[asterisk-bugs] [Asterisk 0015634]: psi 0.13 can't call chan_gtalk

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Oct 27 02:09:34 CDT 2009


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15634 
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Reported By:                sles
Assigned To:                phsultan
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Project:                    Asterisk
Issue ID:                   15634
Category:                   Channels/chan_jingle
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-08-03 00:59 CDT
Last Modified:              2009-10-27 02:09 CDT
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Summary:                    psi 0.13 can't call chan_gtalk
Description: 
Hello!

psi 0.13 is just released, with jingle support.
it is not gtalk compatible, but, according to developers , it is jingle
specification compatible
http://forum.psi-im.org/thread/5335
it will be good if psi will work with asterisk...

====================================================================== 

---------------------------------------------------------------------- 
 (0112792) sles (reporter) - 2009-10-27 02:09
 https://issues.asterisk.org/view.php?id=15634#c112792 
---------------------------------------------------------------------- 
Hello!

I have following problem with call from psi-0.13- no sound.
Here is what I see on console:

*CLI> module  load chan_jingle
*CLI>   == Parsing '/etc/asterisk/jingle.conf':   == Found
  == Registered RTP glue 'Jingle'
  == Registered channel type 'Jingle' (Jingle Channel Driver)
 Loaded chan_jingle.so => (Jingle Channel Driver)
*CLI> 
*CLI> 
*CLI> [Oct 27 11:02:58] NOTICE[653]: chan_jingle.c:811 jingle_new: Codec
initialized to : SpeeX
    -- Executing [s at jabber-in:1] NoOp("Jingle/jnk-e2ed", " Call from Gtalk
") in new stack
    -- Executing [s at jabber-in:2] Set("Jingle/jnk-e2ed",
"CALLERID(name)=”From Google Talk”") in new stack
    -- Executing [s at jabber-in:3] MeetMe("Jingle/jnk-e2ed", "6000") in new
stack
[Oct 27 11:02:58] NOTICE[676]: chan_jingle.c:1349 jingle_indicate: Don't
know how to indicate condition '-1'
[Oct 27 11:02:58] NOTICE[653]: chan_jingle.c:1174 jingle_add_candidate:
Processing candidate
[Oct 27 11:02:58] NOTICE[653]: chan_jingle.c:1174 jingle_add_candidate:
Processing candidate
[Oct 27 11:02:58] NOTICE[653]: chan_jingle.c:1206 jingle_add_candidate:
Candidates parsed
  == Parsing '/etc/asterisk/meetme.conf':   == Found
[Oct 27 11:02:59] WARNING[676]: channel.c:4535 ast_request: No channel
type registered for 'DAHDI'
    -- Created MeetMe conference 1023 for conference '6000'
    -- <Jingle/jnk-e2ed> Playing 'conf-onlyperson.gsm' (language 'en')
[Oct 27 11:03:02] NOTICE[676]: chan_jingle.c:1349 jingle_indicate: Don't
know how to indicate condition '-1'
[Oct 27 11:03:06] WARNING[676]: frame.c:1385 speex_samples: Not enough
bits remaining after wide band for speex samples.
[Oct 27 11:03:06] WARNING[676]: frame.c:1385 speex_samples: Not enough
bits remaining after wide band for speex samples.
[Oct 27 11:03:06] WARNING[676]: frame.c:1385 speex_samples: Not enough
bits remaining after wide band for speex samples.
[Oct 27 11:03:06] WARNING[676]: frame.c:1385 speex_samples: Not enough
bits remaining after wide band for speex samples. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-10-27 02:09 sles           Note Added: 0112792                          
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