[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
Thu Feb 12 13:28:38 CST 2009


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=13243 
====================================================================== 
Reported By:                samdell3
Assigned To:                file
====================================================================== 
Project:                    Asterisk
Issue ID:                   13243
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            always
Severity:                   tweak
Priority:                   normal
Status:                     assigned
Asterisk Version:           Older 1.4 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2008-08-05 18:32 CDT
Last Modified:              2009-02-12 13:28 CST
====================================================================== 
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.

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

---------------------------------------------------------------------- 
 (0100032) file (administrator) - 2009-02-12 13:28
 http://bugs.digium.com/view.php?id=13243#c100032 
---------------------------------------------------------------------- 
Right, while this patch may not be the solution we are discussing where we
should go and what we should do to fix this in the proper manner. The issue
I have with changing SIP_CODEC to be outbound is compatibility with all the
dialplans that use it already. If we change the behavior and they don't
change it... things will break. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-02-12 13:28 file           Note Added: 0100032                          
======================================================================




More information about the asterisk-bugs mailing list