly to a GV number, and be able to make outgoing calls without having to wire in a callback system...

 

[wardmundy]

On a Mac, you'll find the log at:

/Users/yourusername/Library/Application Support/Google/Google Talk Plugin

cp talkplugin-SIP1654612837-10-13-105-18.log.bz2 tmp.log.bz2
bunzip2 tmp.log.bz2
cat tmp.log

Good luck. It's uses standard Google cookie session crapola (i.e. encrypted login)...

Sending ["trstate","[email protected]/srvres-MTAuMTIuMGM2LkU6OTp1Mg==","[email protected]",{"allWritable":false,"anyWritable":false}] to BrowserConnection-0

Sending ["jc","[email protected]/srvres-MTAuMTIuMGM2LkU6OTp1Mg==","[email protected]",[["143.210.108.104","53560","rtp","feKyLOqLsr2Svfny","epdpdeNWwP3e/VTH","0.9","udp","0","stun","en1"]]] to BrowserConnection-0

Translation: The Russians or Chinese could probably figure it out (and will), but for the rest of us... the wait for SIP continues.

 

[Lost Trunk]

Well, I don't know what nationality the FreeSWITCH people are, but...

http://freeswitch.org/node/280

http://wiki.freeswitch.org/wiki/Google_Voice_Call

Edit: It appears their method in some way utilizes a module called mod_dingaling, which "is used to implement support for the jingle protocol (google talk)." I don't know how similar this would be to Asterisk's res_jabber module.

Maybe it's time for a version of PiaF that uses FreeSWITCH and either blue.box or FusionPBX?

Or not.

 

[wardmundy]

Tempting after reading this.

 

[mtennant]

Tempting after reading your link and after buying a Seagate Dockstar for $25 delivered and doing nothing with it yet.

 

[titodj]

Tempting after reading this.

Whatever was there, is now gone....

 

[Lost Trunk]

Whatever was there, is now gone....

EDIT: It's been renamed and moved here:

http://wiki.freeswitch.org/wiki/Google_Voice

Also it appears they've added information on "Receiving Calls From Google Voice" - and note that it does NOT appear to require a DID or Gizmo5 account!

I'm seriously jealous now!

 

[Lost Trunk]

I'm going to just throw this out there even though it may be a really bad idea for a number of reasons (SIP port conflicts come to mind) - but - I have to wonder if it would be possible to run both FreeSWITCH and Asterisk on the same server, using FreeSWITCH only to do the Google Voice stuff?

The idea would be that FreeSWITCH would only have a SIP trunk to the Asterisk server, and a Google Voice setup. On an outgoing call, you'd send it out the trunk to FreeSWITCH, and then FreeSWITCH would do the magic of bridging it to Google. On an incoming Google Voice call, FreeSWITCH would receive it and then pass it along to Asterisk using SIP. In essence, we'd be using FreeSWITCH as a protocol converter (I don't know if that would be the correct way to describe it in the strictest sense, but it sort of conveys the idea). I know this would make some additional demands on system resources, but I wonder if it's even possible? At very least, FreeSWITCH would have to be reconfigured to use some non-standard ports for SIP and possibly RTP. Doing something like that would not be anything I'd ever attempt, but I just thought I'd throw that idea out there and see if anyone thinks it's not totally insane.

 

[wardmundy]

The elves are busy working on this. Give us a little time.

:scooterb:

 

[mtennant]

I thought Asterisk had a Google Talk interface? Hopefully there isn't a need to install FreeSwitch and Asterisk.

 

[Lost Trunk]

I thought Asterisk had a Google Talk interface? Hopefully there isn't a need to install FreeSwitch and Asterisk.

It does, in theory. Check your /usr/lib/asterisk/modules directory and see if you have res_jabber.so and chan_gtalk.so in there.

One other potential issue is that many of the systems out today were built using an old version of iksemel that doesn't have the latest bugfixes - see http://code.google.com/p/iksemel/ and http://code.google.com/p/iksemel/downloads/list - I don't know what difference having an older version of this software might make, but it's a dependency for res_jabber.so, which in turn is a dependency for chan_gtalk.so. And even the latest downloadable version of iksemel is dated Jul 2009, but there have been many issues reported since (see http://code.google.com/p/iksemel/updates/list) however I do not know if any of those issues would affect the operation of chan_gtalk.so. In case there is a possibility they might, there is a way to do a "subversion checkout" of the latest source code (see http://code.google.com/p/iksemel/source/checkout) but at that point I'd be in way over my head, as I suspect most users would be.

Maybe best to wait for Ward and the elves to conjure up a solution!

See also: http://www.voip-info.org/wiki/view/Asterisk+Google+Talk (and http://leifmadsen.wordpress.com/2009/07/17/howto-getting-jabberxmpp-notifications-from-your-pbx/ if you want a way to test the Jabber module).

If there are any true hackers in the crowd (the good kind), an interesting exercise might be to compare FreeSWITCH's mod_dingaling module to Asterisk's res_jabber.so and/or chan_gtalk.so. I will just bet you that their mod_dingaling is somehow based on the earlier Asterisk modules, except they may have expunged some bugs. But I do not even come close to having the expertise to figure out what the differences might be.

 

[twinclouds]

Talking about the Dockstar, I went to the Debian route and successfully implemented GV support on Asterisk/FreePBX. It is actually quite straightforward once you get familiar with it. I have run it as my home server for over a week now and seems very stable. Actually, I have used a lot of things I learned from Orgasmatron/I-PBX, as well as the scripts developed by Ward.

 

[wardmundy]

Well, no cigar on the Gtalk/Jingle approach, but this is way above my pay grade. Luckily, there are sharper tools in the shed so we're going to throw it out there with a $100 bounty to the first whiz kid that can document how to get phone calls through Google Voice using Jabber/Jingle working on our platform.

Within an hour or so, you'll be able to download the script to install all of the components on your 1.7.5.5 Gold, Silver, or Bronze build. Be aware that EVERYTHING pertaining to Jingle is different with Asterisk 1.4 and 1.6 so you'll probably want to experiment with Asterisk 1.6. The script to install everything is install-iksemel. Config files are in /etc/asterisk.


Asterisk 1.4 Setup:

With Asterisk 1.4, you'll only have these config files to deal with.

In modules.conf, add the following:

load = res_jabber.so
load = chan_gtalk.so

In jabber.conf, make it look like this with your Gmail credentials:

[general]
debug=no
autoprune=no
autoregister=yes

[asterisk]
type=client
serverhost=talk.google.com
[email protected]/Talk
secret=yourpassword
priority=1
port=5222
usetls=yes
usesasl=yes
[email protected]
timeout=100 
status=available
statusmessage="PIAF Online"

In gtalk.conf, make it look like this:

[general]
context=from-gtalk
allowguest=yes
                                               
[guest]
disallow=all
allow=ulaw
context=from-gtalk

[yourbuddyname]
[email protected]
disallow=all
allow=ulaw
context=from-gtalk
connection=asterisk

At the bottom of extensions_custom.conf, add the following:

[google-out]
exten => 299,1,Dial(gtalk/asterisk/[email protected])

[from-gtalk]
exten => s,1,Set(__FROM_DID=${EXTEN})
exten => s,n,Gosub(app-blacklist-check,s,1)
exten => s,n,ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)}))
exten => s,n,Ringing()
exten => s,n,Set(__CALLINGPRES_SV=${CALLERPRES()})
exten => s,n,Set(CALLERPRES()=allowed_not_screened)
exten => s,n,Set(_RGPREFIX=GTALK: )
exten => s,n,Set(CALLERID(name)=${RGPREFIX}${CALLERID(name)})
exten => s,n(dest-ext),Goto(app-daynight,1,1)
exten => s,n,Hangup

If you'd like to test things out, add something like this to the top of extensions_custom.conf just below [from-internal-custom]:

exten => 333,1,Answer
exten => 333,n,Wait(2)
exten => 333,n,jabberstatus(asterisk,[email protected],STATUS)
exten => 333,n,NoOp(${STATUS})
exten => 333,n,JABBERSend(asterisk,[email protected],Just testing Jabber)
exten => 333,n,Hangup

Once you get the pieces in place: amportal restart


From the Asterisk CLI:

jabber show connected
jabber test

Be sure to open UDP 5222 on your IPtables firewall and also map the port from your hardware firewall to your server:

-A INPUT -p udp -m udp --dport 5222 -j ACCEPT

Then: service iptables restart



Asterisk 1.6 tips coming shortly!

 

[rossiv]

This could just be my system, but:

root@piaf-dev1:/etc/asterisk $ asterisk -vr
Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.4.21.2 currently running on piaf-dev1 (pid = 28705)
Verbosity is at least 3
piaf-dev1*CLI> jabber show connected
No such command 'jabber show connected' (type 'help jabber show' for other possible commands)
piaf-dev1*CLI> jabber test
No such command 'jabber test' (type 'help jabber test' for other possible commands)
piaf-dev1*CLI> exit
Executing last minute cleanups

[general]
debug=no                                ;;Turn on debugging by default.
autoprune=no                            ;;Auto remove users from buddy list.
autoregister=yes                        ;;Auto register users from buddy list.

[asterisk]                              ;;label
type=client                             ;;Client or Component connection
serverhost=talk.gogle.com               ;;Route to server for example,
                                        ;;    talk.google.com
[email protected]/Talk   ;;Username with optional roster.
secret=MyP@$$\/\/()r|)       ;;Password
port=5222
priority=1                      ;;Port to use defaults to 5222
usetls=yes                              ;;Use tls or not
usesasl=yes                             ;;Use sasl or not
[email protected]          ;;Manual addition of buddy to list.
statusmessage="PIAF-DEV1 is Online"             ;;Have custom status message for
status=available                                        ;;Asterisk.
timeout=100                             ;;Timeout on the message stack.

[general]
context=from-gtalk              ;;Context to dump call into
allowguest=yes                  ;;Allow calls from people not in
                                ;;list of peers
;
[guest]                 ;;special account for options on guest account
disallow=all
allow=ulaw
context=from-gtalk
;
[myothergmail]
[email protected]    ;;username of the peer your
                                ;;calling or accepting calls from
disallow=all
allow=ulaw
context=from-gtalk
connection=asterisk             ;;client or component in jabber.conf
                                ;;for the call to leave on.
;

and in modules.conf I added

load => res_jabber.so
load => chan_gtalk.so
(Tried it both ways...)
load = res_jabber.so
load = chan_gtalk.so

What did I do wrong?

EDIT:
Well, it must be too late for my to be messing with phones....

It does, in theory. Check your /usr/lib/asterisk/modules directory and see if you have res_jabber.so and chan_gtalk.so in there.

And I don't have either of those and I am running 1.7.5.5 gold on my dev box.

How would I go about getting those modules?

----
Well, I found this:
http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk
and got as far as

[B]Compiling Asterisk [/B]

[FONT=monospace] cd /usr/src/asterisk [/FONT]
[FONT=monospace] ./configure [/FONT]
[FONT=monospace] make clean [/FONT]
[FONT=monospace] make install [/FONT]
[FONT=monospace] make samples 
[/FONT]

and then Asterisk quit working.. :-(
Should I just start everything over like the whole system? That's why I did this on a dev box first!
BUT...I do have chan_gtalk and res_jabber.so now....

Edit 2:
Well, do I feel stupid!

The script to install everything is install-iksemel.

That didn't work at first - until I ran update-scripts and update-fixes..........and it (waiting....to see if it works....) fix my asterisk problem....

Oh kill me......
EDIT 3
Now that I ran the new script...

root@piaf-dev1:/etc/asterisk $ amportal start


SETTING FILE PERMISSIONS
Permissions OK

STARTING ASTERISK
Asterisk ended with exit status 1
Asterisk died with code 1.
cat: /var/run/asterisk.pid: No such file or directory
Automatically restarting Asterisk.
mpg123: no process killed
Asterisk ended with exit status 1
Asterisk died with code 1.
cat: /var/run/asterisk.pid: No such file or directory
Automatically restarting Asterisk.
mpg123: no process killed

-----------------------------------------------------
Asterisk could not start!
Use 'tail /var/log/asterisk/full' to find out why.
-----------------------------------------------------
root@piaf-dev1:/etc/asterisk $ tail /var/log/asterisk/full
[2010-09-09 22:14:45] NOTICE[29327] chan_iax2.c: Peer 'iax-fax0' is not dynamic (from 127.0.0.1)
[2010-09-09 22:14:45] NOTICE[29330] chan_iax2.c: Peer 'iax-fax1' is not dynamic (from 127.0.0.1)
[2010-09-09 22:14:52] VERBOSE[29002] logger.c:   == Parsing '/etc/asterisk/manager.conf': [2010-09-09 22:14:52] VERBOSE[29002] logger.c: Found
[2010-09-09 22:14:52] VERBOSE[29002] logger.c:   == Parsing '/etc/asterisk/users.conf': [2010-09-09 22:14:52] VERBOSE[29002] logger.c: Found
[2010-09-09 22:14:52] NOTICE[29002] manager.c: 127.0.0.1 tried to authenticate with nonexistent user 'admin'
[2010-09-09 22:14:53] VERBOSE[29002] logger.c:   == Connect attempt from '127.0.0.1' unable to authenticate
[2010-09-09 22:14:53] VERBOSE[29339] logger.c: Beginning asterisk shutdown....
[2010-09-09 22:14:53] VERBOSE[29339] logger.c: Executing last minute cleanups
[2010-09-09 22:14:53] VERBOSE[29339] logger.c:   == Destroying musiconhold processes
[2010-09-09 22:14:53] VERBOSE[29339] logger.c: Asterisk cleanly ending (0).

Going to reinstall.....

 

[garyfett]

You just said the SIP info is inside the BZ2 file. Maybe its in a readable text format. Care to send it here?

 

[wardmundy]

Did you forget to run install-iksemel

 

[rossiv]

Not exactly, Ward.
Install-iksemel wouldn't run the first time I tried it of course - I forgot to run update-scripts and update-fixes and then it came up...after I ran the other commands that I posted in my post.
So I am now downloading 1.7.55.2 and (re)installing it onto my dev box aka old laptop
Thanks

 

[rossiv]

Yay! Working, sort of....
I can call and Asterisk recognizes the call but it never answers..
I don't know what I should sanitize here except my emails so here is the log:

pbx*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk1AF7BABF" type="set" id="687CA7A1BA78C840" from="[email protected]/gmail.F9D54BB1"><ses:session type="initiate" initiator="[email protected]/gmail.F9D54BB1" id="c577598885" xmlns:ses="http://www.google.com/session"><pho:description xmlns:pho="http://www.google.com/session/phone"><pho:payload-type id="103" name="ISAC" clockrate="16000"/><pho:payload-type id="119" name="ISACLC" bitrate="40000" clockrate="16000"/><pho:payload-type id="99" name="speex" bitrate="22000" clockrate="16000"/><pho:payload-type id="97" name="IPCMWB" bitrate="80000" clockrate="16000"/><pho:payload-type id="9" name="G722" bitrate="64000" clockrate="16000"/><pho:payload-type id="102" name="iLBC" bitrate="13300" clockrate="8000"/><pho:payload-type id="98" name="speex" bitrate="11000" clockrate="8000"/><pho:payload-type id="3" name="GSM" bitrate="13000" clockrate="8000"/><pho:payload-type id="100" name="EG711U" bitrate="64000" clockrate="8000"/><pho:payload-type id="101" name="

JABBER: gtalk_account INCOMING: EG711A" bitrate="64000" clockrate="8000"/><pho:payload-type id="0" name="PCMU" bitrate="64000" clockrate="8000"/><pho:payload-type id="8" name="PCMA" bitrate="64000" clockrate="8000"/><pho:payload-type id="117" name="red" clockrate="8000"/><pho:payload-type id="106" name="telephone-event" clockrate="8000"/></pho:description></ses:session></iq>
pbx*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk1AF7BABF" type="set" id="7C9F0AF4D898FE99" from="[email protected]/gmail.F9D54BB1"><ses:session type="candidates" initiator="[email protected]/gmail.F9D54BB1" id="c577598885" xmlns:ses="http://www.google.com/session"><ses:candidate address="192.168.X.XXX" port="57283" name="rtp" username="I347OxlgZgl5ARfi" password="EQAjWcdMZYaEoDKe" preference="1" protocol="udp" generation="0" network="en1" type="local"/></ses:session></iq>
pbx*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk1AF7BABF" type="set" id="70518C1292FD6C83" from="[email protected]/gmail.F9D54BB1"><ses:session type="candidates" initiator="[email protected]/gmail.F9D54BB1" id="c577598885" xmlns:ses="http://www.google.com/session"><ses:candidate address="5.191.XXX.XXX" port="51618" name="rtp" username="cTY9mSGudNyUsW2v" password="p8zsXBy+vPLwgmG6" preference="1" protocol="udp" generation="0" network="tap0" type="local"/></ses:session></iq>
pbx*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk1AF7BABF" type="set" id="1D9566F7419ED22A" from="[email protected]/gmail.F9D54BB1"><ses:session type="candidates" initiator="[email protected]/gmail.F9D54BB1" id="c577598885" xmlns:ses="http://www.google.com/session"><ses:candidate address="24.31.XXX.XXX" port="64171" name="rtp" username="bqXtGdlaDfyzyhBG" password="9y3vhwFupCH89Jsg" preference="0.9" protocol="udp" generation="0" network="en1" type="stun"/></ses:session></iq>
pbx*CLI> 
JABBER: gtalk_account INCOMING:  
pbx*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk1AF7BABF" type="set" id="BFD0756FB197A58C" from="[email protected]/gmail.F9D54BB1"><ses:session type="candidates" initiator="[email protected]/gmail.F9D54BB1" id="c577598885" xmlns:ses="http://www.google.com/session"><ses:candidate address="192.168.X.XXX" port="61997" name="rtp" username="xzd+puBZMvDE1xB5" password="MP9/I1EDyUOqrbmm" preference="1" protocol="udp" generation="1" network="en1" type="local"/></ses:session></iq>
pbx*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk1AF7BABF" type="set" id="C51ED16DE9CCDF8E" from="[email protected]/gmail.F9D54BB1"><ses:session type="candidates" initiator="[email protected]/gmail.F9D54BB1" id="c577598885" xmlns:ses="http://www.google.com/session"><ses:candidate address="5.191.XXX.XXX" port="50283" name="rtp" username="WeQGwEDC8XfxsX1u" password="5i7mvnypseHBl23C" preference="1" protocol="udp" generation="1" network="tap0" type="local"/></ses:session></iq>
pbx*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk1AF7BABF" type="set" id="8A79C1A4CA077CF5" from="[email protected]/gmail.F9D54BB1"><ses:session type="candidates" initiator="[email protected]/gmail.F9D54BB1" id="c577598885" xmlns:ses="http://www.google.com/session"><ses:candidate address="24.31.XXX.XXX" port="57974" name="rtp" username="wiKHdoVUw/bbQGVF" password="OY+tqD8zZ8DMFQ3c" preference="0.9" protocol="udp" generation="1" network="en1" type="stun"/></ses:session></iq>
pbx*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk1AF7BABF" type="set" id="EF99A1C932E706F9" from="[email protected]/gmail.F9D54BB1"><ses:session type="terminate" initiator="[email protected]/gmail.F9D54BB1" id="c577598885" xmlns:ses="http://www.google.com/session"><ses:reason><ses:connectivity-error/></ses:reason><pho:call-ended xmlns:pho="http://www.google.com/session/phone"/></ses:session></iq>
pbx*CLI>

EDIT 1:
Cool! If you call your GV number and have GTalk enabled as a forwarding phone, it calls your PBX! (Clarification: Run install-iksemel on 1.7.55 first and configure it) However, the pbx is not answering as seen in the contents of this post...
Does anyone know how to get the system to answer the call?

piaf-dev1*CLI> 
JABBER: gtalk_account INCOMING: <iq from="[email protected]/srvres-MTAuMTc2LjE0MS44Ojk4MjY=" to="[email protected]/Talk8193E285" id="jingle:10.176.141.8-29462690:1:F2030DB5" type="set"><ses:session type="initiate" id="[email protected]" initiator="[email protected]/srvres-MTAuMTc2LjE0MS44Ojk4MjY=" xmlns:ses="http://www.google.com/session"><pho:description xmlns:pho="http://www.google.com/session/phone"><pho:payload-type id="0" name="PCMU" clockrate="8000"/><pho:payload-type id="101" name="telephone-event"/></pho:description><transport behind-symmetric-nat="false" can-receive-from-symmetric-nat="false" xmlns="http://www.google.com/transport/raw-udp"/><transport xmlns="http://www.google.com/transport/p2p"/></ses:session></iq>
piaf-dev1*CLI> 
JABBER: gtalk_account INCOMING: <iq from="[email protected]/srvres-MTAuMTc2LjE0MS44Ojk4MjY=" to="[email protected]/Talk8193E285" id="jingle:10.176.141.8-29462690:1:F2030DDF" type="set"><ses:session type="terminate" id="[email protected]" initiator="[email protected]/srvres-MTAuMTc2LjE0MS44Ojk4MjY=" xmlns:ses="http://www.google.com/session"><pho:call-ended xmlns:pho="http://www.google.com/session/phone">Call cancelled</pho:call-ended></ses:session></iq>
piaf-dev1*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk8193E285" id="F42AAB2A" type="set"><query xmlns="jabber:iq:roster"/></iq>
piaf-dev1*CLI>

 

[tm1000]

As a starting point try turning on "Allow anonymous sip calls"

But remember to turn it off when done testing.

Yay! Working, sort of....
I can call and Asterisk recognizes the call but it never answers..
I don't know what I should sanitize here except my emails so here is the log:

pbx*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk1AF7BABF" type="set" id="687CA7A1BA78C840" from="[email protected]/gmail.F9D54BB1"><ses:session type="initiate" initiator="[email protected]/gmail.F9D54BB1" id="c577598885" xmlns:ses="http://www.google.com/session"><pho:description xmlns:pho="http://www.google.com/session/phone"><pho:payload-type id="103" name="ISAC" clockrate="16000"/><pho:payload-type id="119" name="ISACLC" bitrate="40000" clockrate="16000"/><pho:payload-type id="99" name="speex" bitrate="22000" clockrate="16000"/><pho:payload-type id="97" name="IPCMWB" bitrate="80000" clockrate="16000"/><pho:payload-type id="9" name="G722" bitrate="64000" clockrate="16000"/><pho:payload-type id="102" name="iLBC" bitrate="13300" clockrate="8000"/><pho:payload-type id="98" name="speex" bitrate="11000" clockrate="8000"/><pho:payload-type id="3" name="GSM" bitrate="13000" clockrate="8000"/><pho:payload-type id="100" name="EG711U" bitrate="64000" clockrate="8000"/><pho:payload-type id="101" name="

JABBER: gtalk_account INCOMING: EG711A" bitrate="64000" clockrate="8000"/><pho:payload-type id="0" name="PCMU" bitrate="64000" clockrate="8000"/><pho:payload-type id="8" name="PCMA" bitrate="64000" clockrate="8000"/><pho:payload-type id="117" name="red" clockrate="8000"/><pho:payload-type id="106" name="telephone-event" clockrate="8000"/></pho:description></ses:session></iq>
pbx*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk1AF7BABF" type="set" id="7C9F0AF4D898FE99" from="[email protected]/gmail.F9D54BB1"><ses:session type="candidates" initiator="[email protected]/gmail.F9D54BB1" id="c577598885" xmlns:ses="http://www.google.com/session"><ses:candidate address="192.168.X.XXX" port="57283" name="rtp" username="I347OxlgZgl5ARfi" password="EQAjWcdMZYaEoDKe" preference="1" protocol="udp" generation="0" network="en1" type="local"/></ses:session></iq>
pbx*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk1AF7BABF" type="set" id="70518C1292FD6C83" from="[email protected]/gmail.F9D54BB1"><ses:session type="candidates" initiator="[email protected]/gmail.F9D54BB1" id="c577598885" xmlns:ses="http://www.google.com/session"><ses:candidate address="5.191.XXX.XXX" port="51618" name="rtp" username="cTY9mSGudNyUsW2v" password="p8zsXBy+vPLwgmG6" preference="1" protocol="udp" generation="0" network="tap0" type="local"/></ses:session></iq>
pbx*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk1AF7BABF" type="set" id="1D9566F7419ED22A" from="[email protected]/gmail.F9D54BB1"><ses:session type="candidates" initiator="[email protected]/gmail.F9D54BB1" id="c577598885" xmlns:ses="http://www.google.com/session"><ses:candidate address="24.31.XXX.XXX" port="64171" name="rtp" username="bqXtGdlaDfyzyhBG" password="9y3vhwFupCH89Jsg" preference="0.9" protocol="udp" generation="0" network="en1" type="stun"/></ses:session></iq>
pbx*CLI> 
JABBER: gtalk_account INCOMING:  
pbx*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk1AF7BABF" type="set" id="BFD0756FB197A58C" from="[email protected]/gmail.F9D54BB1"><ses:session type="candidates" initiator="[email protected]/gmail.F9D54BB1" id="c577598885" xmlns:ses="http://www.google.com/session"><ses:candidate address="192.168.X.XXX" port="61997" name="rtp" username="xzd+puBZMvDE1xB5" password="MP9/I1EDyUOqrbmm" preference="1" protocol="udp" generation="1" network="en1" type="local"/></ses:session></iq>
pbx*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk1AF7BABF" type="set" id="C51ED16DE9CCDF8E" from="[email protected]/gmail.F9D54BB1"><ses:session type="candidates" initiator="[email protected]/gmail.F9D54BB1" id="c577598885" xmlns:ses="http://www.google.com/session"><ses:candidate address="5.191.XXX.XXX" port="50283" name="rtp" username="WeQGwEDC8XfxsX1u" password="5i7mvnypseHBl23C" preference="1" protocol="udp" generation="1" network="tap0" type="local"/></ses:session></iq>
pbx*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk1AF7BABF" type="set" id="8A79C1A4CA077CF5" from="[email protected]/gmail.F9D54BB1"><ses:session type="candidates" initiator="[email protected]/gmail.F9D54BB1" id="c577598885" xmlns:ses="http://www.google.com/session"><ses:candidate address="24.31.XXX.XXX" port="57974" name="rtp" username="wiKHdoVUw/bbQGVF" password="OY+tqD8zZ8DMFQ3c" preference="0.9" protocol="udp" generation="1" network="en1" type="stun"/></ses:session></iq>
pbx*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk1AF7BABF" type="set" id="EF99A1C932E706F9" from="[email protected]/gmail.F9D54BB1"><ses:session type="terminate" initiator="[email protected]/gmail.F9D54BB1" id="c577598885" xmlns:ses="http://www.google.com/session"><ses:reason><ses:connectivity-error/></ses:reason><pho:call-ended xmlns:pho="http://www.google.com/session/phone"/></ses:session></iq>
pbx*CLI>

EDIT 1:
Cool! If you call your GV number and have GTalk enabled as a forwarding phone, it calls your PBX! (Clarification: Run install-iksemel on 1.7.55 first and configure it) However, the pbx is not answering as seen in the contents of this post...
Does anyone know how to get the system to answer the call?

piaf-dev1*CLI> 
JABBER: gtalk_account INCOMING: <iq from="[email protected]/srvres-MTAuMTc2LjE0MS44Ojk4MjY=" to="[email protected]/Talk8193E285" id="jingle:10.176.141.8-29462690:1:F2030DB5" type="set"><ses:session type="initiate" id="[email protected]" initiator="[email protected]/srvres-MTAuMTc2LjE0MS44Ojk4MjY=" xmlns:ses="http://www.google.com/session"><pho:description xmlns:pho="http://www.google.com/session/phone"><pho:payload-type id="0" name="PCMU" clockrate="8000"/><pho:payload-type id="101" name="telephone-event"/></pho:description><transport behind-symmetric-nat="false" can-receive-from-symmetric-nat="false" xmlns="http://www.google.com/transport/raw-udp"/><transport xmlns="http://www.google.com/transport/p2p"/></ses:session></iq>
piaf-dev1*CLI> 
JABBER: gtalk_account INCOMING: <iq from="[email protected]/srvres-MTAuMTc2LjE0MS44Ojk4MjY=" to="[email protected]/Talk8193E285" id="jingle:10.176.141.8-29462690:1:F2030DDF" type="set"><ses:session type="terminate" id="[email protected]" initiator="[email protected]/srvres-MTAuMTc2LjE0MS44Ojk4MjY=" xmlns:ses="http://www.google.com/session"><pho:call-ended xmlns:pho="http://www.google.com/session/phone">Call cancelled</pho:call-ended></ses:session></iq>
piaf-dev1*CLI> 
JABBER: gtalk_account INCOMING: <iq to="[email protected]/Talk8193E285" id="F42AAB2A" type="set"><query xmlns="jabber:iq:roster"/></iq>
piaf-dev1*CLI>

 

[rossiv]

No dice on enabling anon sip calls....
Anything else?