[asterisk-bugs] [Asterisk 0013045]: No voice joining snom 190 through asterisk to cisco voice gateway occationally.

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=13045 
====================================================================== 
Reported By:                kactus
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13045
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            sometimes
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.17 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             07-10-2008 03:48 CDT
Last Modified:              07-15-2008 06:03 CDT
====================================================================== 
Summary:                    No voice joining snom 190 through asterisk to cisco
voice gateway occationally.
Description: 
Hello

I have a asterisk 1.4.17 box which I have configured to troubleshoot case
12708 and while it resolves that issue, I have come across a concerning bug
that has now occured 3 times in 1.5 days of testing. 

I have a snom 190 talking to asterisk via sip, talking to a cisco voice
gateway via sip. Everything is in realtime talking via odbc and freetds to
a mssql db. 

Calling out works fine most of the time however occationally (3 times)
when I call, the call connects, but no voice traverses in either
direction.

I will attach two sip debugs as I was able to catch it in action, but
please note that rtp debug and rtcp debug was not turned on until part way
through on the broken one. 

Things I noticed: when the first "invite sip" occurs on the broken one, it
does not pass media  or media atributes. Later on it adds ilbc to the
supported codecs and sets mode to 30ms.
====================================================================== 

---------------------------------------------------------------------- 
 kactus - 07-15-08 06:03  
---------------------------------------------------------------------- 
Hello

I've uploaded pbx and gateway debugs as requested by jsmith in the
#asterisk-bugs channel. I had to zip the gateway file as at 1844 KB it was
throwing database size errors on upload.

The number that goes dead is DeadNum in the configs.

pbx debugs are from our office trixbox.

Gateway debugs are from the 1.4.17 box running as a gateway to the cisco
isdn gateway.

This was all done with Verbose at least 4, debug at least 4, sip debug,
iax debug and rtp debug.

Please let me know what you require further. If I have ommited something
and you would like more around debug from around the call, I still have the
original log files 714MB and 138MB respectively. I have had to reformat the
debugs as the putty loggging seemed to have difficulty as it progressed and
started substituting characters.

Thank you 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
07-15-08 06:03  kactus         Note Added: 0090272                          
======================================================================