[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
Mon Dec 10 15:06:52 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: 12-10-2007 15:06 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 - 12-10-07 15:06
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the following patch against: channels/chan_sip.c
will force the use of rfc2833 if rfc2833 is configured in sip.conf:
5357c5357,5361
<
---
>
> /* If we locally define rfc2833, then force the use of rfc2833 */
> if ((p->noncodeccapability & AST_RTP_DTMF) == AST_RTP_DTMF) {
> newnoncodeccapability |= AST_RTP_DTMF;
> }
Please consider this patch. This would allow for asterisk-1.4.15 and later
to continue to be compatible to cisco equipment when using sip trunks.
Issue History
Date Modified Username Field Change
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12-10-07 15:06 macbrody Note Added: 0075162
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