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

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Nov 18 23:00:01 CST 2009


A NOTE has been added to this issue. 
====================================================================== 
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:              
====================================================================== 
Date Submitted:             2009-08-03 00:59 CDT
Last Modified:              2009-11-18 23:00 CST
====================================================================== 
Summary:                    [branch] psi 0.13 can't call chan_jingle
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...

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

---------------------------------------------------------------------- 
 (0113970) sles (reporter) - 2009-11-18 23:00
 https://issues.asterisk.org/view.php?id=15634#c113970 
---------------------------------------------------------------------- 
Hello!

I almost copied your config:

[general]
context=default
allowguest=yes
debug=yes
bindaddr=192.168.22.229
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=speex

[guest]
disallow=all
allow=ulaw
allow=alaw
allow=speex
context=jingle-in

(btw, only speex was used before just because I allowed only speex).

and extensions.conf :

[jingle-in]
exten => s,1,NoOp(In jingle context);
exten => s,n,Dial(SIP/6052)
;exten => s,1,Answer
;exten => s,n,Playback(beep)
;exten => s,n,Echo
exten => s,n,Hangup


Result is the same:

Call from windows psi 0.14pre1:

*CLI> [Nov 19 08:48:05] NOTICE[22047]: chan_jingle.c:811 jingle_new: Codec
initialized to : G.711 u-law
  == Starting Jingle/kasimov-3882 at jingle-in,s,1 failed so falling back
to exten 's'
  == Starting Jingle/kasimov-3882 at jingle-in,s,1 still failed so falling
back to context 'default'
[Nov 19 08:48:05] WARNING[22223]: pbx.c:4710 __ast_pbx_run: Channel
'Jingle/kasimov-3882' sent into invalid extension 's' in context 'default',
but no invalid handler


Call from Empathy:

*CLI> [Nov 19 08:41:39] NOTICE[22047]: chan_jingle.c:811 jingle_new: Codec
initialized to : G.711 u-law
  == Starting Jingle/dm-eb6b at jingle-in,s,1 failed so falling back to
exten 's'
  == Starting Jingle/dm-eb6b at jingle-in,s,1 still failed so falling back
to context 'default'
[Nov 19 08:41:39] WARNING[22129]: pbx.c:4710 __ast_pbx_run: Channel
'Jingle/dm-eb6b' sent into invalid extension 's' in context 'default', but
no invalid handler

May be this is because I compiled asterisk on 64-bit host ?- my desktop,
ubuntu 9.10, amd64.... 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-11-18 23:00 sles           Note Added: 0113970                          
======================================================================




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