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Summary:ASTERISK-13786: No audio from Gtalk to Asterisk
Reporter:tootai (tootai)Labels:
Date Opened:2009-03-20 05:28:10Date Closed:2009-06-16 07:40:39
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_gtalk
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) jabber-debug.txt
Description:When calling from Gtalk client to an Asterisk user we have no audio in both direction. We have calling signal in GTalk client until called party answers, on callee side phone is ringing normally and that's all. In logs we have:

[Mar 20 10:20:08] VERBOSE[20006] logger.c:     -- Local/821@ServiceNumbers-754b,1 answered Gtalk/ivan.trudnai-2df4
[Mar 20 10:20:08] NOTICE[20006] chan_gtalk.c: Don't know how to indicate condition '-1'
[Mar 20 10:20:08] NOTICE[20006] chan_gtalk.c: Don't know how to indicate condition '20'
[Mar 20 10:20:08] NOTICE[20006] chan_gtalk.c: Don't know how to indicate condition '20'
[Mar 20 10:20:08] NOTICE[20006] chan_gtalk.c: Don't know how to indicate condition '-1'
[Mar 20 10:20:08] NOTICE[20006] chan_gtalk.c: Don't know how to indicate condition '20'
[Mar 20 10:20:08] NOTICE[20006] chan_gtalk.c: Don't know how to indicate condition '20'
[Mar 20 10:20:49] NOTICE[20006] chan_gtalk.c: Don't know how to indicate condition '20'
[Mar 20 10:20:49] VERBOSE[20007] logger.c:     -- Executing [h@ServiceNumbers:1] GotoIf("Local/821@ServiceNumbers-754b,2", "0?end") in new stack
[Mar 20 10:20:49] VERBOSE[20007] logger.c:     -- Executing [h@ServiceNumbers:2] Hangup("Local/821@ServiceNumbers-754b,2", "") in new stack
[Mar 20 10:20:49] VERBOSE[20007] logger.c:   == Spawn h extension (ServiceNumbers, h, 2) exited non-zero on 'Local/821@ServiceNumbers-754b,2'
[Mar 20 10:20:49] VERBOSE[20007] logger.c:   == Spawn extension (ServiceNumbers, 104, 7) exited non-zero on 'Local/821@ServiceNumbers-754b,2'
[Mar 20 10:20:49] NOTICE[20006] chan_gtalk.c: Don't know how to indicate condition '20'
[Mar 20 10:20:49] VERBOSE[20006] logger.c:   == Spawn extension (from-GTALK, tootainet@gmail.com, 4) exited non-zero on 'Gtalk/ivan.trudnai-2df4'
[Mar 20 10:20:50] NOTICE[3937] chan_gtalk.c: Whoa, didn't find call!

The last NOTICE seems strange.

--
Daniel
Comments:By: tootai (tootai) 2009-04-09 07:21:56

Hi, any news about this matter? Thanks for feedback.
By: Leif Madsen (lmadsen) 2009-04-09 10:19:31

This is currently not a blocking issue for any releases, and thus might not get the quick response you are looking for. Generally chan_gtalk is going to have lower priority than issues with other channels such as chan_sip, so might be a bit until someone can come along and confirm this issue by testing and reproducing.

You could potentially get more traction if you ask on #asterisk-users for some assistance in tracking down the issue, and determining if anyone else has this issue.

Additionally, you can help by checking if previous versions worked, and if this is a regression, or some edge case that isn't working, in which case providing information about the scenario where it does and/or does not work would be useful.

Thanks!
By: Clod Patry (junky) 2009-04-09 21:00:49

Also, could you include all jabber messages for you account tootainet@gmail.com.

thanks.
By: tootai (tootai) 2009-04-10 07:20:30

Attached you will found jabber debug message. The incoming call is finish on Asterisk demo stuff (congratulation, you have successfull aso ...)

Thanks, Daniel.
By: phsultan (phsultan) 2009-04-10 08:15:03

What's the version of the remote Gtalk client? Some versions won't work with Asterisk : http://bugs.digium.com/view.php?id=10512

Please also check that you're not falling into that issue : http://bugs.digium.com/view.php?id=13985
By: tootai (tootai) 2009-04-10 10:30:28

Gtalk client is 1.0.0.105. Also the /etc/hosts issue doesn't apply to my setup. Please note that I connected Gtalk calls to the demo of our Asterisk test platform, feel free to call our account to check if you face the problem.

Thanks for your help and time

--
Daniel
By: phsultan (phsultan) 2009-04-22 03:15:39

Daniel, I just tried to call your test platform but my call has not been answered and your contact just disappeared, I'm sorry if I was disturbing. Maybe you can just set up an Echo server at this address if you didn't already.

As for the GoogleTalk client, I'm using version 1.0.0.104, and I was not able to upgrade it to 1.0.0.105. In fact, I can find a googletalk-setup.exe file on the Web that matches with version 1.0.0.105, but my client stays in 1.0.0.104 after I installed the so called new version.

Also, when I check for updates in my GoogleTalk client interface, the client keeps quiet.

Can you post a link where to find the 1.0.0.105 version?
By: tootai (tootai) 2009-04-22 11:24:43

Philippe,

your call *WAS* answered, you simply face the problem ;-). The extension you where redirected is the demo setup from original Asterisk. Hit ext 600 to get the echo test. Here are the console logs:

   -- Executing [tootainet@gmail.com@from-GTALK:1] NoOp("Gtalk/philippe.sultan-c794", "Incoming GoogleTALK call from "philippe.sultan@gmail.com/Talk.v104414E950E" <> to tootainet@gmail.com") in new stack
   -- Executing [tootainet@gmail.com@from-GTALK:2] Set("Gtalk/philippe.sultan-c794", "CALLERID(name)=GoogleTALK ") in new stack
   -- Executing [tootainet@gmail.com@from-GTALK:3] Set("Gtalk/philippe.sultan-c794", "__DIALEDNUMBER=800") in new stack
   -- Executing [tootainet@gmail.com@from-GTALK:4] Dial("Gtalk/philippe.sultan-c794", "Local/800@ServiceNumbers/n") in new stack
   -- Executing [800@ServiceNumbers:1] Goto("Local/800@ServiceNumbers-e435,2", "s|1") in new stack
   -- Goto (ServiceNumbers,s,1)
   -- Called 800@ServiceNumbers/n
   -- Executing [s@ServiceNumbers:1] Wait("Local/800@ServiceNumbers-e435,2", "1") in new stack
   -- Executing [s@ServiceNumbers:2] Answer("Local/800@ServiceNumbers-e435,2", "") in new stack
   -- Local/800@ServiceNumbers-e435,1 answered Gtalk/philippe.sultan-c794
[Apr 22 10:03:36] NOTICE[711]: chan_gtalk.c:1514 gtalk_indicate: Don't know how to indicate condition '-1'
[Apr 22 10:03:36] NOTICE[711]: chan_gtalk.c:1514 gtalk_indicate: Don't know how to indicate condition '20'
   -- Executing [s@ServiceNumbers:3] Set("Local/800@ServiceNumbers-e435,2", "TIMEOUT(digit)=5") in new stack
   -- Digit timeout set to 5
   -- Executing [s@ServiceNumbers:4] Set("Local/800@ServiceNumbers-e435,2", "TIMEOUT(response)=10") in new stack
   -- Response timeout set to 10
   -- Executing [s@ServiceNumbers:5] BackGround("Local/800@ServiceNumbers-e435,2", "demo-congrats") in new stack
   -- <Local/800@ServiceNumbers-e435,2> Playing 'demo-congrats' (language 'en')
[Apr 22 10:03:36] NOTICE[711]: chan_gtalk.c:1514 gtalk_indicate: Don't know how to indicate condition '20'
[Apr 22 10:03:36] NOTICE[711]: chan_gtalk.c:1514 gtalk_indicate: Don't know how to indicate condition '-1'
[Apr 22 10:03:36] NOTICE[711]: chan_gtalk.c:1514 gtalk_indicate: Don't know how to indicate condition '20'
 == Spawn extension (ServiceNumbers, s, 5) exited non-zero on 'Local/800@ServiceNumbers-e435,2'
 == Spawn extension (from-GTALK, tootainet@gmail.com, 4) exited non-zero on 'Gtalk/philippe.sultan-c794'

for Gtalk, I took it from their web site at this address -it was a new installation-:

http://www.google.com/talk/intl/fr/#utm_source=fr-et-more&utm_medium=et&utm_campaign=fr
By: tootai (tootai) 2009-05-14 03:18:22

Philippe, any news on this?
By: phsultan (phsultan) 2009-05-15 03:37:24

Hi Daniel, sorry for not getting back to you on this bug. Please keep your Gtalk client online so that I can test it, I'll give it a try in the next few days.
By: peterx86 (peterx86) 2009-05-19 08:27:57

Hi This is Peter. For my hobby, I tried myself to write some code to call from gmail to sip.

(1)what i do is to test my concept, i have a separate process to work as back to back call gateway. The gmail call to my program (which works as xmpp external component), my program call asterisk through SIP and do session negotiate.

(2)Ring and answer all works. I am using pcmu because i know google talk use different payload type constants except a few types. Anyway, for testing concept, it is fine.

(3)After answer, only SIP phone can hear voice. It is within my expectation because i have not do SIP re-invite to try different candidates. Even i do re-invite, it can only support pcmu,a types. It is not what i want.

Now, i stopped. I am thinking what is the simplest way to do? google jingle has very complex p2p negotiation. Also, i must do relay to do payload rewriting.

Question:
(A)Must i simulate the p2p negotiation process?
(B)Or just relay.

Because i am not familiar with asterisk channel development, i have no idea how to go through the chan_gtalk.c file.

By the way, if i want to test chan_gtalk, which asterisk version should i use?

Can you share some idea?

Best Regards
Peter
By: phsultan (phsultan) 2009-05-19 09:32:38

Daniel, I just installed a 1.4.23 version, and succesfully tested the Gtalk connection.

I also made a call to your Asterisk Gtalk account using my Gtalk client (1.0.0.104), again with success. However, I could not go through the menus of the IVR as thr Gtalk client cannot send DTMF to Asterisk (maybe you can check the jabberreceive branch for that : https://issues.asterisk.org/view.php?id=12569).

You can contact me directly if needed.
By: Leif Madsen (lmadsen) 2009-05-19 13:04:55

So what do we need to do with this issue? Is there a bug here, or can this issue be closed?
By: Leif Madsen (lmadsen) 2009-06-16 07:40:39

Closing this issue as unable to reproduce. Please reopen the issue if you feel this was done in error. Thanks!