[asterisk-bugs] [Asterisk 0012353]: Generated DTMF (from features) broken with some SIP providers

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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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:                     acknowledged
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:              04-12-2008 21:48 CDT
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Summary:                    Generated DTMF (from features) broken with some SIP
providers
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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---------------------------------------------------------------------- 
 tasker - 04-12-08 21:48  
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Isn't broken DTMF over SIP considered critical? This issues should be
prioritized as critical, not normal.

I had to revert back to 1.4.18.1 because it broke DTMF with nearly every
provider on my network. 

Issue History 
Date Modified   Username       Field                    Change               
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04-12-08 21:48  tasker         Note Added: 0085414                          
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