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

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Apr 15 14:23:04 CDT 2009


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:              2009-04-15 14:23 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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---------------------------------------------------------------------- 
 (0103278) mmichelson (administrator) - 2009-04-15 14:23
 http://bugs.digium.com/view.php?id=13865#c103278 
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ourip is typically set using the function ast_sip_ouraddrfor. Looking at
the code, there is a lot of logic to figure out whether to use an internal
or external IP address when determining what to place in SIP packets.

With basic setups, such as mine, the outcome is that ourip is set to the
bindaddr (which is the udpbindaddr). A possible remedy for this would be to
pass the transport to the sip_ouraddrfor function so that we can determine
whether to use the udp, tcp, or tls bindaddr. I will put together a patch
that implements this change.

In the end, I think both the change I have described here and the one you
posted as a patch need to be implemented. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-04-15 14:23 mmichelson     Note Added: 0103278                          
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