[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. 
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http://bugs.digium.com/view.php?id=13045 
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Reported By:                kactus
Assigned To:                
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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:              
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Date Submitted:             07-10-2008 03:48 CDT
Last Modified:              07-13-2008 20:51 CDT
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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.
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---------------------------------------------------------------------- 
 kactus - 07-13-08 20:51  
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Hmm it appears that its not an issue with the snom talking sip to the
asterisk box after all.

I setup the office pbx (trixbox) to call out via the asterisk 1.4.17 as
the default outbound trunk via IAX. We only have a skeleton crew on
weekends so thought this would not effect too many people.

Cisco 7960/7940 handsets -SIP-> trixbox 2.6 -IAX2-> Asterisk 1.4.17 -SIP->
Cisco call gateway -PRI-> The world.

I'm going to upload a debug that shows 4 calls going through. The first
works, the next three are to the same number. The call at 10:09 and 10:36
comes through as silence once it stops ringing. The call at just before
10:39 works without issue.

We had one call to a different number provide the same behaviour but this
was before I turned on sip and iax debugging.

Just to clarify something that might throw you in the debug, the outbound
number is prefixed with an 8 for all calls. This is including the
workingOutboundNumber even though it doesn't show it (as after I had
replaced all I realised it might be useful to see when it is working with
the pre and post processed number.

Let me know how you would like me to proceed or if there is anything else
you would like me to test. I've since reverted to our old gateway box so I
can do anything to this one. 

Issue History 
Date Modified   Username       Field                    Change               
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07-13-08 20:51  kactus         Note Added: 0090182                          
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