[asterisk-bugs] [Asterisk 0015906]: rtptimeout option doesn't work for inbound calls
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Sep 17 10:38:02 CDT 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=15906
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Reported By: makoto
Assigned To:
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Project: Asterisk
Issue ID: 15906
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.6.1.6
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-09-15 21:05 CDT
Last Modified: 2009-09-17 10:38 CDT
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Summary: rtptimeout option doesn't work for inbound calls
Description:
rtptimeout option for peers works for outbound calls,
but doesn't work for inbound calls,
Global rtptimeout option works for both directions.
Steps:
1. Set global rtptimeout option to 0
2. Set rtptimeout option for Phone A to 30
3. Call from Phone B to Phone A
4. Unplug network cable from Phone A
5. Wait 30 seconds
6. Asterisk disconnects the call
7. Call from Phone A to Phone B
8. Unplug network cable from Phone A
9. Wait 30 seconds
10. Asterisk should disconnect the call, but it doesn't
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(0110869) davidw (reporter) - 2009-09-17 10:38
https://issues.asterisk.org/view.php?id=15906#c110869
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This wouldn't have worked for 1.6.0.x or earlier, as noted in
https://issues.asterisk.org/view.php?id=13056. My
understanding was that the merging of sip_user into sip_peer that happened
for 1.6.1.x was supposed to have removed the problem, although I haven't
tested this as it hasn't been a priority recently, and I only checked that
the merging of the structures had occured.
Issue History
Date Modified Username Field Change
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2009-09-17 10:38 davidw Note Added: 0110869
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