Gtalk incoming call won't go to dialplan context

Asterisk Asterisk Support
1

I am having a strange issue with my Gtalk installation.
Was working for long, and is now broken for some yet unknown reason.

Asterisk 1.8.4.3
Skype for Asterisk also installed (had been working with SFA installed).

Issue is the following:

  • Gtalk user registered.
  • Buddy does audio call to Gtalk asterisk user.
  • Rings but asterisk does not pickup.

I have turned “jabber set debug on”, which shows me that asterisk is receiving the call request:

[quote]JABBER: gtalk_account INCOMING: <jin:jingle action=“session-initiate” sid=“c1639980010” initiator=""xxxxxx@gmail.com/gmail.C438B86A" xmlns:jin=“urn:xmpp:jingle:1”><jin:content name=“audio” creator=“initiator”><rtp:description media=“audio” xmlns:rtp=“urn:xmpp:jingle:apps:rtp:1”><rtp:payload-type id=“103” name=“ISAC” clockrate=“16000”><rtp:parameter name=“bitrate” value=“32000”/></rtp:payload-type><rtp:payload-type id=“104” name=“ISAC” clockrate=“32000”><rtp:parameter name=“bitrate” value=“56000”/></rtp:payload-type><rtp:payload-type id=“119” name=“ISACLC” clockrate=“16000”><rtp:parameter name=“bitrate” value=“40000”/></rtp:payload-type><rtp:payload-type id=“99” name=“speex” clockrate=“16000”><rtp:parameter name=“bitrate” value=“22000”/></rtp:payload-type><rtp:payload-type id=“97” name=“IPCMWB” clockrate=“16000”><rtp:parameter name=“bitrate” value=“80000”/></rtp:
[/quote]

and other packets …
Having set “core set verbose 6”,

I see that asterisk never goes to the context 1st line…

Her is my config:

jabber.conf

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

[gtalk_account]
type=client
serverhost=talk.google.com
username=xxxxxx@gmail.com
secret=password
port=5222
usetls=yes
usesasl=yes
statusmessage="asterisk server"
timeout=100

gtalk.conf

[general]
context=incoming-google        
bindaddr=0.0.0.0
allowguest=yes                

[guest]                
allow=all
context=incoming-google
connection=gtalk_account

extensions.conf

[incoming-google]
exten => s,1,wait(5)
exten => s,2,Answer()
exten => s,3,SendText(Welcome to the GTalk server.)

Anyone knows how to debug why call does not go from Gtalk to the dialplan?
in advance, thanks,

Morchea

2

See:

issues.asterisk.org/jira/browse/ASTERISK-17993

3

Thanks for that Malcolm.

Do you know if this is getting some kind of attention from asterisk developers? Seems not too much from the issue tracker. Cause this impacts my business and my consider dropping GTalk support today if issue wont be solved in days.

Kind regards,
Morchea

4

Howdy,

Given that Google’s providing a free service and that they can (have) change (already changed) the protocol at will (whim), I probably wouldn’t base my business on it.

I haven’t seen any patch movement from the non-Digium development community.

Cheers.