[asterisk-bugs] [Asterisk 0017005]: Asterisk sends session-timer with "require" after 15 minutes

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17005 
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Reported By:                alexrecarey
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17005
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.6.1.17 
JIRA:                       SWP-1067 
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:             2010-03-10 21:02 CST
Last Modified:              2010-06-04 19:01 CDT
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Summary:                    Asterisk sends session-timer with "require" after 15
minutes
Description: 
When asterisk is set to defaults in session timer handling, after 15
minutes it will send a re-invite with the "required" tag.

This will cause an asterisk server with session-timers=refuse setting to
respond with a sip 420 and disconnect the call.

According to developer documentation at
https://issues.asterisk.org/file_download.php?file_id=15454&type=bug
asterisk should NEVER send a session timer with the "require" tag even if
set to session-timers=force

This issue is easily reproduced by registering a sip client with one
asterisk, set to session-timers=refuse, and place a call to another
asterisk with default session-timers settings. After exactly 15 minutes the
call will drop.

I have included a SIP Trace of the last 30 seconds of the call.
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 (0122998) mpi (reporter) - 2010-06-04 19:01
 https://issues.asterisk.org/view.php?id=17005#c122998 
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I can confirm the behavior mentioned by rbreidenstein with exactly 15
minutes between events.
It occurs for me on calls over 30 minutes.
As of an update on June 4, I am running Asterisk 1.6.2.8 (Elastix
2.0.0-rc2 with latest updates applied) and no setting for session-timer in
sip.conf so it is operating in default "accept" mode.
The events prior to June 4 are on a prior version of asterisk, but I did
not record which before the update.
The client is an Aastra 6731i with sip session timer set to 0 as mentioned
here: http://www.voip-info.org/wiki/view/Asterisk+and+Aastra+Phones


[Jun  1 11:12:37] VERBOSE[4863] pbx.c:     -- Executing
[91NXXNXXXXXX at from-internal:1] Macro("SIP/109-0000000c",
"user-callerid,SKIPTTL,") in new stack
[Jun  1 11:42:37] WARNING[3439] chan_sip.c: just did sched_add
waitid(14886) for sip_reinvite_retry for dialog 42882908f8fe8ddf in
handle_response_invite
[Jun  1 11:57:37] WARNING[3439] chan_sip.c: Re-invite to non-existing call
leg on other UA. SIP dialog '42882908f8fe8ddf'. Giving up.

[Jun  3 14:08:51] VERBOSE[24322] pbx.c:     -- Executing
[91NXXNXXXXXX at from-internal:1] Macro("SIP/109-00000060",
"user-callerid,SKIPTTL,") in new stack
[Jun  3 14:38:51] WARNING[3439] chan_sip.c: just did sched_add
waitid(230833) for sip_reinvite_retry for dialog 8625cc9644d54580 in
handle_response_invite
[Jun  3 14:53:51] WARNING[3439] chan_sip.c: Re-invite to non-existing call
leg on other UA. SIP dialog '8625cc9644d54580'. Giving up.

[Jun  4 15:11:53] VERBOSE[3843] pbx.c:     -- Executing
[91NXXNXXXXXX at from-internal:1] Macro("SIP/104-00000024",
"user-callerid,SKIPTTL,") in new stack
[Jun  4 15:45:47] WARNING[708] chan_sip.c: just did sched_add
waitid(17947) for sip_reinvite_retry for dialog 67d27075a945a1dd in
handle_response_invite
[Jun  4 16:00:47] WARNING[708] chan_sip.c: Re-invite to non-existing call
leg on other UA. SIP dialog '67d27075a945a1dd'. Giving up.


As a test, I then added:
session-timers=refuse
to sip.conf and restarted asterisk.
I placed a 60 minute call without receiving the WARNING messages so it
would appear that in my case, the session-timers=refuse setting seems to
have resolved the issue.
I don't know how to do a sip trace so I don't have as much detail to
determine if the fault lies with asterisk or the Aastra handset. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-06-04 19:01 mpi            Note Added: 0122998                          
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