[asterisk-bugs] [Asterisk 0014168]: dahdi timing doesn't work on sip channel in non-answer mode.

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Feb 24 11:36:39 CST 2009


The following issue requires your FEEDBACK. 
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http://bugs.digium.com/view.php?id=14168 
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Reported By:                fsantulli
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   14168
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.0 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0 
SVN Revision (number only!): 167059 
Request Review:              
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Date Submitted:             2009-01-04 12:40 CST
Last Modified:              2009-02-24 11:36 CST
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Summary:                    dahdi timing doesn't work on sip channel in
non-answer mode.
Description: 
I'm used to play an announcement to the caller during "Session in Progress
(183)", next the user is added to a Queue and the caller hears queue
sounds.
I have installed dahdi, and configured kernel to use rtc timing, next i
added in "asterisk.conf" the option "internaltiming=yes" to use dahdi
pseudo timing as my SIP provider has enabled "Comfort Noise Support" and we
need to send audio frames without waiting for provider's ones.
BTW dahdi timing works if i answer the call, but not if playback or queue
are called with non answer options or before answer.
I didn't tried if Dial application does the same. I will report later.
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Relationships       ID      Summary
----------------------------------------------------------------------
related to          0012688 serious problems in VAD and CNG support
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---------------------------------------------------------------------- 
 (0100633) file (administrator) - 2009-02-24 11:36
 http://bugs.digium.com/view.php?id=14168#c100633 
---------------------------------------------------------------------- 
Can you confirm your configuration please? The option is internal_timing
under the options context in asterisk.conf. Console output and dialplan
logic to reproduce it would also be useful as I have tried different ways
and failed miserably to make it break. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-02-24 11:36 file           Note Added: 0100633                          
2009-02-24 11:36 file           Status                   assigned => feedback
======================================================================




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