[asterisk-bugs] [Asterisk 0011489]: During call setup signalling asterisk does not offer telephone-event (rfc2833) anymore
noreply at bugs.digium.com
noreply at bugs.digium.com
Thu Dec 20 09:14:21 CST 2007
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=11489
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Reported By: macbrody
Assigned To:
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Project: Asterisk
Issue ID: 11489
Category: Channels/chan_sip/General
Reproducibility: always
Severity: block
Priority: normal
Status: feedback
Asterisk Version: 1.4.15
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!): 91736
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 12-07-2007 07:36 CST
Last Modified: 12-20-2007 09:14 CST
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Summary: During call setup signalling asterisk does not offer
telephone-event (rfc2833) anymore
Description:
What happened:
As of asterisk-1.4.15 (tested trunk too) asterisk, when
connected to a peer via sip trunk does not recognize dtmf
anymore (same config worked with 1.4.13).
What I found:
During signalling negotiation at call setup in the sdp body,
asterisk does not offer the rtpmap: telephone-event anymore,
even if I configure dtmfmode=rfc2833.
Therefore the other end will not send any rfc2833 specific rtp events.
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macbrody - 12-20-07 09:14
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Uploaded the patch: diff-main_rtp.c
As discussed on asterisk-dev this will tell asterisk
to accept the sdp option 'X-NSE' to be equal to 'telephone-event'.
X-NSE is used by cisco as a supplement to rfc2833. So if this occurs
we can assume that the cisco device on the other side uses rfc2833
even if they don't indicate it with telephone-event.
In fact when the other side sends 'X-NSE' they signal that they
understand
more than just the basic subset of rfc2833.
Issue History
Date Modified Username Field Change
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12-20-07 09:14 macbrody Note Added: 0075754
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