[asterisk-bugs] [Asterisk 0019214]: Call declined when rtupdate=no

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
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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:                     acknowledged
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:              2011-05-10 17:17 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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Issue History 
Date Modified    Username       Field                    Change               
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2011-05-10 17:17 lmadsen        Description Updated                          
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