[asterisk-bugs] [Asterisk 0019214]: Call declined when rtupdate=no
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu May 5 09:18:13 CDT 2011
The following issue requires your FEEDBACK.
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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: 2011-05-05 09:18 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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(0134486) lmadsen (administrator) - 2011-05-05 09:18
https://issues.asterisk.org/view.php?id=19214#c134486
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Please provide from debugging information from the console, included SIP
debug and DEBUG level logging. I also think it would be useful to see what
SQL statements are being sent to the SQL server, and how it is responding,
so please provide the sql statement logs from the SQL server.
Issue History
Date Modified Username Field Change
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2011-05-05 09:18 lmadsen Note Added: 0134486
2011-05-05 09:18 lmadsen Status new => feedback
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