[asterisk-bugs] [Asterisk 0015922]: Asterisk generates a BYE after 15 minutes or more consistently on trunk calls

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
====================================================================== 
https://issues.asterisk.org/view.php?id=15922 
====================================================================== 
Reported By:                Micc
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   15922
Category:                   Channels/chan_sip/General
Reproducibility:            have not tried
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.1.6 
JIRA:                        
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-09-19 23:30 CDT
Last Modified:              2009-09-21 09:42 CDT
====================================================================== 
Summary:                    Asterisk generates a BYE after 15 minutes or more
consistently on trunk calls
Description: 
All calls I have made from phone to asterisk to phone works fine for long
calls. No problem. But when I make a call from phone to asterisk to sip
provider to asterisk to phone, I notice asterisk generates a BYE at random
time, usually after 15 to 20 minutes. I've never seen it happen before 15
minutes. I've done sip debug and sip trace, neither show any other packets
except the RTP traffic working perfectly, then all the sudden asterisk
sends a BYE sip packet and the call drops. This happens when calling a PSTN
number as well, or another asterisk server over IAX2.
====================================================================== 

---------------------------------------------------------------------- 
 (0111061) lmadsen (administrator) - 2009-09-21 09:42
 https://issues.asterisk.org/view.php?id=15922#c111061 
---------------------------------------------------------------------- 
As per the bug guidelines at
http://www.asterisk.org/developers/bug-guidelines you need to provide the
necessary information when reporting SIP issues:

SIP problem?
Include debug output! Please include output from "sip debug" if you have a
SIP problem. This seems obvious, but apparently is not. Set debug to 4,
verbose to 4, turn on sip history and dumphistory in sip.conf and capture
all output. A packet trace from ethereal will not tell us what is happening
inside your Asterisk server, so that is not a replacement.

I have a suspicion something with the RTP timers is happening that is
causing Asterisk to drop the call because it isn't seeing RTP traffic when
it expects to.

However, we'll need to see the SIP traces as described above to be
certain.

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-09-21 09:42 lmadsen        Note Added: 0111061                          
2009-09-21 09:42 lmadsen        Status                   new => feedback     
======================================================================