[asterisk-bugs] [Asterisk 0015014]: Asterisk loses SIP phones, possible deadlock, 1.6.1.0

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=15014 
====================================================================== 
Reported By:                madkins
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   15014
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.1.0 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-05-01 11:03 CDT
Last Modified:              2009-05-01 18:01 CDT
====================================================================== 
Summary:                    Asterisk loses SIP phones, possible deadlock,
1.6.1.0
Description: 
I have two Cisco 7905g SIP phones connected to an Asterisk 1.6.1.0 instance
running on a 64 bit Xen instance of Debian 4.0.  My initial configuration
was more complex, but I've removed a lot of the complexity searching for
the problem.

Basically, I can start the Asterisk server and pick up a SIP phone and
call either a test extension or the other phone.  Works fine.  If I leave
it alone for a time ... say over a long lunch or overnight ... I come back
and the phones won't connect to Asterisk.

This repeats reliably but at unknown intervals.

====================================================================== 

---------------------------------------------------------------------- 
 (0104047) madkins (reporter) - 2009-05-01 18:01
 http://bugs.digium.com/view.php?id=15014#c104047 
---------------------------------------------------------------------- 
i was given some advice about using nat=yes, but that was actually a
mistake, i had set nat=no in the general level of sip.conf but not removed
it from the individual sip phones, doing so now 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-05-01 18:01 madkins        Note Added: 0104047                          
======================================================================