[asterisk-bugs] [Asterisk 0014295]: SIP on hold problems
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Jan 23 12:03:42 CST 2009
The following issue has been RESOLVED.
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http://bugs.digium.com/view.php?id=14295
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Reported By: klaus3000
Assigned To: file
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Project: Asterisk
Issue ID: 14295
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: resolved
Asterisk Version: 1.4.22
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
Resolution: fixed
Fixed in Version:
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Date Submitted: 2009-01-21 07:20 CST
Last Modified: 2009-01-23 12:03 CST
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Summary: SIP on hold problems
Description:
Hi!
I have found 2 problems when the SIP phone puts a call on hold (a SNOM
phone which uses a=sendonly).
1. Although only RTP should be activated it looks like RTCP is deactivated
too, as I got these error message on the console:
RTCP SR transmission error, rtcp halted
2. When the call is on hold and I unplug the phone, I would think that the
session is terminated after "rtpholdtimeout". Instead the session is
termianted after rtptimeout.
see trace below: rtptimeout=15, rtpholdtimeout=25
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Relationships ID Summary
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related to 0013835 "RTCP SR transmission error, rtcp ...
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(0098591) svnbot (reporter) - 2009-01-23 12:03
http://bugs.digium.com/view.php?id=14295#c98591
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Repository: asterisk
Revision: 170504
U branches/1.4/channels/chan_sip.c
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r170504 | file | 2009-01-23 12:03:41 -0600 (Fri, 23 Jan 2009) | 4 lines
Use the on hold flag to see if the call is on hold or not. It is possible
that our address for them will still be valid even though they are on hold.
(closes issue http://bugs.digium.com/view.php?id=14295)
Reported by: klaus3000
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http://svn.digium.com/view/asterisk?view=rev&revision=170504
Issue History
Date Modified Username Field Change
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2009-01-23 12:03 svnbot Checkin
2009-01-23 12:03 svnbot Note Added: 0098591
2009-01-23 12:03 svnbot Status assigned => resolved
2009-01-23 12:03 svnbot Resolution open => fixed
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