[asterisk-bugs] [Asterisk 0011638]: Can't make two channels compatible if they have common video codec

noreply at bugs.digium.com noreply at bugs.digium.com
Wed Dec 26 21:45:22 CST 2007


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=11638 
====================================================================== 
Reported By:                cwhuang
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   11638
Category:                   Channels/chan_sip/Video
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 94824 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             12-26-2007 21:42 CST
Last Modified:              12-26-2007 21:45 CST
====================================================================== 
Summary:                    Can't make two channels compatible if they have
common video codec
Description: 
I'm testing asterisk 1.4 with some video phones. I usually got such
errors:

[Dec 27 11:24:53] WARNING[13425]: channel.c:3038 set_format: Unable to
find a codec translation path from unknown to unknown
[Dec 27 11:24:53] WARNING[13425]: channel.c:3423
ast_channel_make_compatible: Unable to set read format on channel
SIP/401-08ee0760 to 524288
[Dec 27 11:24:53] WARNING[13425]: app_dial.c:1685 dial_exec_full: Had to
drop call because I couldn't make SIP/401-08ee0760 compatible with
SIP/405-08ef0908

One of my phone has codec (gsm|h263), the other has (g729|h263). They
should be compatible.
====================================================================== 

---------------------------------------------------------------------- 
 russell - 12-26-07 21:45  
---------------------------------------------------------------------- 
Please provide a SIP debug trace of the call when the problem occurs. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
12-26-07 21:45  russell        Note Added: 0075993                          
======================================================================




More information about the asterisk-bugs mailing list