[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:                     ready for testing
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:              2009-04-15 22:38 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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 (0103292) vrban (reporter) - 2009-04-15 22:38
 http://bugs.digium.com/view.php?id=13865#c103292 
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well, this was bit more complicated as expected:

1.
ast_sip_ouraddrfor is called by sip_alloc if the call come,
and p->socket.type was static set to SIP_TRANSPORT_UDP in sip_alloc,
so i forwarded also struct sip_request from tcp_helper_thread that we can
set p->socket.type dynamic. Only then is was possibly to get the
transport
type in ast_sip_ouraddrfor to know which ip and port we need to set.

2.
We needed to watch in ast_sip_ouraddrfor also if the default ip is 
0.0.0.0, then we set only the port for the specific transport.
and the ip we internip 

3.
In build_contact we still need four differnt ast_string_field_build
to set the ;transport= the snome phone need it to send us TCP or TLS
otherwise it always fall back to UDP

4.
Can i use this patch as candidature to get a traineeship at digium? 

Issue History 
Date Modified    Username       Field                    Change               
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2009-04-15 22:38 vrban          Note Added: 0103292                          
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