[asterisk-bugs] [Asterisk 0013243]: [patch] Set(SIP_CODEC=xxxx) only applies to first inbound leg of call

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Aug 27 17:06:31 CDT 2008


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13243 
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Reported By:                samdell3
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13243
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            always
Severity:                   tweak
Priority:                   normal
Status:                     confirmed
Asterisk Version:           Older 1.4 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-08-05 18:32 CDT
Last Modified:              2008-08-27 17:06 CDT
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Summary:                    [patch] Set(SIP_CODEC=xxxx) only applies to first
inbound leg of call
Description: 
We have had a long standing requirement to be able to force the use of g711
codec based on dialled number, eg known modem destinations etc.
We still need to use g729 by default for voice calls.

The obvious choice is to Set(SIP_CODEC=alaw) prior to Dial()

However, SIP_CODEC only ever forced the inbound (first) leg of the call to
use alaw. If the outbound leg codec priority was 1st G729 2nd alaw, then
g729 was always used.

Attached is a very simple patch against 1.4.14 that solves our problem. It
works for both reinvited and non reinvited media.
Due to the patch only being 2 lines of additional code, it would be easy
to apply to later versions of Asterisk

It's now running in a production environment, but I would really like some
feedback from other users.

 
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---------------------------------------------------------------------- 
 (0091837) Corydon76 (administrator) - 2008-08-27 17:06
 http://bugs.digium.com/view.php?id=13243#c91837 
---------------------------------------------------------------------- 
I'm still a little uncomfortable about committing changes like this to the
SIP channel driver, so I think I'd want someone else to look at this before
I commit it. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-08-27 17:06 Corydon76      Note Added: 0091837                          
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