[asterisk-bugs] [Asterisk 0013865]: SIP/TLS enabled - just one call possible - 481 Call/Transaction Does Not Exist

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13865 
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Reported By:                st
Assigned To:                mmichelson
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Project:                    Asterisk
Issue ID:                   13865
Category:                   Channels/chan_sip/TCP-TLS
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.1-beta1 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2008-11-09 10:03 CST
Last Modified:              2009-05-06 16:10 CDT
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Summary:                    SIP/TLS enabled - just one call possible - 481
Call/Transaction Does Not Exist
Description: 
Settup consits of Asterisk 1.6.1-beta1 and Snom320 firmware 7.3.7
TLS is enabled using standard port 5061 for the first example and port
5062 for the second example.

Dialplan is very simple: Answer; Wait(1); Playback(...); Hangup;

The first call of the first example does work but it is not possible to
call the same number again: 481 Call/Transaction Does Not Exist

The first call of the second example has no "BYE" and has to be cancelled
at the phone.


(IMHO a new category chan_sip/TLS should be created)
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Relationships       ID      Summary
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has duplicate       0015009 Asterisk's not handling BYE sip-tls mes...
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 (0104322) jmacz (reporter) - 2009-05-06 16:10
 http://bugs.digium.com/view.php?id=13865#c104322 
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Tried the patch against the same revision it was made (190902) and then
applied it to Asterisk-1.6.0.2-beta1 and repeated the test.

In both cases, Polycom phones (320 and 330) were used. Both have "Port:
5061" set on their web config page. In the Asterisk side, tried both
specifying and not specifying a port for the peer with the same results.

In both cases if the A party finishes the call, BYE message is correctly
handled by the B party; but if B party ends the call, then A still answers
with a SIP 481 message.

I'm attaching files 20090506_prueba_tls-svn190902_patch_v5 and
20090506_prueba_tls-1.6.2.0-beta1_patch_v5 with SIP debug for the calls
tested in both installations. Hope this will be of some help.

I'll try the current SVN head and Asterisk-1.6.1.0 and come back with the
results asap. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-05-06 16:10 jmacz          Note Added: 0104322                          
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