[asterisk-bugs] [Asterisk 0011843]: Moved Temporarily Contact Transport information not used in next invite

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=11843 
====================================================================== 
Reported By:                pestermann
Assigned To:                bbryant
====================================================================== 
Project:                    Asterisk
Issue ID:                   11843
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.6.0-beta9 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             01-25-2008 02:34 CST
Last Modified:              07-23-2008 12:50 CDT
====================================================================== 
Summary:                    Moved Temporarily Contact Transport information not
used in next invite
Description: 
When getting back an Moved Temporarily from the called party the transport
information in the contact header is not used for the next invite based on
promiscredir=yes. 

In the SIP debug 
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
has duplicate       0012026 Asterisk 1.6-beta3 does not follow sip ...
has duplicate       0012550 [SIP/TCP] received 302 Moved Temporaril...
====================================================================== 

---------------------------------------------------------------------- 
 pabelanger - 07-23-08 12:50  
---------------------------------------------------------------------- 
bbryant: I think that is our issue.  Our SIP PEER is only capable of using
SIP/TCP but asterisk is sending the invite via UDP. In the past the only
setting we need to enable was transport=tcp under the SIP PEER.  Is there
something else that needs to be set in the latest 1.6.0 branch?

Here is how we dial the peer.

extensions.conf
---
[from-zap]
exten => s,1,Dial(SIP/sv0071ivr,5) 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
07-23-08 12:50  pabelanger     Note Added: 0090619                          
======================================================================