[asterisk-bugs] [Asterisk 0013438]: Call disconnection with branch in via header changing and scheduled destruction of sip call

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13438 
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Reported By:                agupta
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13438
Category:                   Channels/chan_sip/General
Reproducibility:            random
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.21.2 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-09-07 09:42 CDT
Last Modified:              2008-09-10 10:39 CDT
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Summary:                    Call disconnection with branch in via header
changing and scheduled destruction of sip call
Description: 
Randomly the call gets disconnected . The calls are coming on FXO card and
are routed to the IP Phone . The scenario is as --

1. INVITE is originated from asterisk to the IP Phone .

2. IP Phone sends , Trying , Ringing and 200OK

3. Asterisk responds with ACK and the branch in via header is different
from the INVITE/200OK

4. After 32 seconds asterisk puts the call for scheduled destruction and
the call is disconnected . 
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 (0092308) putnopvut (administrator) - 2008-09-10 10:39
 http://bugs.digium.com/view.php?id=13438#c92308 
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All right. Closing. If it turns out that there is some other reason that
the communication between Asterisk and your phone is broken, please feel
free to open a new bug report. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-09-10 10:39 putnopvut      Note Added: 0092308                          
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