[asterisk-bugs] [Asterisk 0012353]: Generated DTMF (from features) broken with some SIP providers
noreply at bugs.digium.com
noreply at bugs.digium.com
Mon Apr 14 08:47:52 CDT 2008
The following issue has been ASSIGNED.
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http://bugs.digium.com/view.php?id=12353
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Reported By: dimas
Assigned To: file
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Project: Asterisk
Issue ID: 12353
Category: Core/PBX
Reproducibility: always
Severity: minor
Priority: normal
Status: assigned
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: 04-14-2008 08:47 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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svnbot - 04-14-08 08:47
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Repository: asterisk
Revision: 114100
U branches/1.4/main/rtp.c
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r114100 | file | 2008-04-14 08:47:49 -0500 (Mon, 14 Apr 2008) | 4 lines
Don't change the SSRC when a new source comes into play, this might happen
quite often and depending on the remote side... they might not like this.
(closes issue http://bugs.digium.com/view.php?id=12353)
Reported by: dimas
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http://svn.digium.com/view/asterisk?view=rev&revision=114100
Issue History
Date Modified Username Field Change
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04-14-08 08:47 svnbot Checkin
04-14-08 08:47 svnbot Note Added: 0085443
04-14-08 08:47 svnbot Status acknowledged => assigned
04-14-08 08:47 svnbot Assigned To => file
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