[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
Sun Dec 9 05:01:50 CST 2007
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=11489
======================================================================
Reported By: macbrody
Assigned To:
======================================================================
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:
======================================================================
Date Submitted: 12-07-2007 07:36 CST
Last Modified: 12-09-2007 05:01 CST
======================================================================
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.
======================================================================
----------------------------------------------------------------------
macbrody - 12-09-07 05:01
----------------------------------------------------------------------
Eliel,
working sip trace is uploaded as requested, thanks for your quick response
btw.
In the working trace and the non working trace the sdp in the invite is
the same, but asterisk-1.4.13 and asterisk-1.4.15 seem to interpret it
differently.
> [Dec 9 11:41:41] VERBOSE[23938] logger.c: Non-codec capabilities
(dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined
- 0x1 (telephone-event)
What may be the reason: Cisco does not use the telephony-event in the sdp
but their X-NSE events. So asterisk-1.4.13 and earlier seem to
accept/understand the
X-NSE events while asterisk-1.4.15 seems to ignore it.
Was it accidentally dropped? Or was it a particular reason? I have found a
long closed issue where the handling of X-NSE was disscussed in 2004: The
issue # is 3041.
Issue History
Date Modified Username Field Change
======================================================================
12-09-07 05:01 macbrody Note Added: 0075093
======================================================================
More information about the asterisk-bugs
mailing list