[asterisk-bugs] [Asterisk 0013865]: [patch] 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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https://issues.asterisk.org/view.php?id=13865 
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Reported By:                st
Assigned To:                dvossel
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Project:                    Asterisk
Issue ID:                   13865
Category:                   Channels/chan_sip/TCP-TLS
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
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:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2008-11-09 10:03 CST
Last Modified:              2009-06-16 12:11 CDT
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Summary:                    [patch] 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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 (0106478) svnbot (reporter) - 2009-06-16 12:11
 https://issues.asterisk.org/view.php?id=13865#c106478 
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Repository: asterisk
Revision: 200992

_U  branches/1.6.0/
U   branches/1.6.0/channels/chan_sip.c

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r200992 | dvossel | 2009-06-16 12:11:52 -0500 (Tue, 16 Jun 2009) | 39
lines

Merged revisions 200946 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

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  r200946 | dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32
lines
  
  SIP transport type issues
  
  What this patch addresses:
  1. ast_sip_ouraddrfor() by default binds to the UDP address/port
  reguardless if the sip->pvt is of type UDP or not.  Now when no
  remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's
  transport type, attempting to set the address and port to the
  correct TCP/TLS bindings if necessary.
  2.  It is not necessary to send the port number in the Contact
  header unless the port is non-standard for the transport type.
  This patch fixes this and removes the todo note.
  3.  In sip_alloc(), the default dialog built always uses transport
  type UDP.  Now sip_alloc() looks at the sip_request (if present)
  and determines what transport type to use by default.
  4.  When changing the transport type of a sip_socket, the file
  descriptor must be set to -1 and in some cases the tcptls_session's
  ref count must be decremented and set to NULL.  I've encountered
  several issues associated with this process and have created a function,
  set_socket_transport(), to handle the setting of the socket type.
  
  
  (closes issue https://issues.asterisk.org/view.php?id=13865)
  Reported by: st
  Patches:
        dont_add_port_if_tls.patch uploaded by Kristijan (license 753)
        13865.patch uploaded by mmichelson (license 60)
        tls_port_v5.patch uploaded by vrban (license 756)
        transport_issues.diff uploaded by dvossel (license 671)
  Tested by: mmichelson, Kristijan, vrban, jmacz, dvossel
  
  Review: https://reviewboard.asterisk.org/r/278/
........

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http://svn.digium.com/view/asterisk?view=rev&revision=200992 

Issue History 
Date Modified    Username       Field                    Change               
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2009-06-16 12:11 svnbot         Checkin                                      
2009-06-16 12:11 svnbot         Note Added: 0106478                          
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