|Summary:||ASTERISK-17783: Call declined when rtupdate=no|
|Date Opened:||2011-05-03 04:47:52||Date Closed:||2011-07-26 14:20:45|
|Environment:||Attachments:||( 0) psql_log_with_rtupdate.txt|
( 1) psql_log_without_rtupdate.txt
( 2) sip_debug_with_rtupdate.txt
( 3) sip_debug_without_rtupdate.txt
|Description:||I am using asterisk 184.108.40.206 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 '220.127.116.11:11473'
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 18.104.22.168 and everything was working fine with rtupdate=no.
|Comments:||By: Leif Madsen (lmadsen) 2011-05-05 09:18:13|
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.
By: deepesh (deepesh) 2011-05-07 07:25:32
I also have rtcachefriends=yes. I have attached the SQL statement logs and the output of SIP debug.
By: Russell Bryant (russell) 2011-07-26 14:20:38.048-0500
Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.4 and 1.6.x branches has ended. For continued maintenance support please move to the 1.8 branch which is a long term support (LTS) branch. For more information about branch support, please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
If this is still an issue, please open a new issue so it can be re-triaged appropriately. Thanks!