[asterisk-bugs] [Asterisk 0015945]: [patch] sip session timer: Does not work if initial INVITE min-se timer is too small

[Asterisk Bug Tracker]

The following issue is now READY FOR REVIEW. 
====================================================================== 
https://issues.asterisk.org/view.php?id=15945 
====================================================================== 
Reported By:                steinwej
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   15945
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     ready for review
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0 
SVN Revision (number only!): 219892 
Request Review:              
====================================================================== 
Date Submitted:             2009-09-23 10:55 CDT
Last Modified:              2009-09-24 09:04 CDT
====================================================================== 
Summary:                    [patch] sip session timer: Does not work if initial
INVITE min-se timer is too small
Description: 
Asterisk sip with session timer enabled.
sip.conf:
session-timers=accept
session-expires=600
session-minse=180


Patton box connected to asterisk. Patton sends INVITE with session timer
90

asterisk responds with 422 session interval too small
patton reinvites with the proposed session timer.
asterisk send 200 ok, nothing happens. no tones or anything.
When patton sends BYE, asterisk sends ACK
But
sip channels remains, audio ports are not released

voip-1*CLI> sip show channels
Peer             User/ANR    Call ID          Format           Hold    
Last Message   
91.128.104.50    (None)      302e3db1464e650  0x0 (nothing)    No      
Rx: OPTIONS               
91.128.104.50    test_user   9e2ec18f1622d61  0x8 (alaw)       No      
Rx: BYE                   
2 active SIP dialogs
voip-1*CLI> 

voip-1*CLI> core show channels
Channel              Location             State   Application(Data)       
     
SIP/test_user-b7     01229922640 at from_sip Down    (None)                  
     
1 active channel
0 active calls
0 calls processed
voip-1*CLI> 

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

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-09-24 09:04 lmadsen        Status                   new => ready for review
======================================================================