[asterisk-bugs] [Asterisk 0012353]: DTMF is boken on SIP trunks

[noreply at bugs.digium.com]

The following issue has been SUBMITTED. 
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http://bugs.digium.com/view.php?id=12353 
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Reported By:                dimas
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12353
Category:                   Core/PBX
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 110163 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             04-02-2008 06:56 CDT
Last Modified:              04-02-2008 06:56 CDT
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Summary:                    DTMF is boken on SIP trunks
Description: 
rfc2833 DTMF on outgoing calls do not for my voip provider (voipcheap.com)
when generic bridge is used (I'm using DTMF features).
With the packet2packet brigde everything works just fine.

After analyzing packet captures I found this happens because SSRC on RTP
packets Asterisk sends to voip provider changes each time DTMF is being
sent.

As a workaound on my installation I just commented out 

//      rtp->ssrc = ast_random();

in the ast_rtp_new_source (main/rtp.c) however it is clear, better
solution needs to be found.
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Issue History 
Date Modified   Username       Field                    Change               
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04-02-08 06:56  dimas          Asterisk Version          => SVN             
04-02-08 06:56  dimas          SVN Branch (only for SVN checkou =>  1.4         
  
04-02-08 06:56  dimas          SVN Revision (number only!) => 110163          
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