[asterisk-bugs] [Asterisk 0015642]: [patch] Fix for Sonus DTMF issues
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Mar 2 15:46:15 CST 2011
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=15642
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Reported By: jasonshugart
Assigned To: twilson
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Project: Asterisk
Issue ID: 15642
Category: Core/RTP
Reproducibility: always
Severity: feature
Priority: normal
Status: assigned
Asterisk Version: SVN
JIRA: SWP-2728
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-08-03 12:28 CDT
Last Modified: 2011-03-02 15:46 CST
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Summary: [patch] Fix for Sonus DTMF issues
Description:
In some cases when Asterisk sends DTMF to Sonus platforms as RFC2833, Sonus
will not recognize the DTMF. Several articles have identified the problem
as being the gap in the audio prior to the DTMF packets being sent. This
patch sends a single G.711 ulaw packet prior to the rfc2833 packets. In
our testing across with two carriers (Level3 and 360 Networks) this patch
fixed our DTMF issues. The rtp.c file seems very similar for the 1.4
branch, so minor changes could also be applied there. I added an option to
the rtp.conf file to enable this fix, called rtpfixdtmf. Attached are both
the rtp.c fix, and the rtp.conf change.
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Relationships ID Summary
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has duplicate 0016625 RFC2833 DTMF is not passed correctly wh...
related to 0017404 [patch] [regression] audio delay when b...
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(0132543) eeman (reporter) - 2011-03-02 15:46
https://issues.asterisk.org/view.php?id=15642#c132543
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my experience with broadvox seemed to always happen on or around the 7th
digit. If I were to call up a number that was directly connected to the
PSTN (like an asterisk box with a pri) that went to a macro that would
collect and read back DTMF, I would use a sequence like 1234567890, usually
around 7 or 8 the digit would get skipped entirely. Make sure when you
testing you use 10 or more digits. When testing shorter numbers its harder
to get failures.
BTW for those on broadvox experiencing pain; we switched to sip info
method of DTMF 6mos ago until this issue gets resolved and havent regretted
it. The only downside is no Packet2Packet transfers but that has not yet
created a problem as we are only 150 - 200 concurrent calls.
Issue History
Date Modified Username Field Change
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2011-03-02 15:46 eeman Note Added: 0132543
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