[asterisk-bugs] [Asterisk 0017236]: [regression] X-Lite disconnects because of RTCP timeout when Local channel and MeetMe/F involved
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue May 18 13:25:59 CDT 2010
The following issue requires your FEEDBACK.
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https://issues.asterisk.org/view.php?id=17236
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Reported By: dimas
Assigned To: dvossel
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Project: Asterisk
Issue ID: 17236
Category: Channels/chan_sip/Interoperability
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Target Version: 1.6.2.9
Asterisk Version: 1.6.2.6
JIRA: SWP-1335
Regression: Yes
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2010-04-23 11:09 CDT
Last Modified: 2010-05-18 13:25 CDT
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Summary: [regression] X-Lite disconnects because of RTCP
timeout when Local channel and MeetMe/F involved
Description:
The dialplan:
context test1 {
start => {
Answer;
Read(code,conf-getpin);
MeetMe(1234,F);
}
test => {
Answer;
Dial(Local/start at test1/);
}
}
1. use X-Lite
2. place a call to test at test1 somehow
3. you are asked for a PIN.
4. enter anything and press #
5. You hear you are only person in the conference
6. 30 seconds later X-Lite hangs up the call because of RTCP inactivity
Note that X-Lite must have RTCP timeout enabled (this is default
setting):
Right click -> Options -> Advanced -> Network ->
In times of network disruption automatically hang up the call after =>
ON
RTCP had been inactive for => 30 seconds
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(0122068) dvossel (administrator) - 2010-05-18 13:25
https://issues.asterisk.org/view.php?id=17236#c122068
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I tried to reproduce this using X-Lite 2.0 for linux and was not successful
(using the same dialplan supplied in the description). Asterisk sent RTCP
updates correctly and the phone never hung up. I was not however able to
find the "In times of network disruption automatically hang up the call
after" option in this version. What version of X-Lite are you all using?
Can someone please provide us with a packet capture with SIP and RTP
traffic on a call that fails.
Issue History
Date Modified Username Field Change
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2010-05-18 13:25 dvossel Note Added: 0122068
2010-05-18 13:25 dvossel Status acknowledged =>
feedback
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