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

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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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:              2009-04-20 16:00 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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 (0103492) mmichelson (administrator) - 2009-04-20 16:00
 http://bugs.digium.com/view.php?id=13865#c103492 
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Sorry I took so long to review your patch. I took a look and saw one thing
that seemed wrong and another that seemed a little bit...off.

In ast_sip_ouraddrfor, you sometimes reference p->ourip. Instead, you
should be using the "us" variable. This way, things are more consistent.

Also, in sip_alloc, you set p->socket.type to the value of
req->socket.type. I am having difficulty finding where req->socket.type is
set when receiving an incoming SIP request. Where and when is
req->socket.type set? 

Issue History 
Date Modified    Username       Field                    Change               
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2009-04-20 16:00 mmichelson     Note Added: 0103492                          
2009-04-20 16:00 mmichelson     Status                   ready for testing =>
feedback
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