[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. 
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http://bugs.digium.com/view.php?id=11843 
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Reported By:                pestermann
Assigned To:                
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Project:                    Asterisk
Issue ID:                   11843
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.6.0-beta1 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             01-25-2008 02:34 CST
Last Modified:              02-23-2008 07:22 CST
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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 
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Relationships       ID      Summary
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has duplicate       0012026 Asterisk 1.6-beta3 does not follow sip ...
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---------------------------------------------------------------------- 
 rjain - 02-23-08 07:22  
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The issue w/ handling 302 in the dial-plan is that not every call uses
dial-plan. I had reported the same issue a while back where I was using a
call file to originate a SIP call and Asterisk couldn't handle 302 in that
case. 

SIP redirection mechanism is something specific to SIP signalling. It
might make sense to keep higher layers (such as dial-plan, call files) be
agnostic of this mechanism and let chan_sip handle 3XX responses by itself. 

Issue History 
Date Modified   Username       Field                    Change               
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02-23-08 07:22  rjain          Note Added: 0082804                          
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