[asterisk-bugs] [Asterisk 0019214]: Call declined when rtupdate=no
Asterisk Bug Tracker
noreply at bugs.digium.com
Sat May 7 07:25:32 CDT 2011
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=19214
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Reported By: deepesh
Assigned To:
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Project: Asterisk
Issue ID: 19214
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.6.2.17.2
JIRA:
Regression: No
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: 2011-05-03 04:47 CDT
Last Modified: 2011-05-07 07:25 CDT
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Summary: Call declined when rtupdate=no
Description:
I am using asterisk 1.6.2.17 with realtime pgsql.
When I turn off rtupdate in sip.conf (rtupdate=no) sip peers are unable to
make calls. The SIP peers are able to register and it shows up in 'sip show
peer <peer>' also, but while making a call the call gets rejected.
A sip debug during the call shows the following
1. INVITE - from peer
After the invite the following message can be seen on the console:- No
matching peer for '1122334' from '191.168.1.10:11473'
2. 100 Trying - sent back to the peer
3. 603 Declined - sent back to the peer
When I switch on rtupdate, everything works fine.
Earlier I was using asterisk 1.6.2.7 and everything was working fine with
rtupdate=no.
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(0134628) deepesh (reporter) - 2011-05-07 07:25
https://issues.asterisk.org/view.php?id=19214#c134628
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I also have rtcachefriends=yes. I have attached the SQL statement logs and
the output of SIP debug.
Issue History
Date Modified Username Field Change
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2011-05-07 07:25 deepesh Note Added: 0134628
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