[asterisk-bugs] [Asterisk 0016674]: [patch] channels stuck in ringing state forever

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16674 
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Reported By:                under
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16674
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     confirmed
Asterisk Version:           Older 1.4 - please test a newer version 
JIRA:                       SWP-781 
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:             2010-01-22 11:13 CST
Last Modified:              2010-02-26 10:44 CST
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Summary:                    [patch] channels stuck in ringing state forever
Description: 
Scenario:
CALLER -> INVITE            ->  ASTERISK -> INVITE            -> CALLEE
CALLER <- SESSION PROGRESS  <-  ASTERISK <- SESSION PROGRESS  -> CALLEE
CALLER <- RINGING           <-  ASTERISK <- RINGING           -> CALLEE

After this short network out of order causes further packets lost and
channels are stuck in "show channels" forever.

Caller and callee gateways seem to hangup channel by retransmission
timeout, so they are clean and tidy.

You can reproduce this with attached sipp scenario being run on callee
side.
It simply doesn't response anything after Ringing - you'll see channels
stuck.
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 (0118598) dvossel (administrator) - 2010-02-26 10:44
 https://issues.asterisk.org/view.php?id=16674#c118598 
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In 1.6.0, does the timerb option do what you are wanting?

;
;--------------------------- SIP timers
----------------------------------------------------
; These timers are used primarily in INVITE transactions. 
; The default for Timer T1 is 500 ms or the measured run-trip time
between
; Asterisk and the device if you have qualify=yes for the device.
;
;t1min=100                      ; Minimum roundtrip time for messages to
monitored hosts
                                ; Defaults to 100 ms
;timert1=500                    ; Default T1 timer
                                ; Defaults to 500 ms or the measured
round-trip
                                ; time to a peer (qualify=yes).
;timerb=32000                   ; Call setup timer. If a provisional
response is not received
                                ; in this amount of time, the call will
autocongest
                                ; Defaults to 64*timert1 

Issue History 
Date Modified    Username       Field                    Change               
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2010-02-26 10:44 dvossel        Note Added: 0118598                          
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