[asterisk-bugs] [Asterisk 0009516]: SIP_HEADER function after re-invite doesn't report headers in initial 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=9516 
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Reported By:                Marquis
Assigned To:                
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Project:                    Asterisk
Issue ID:                   9516
Category:                   Core/General
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     confirmed
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 61220 
Disclaimer on File?:        Yes 
Request Review:              
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Date Submitted:             04-10-2007 17:39 CDT
Last Modified:              09-12-2007 16:30 CDT
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Summary:                    SIP_HEADER function after re-invite doesn't report
headers in initial INVITE
Description: 
Any customer headers that are added to a SIP message are unavailable when
the line of dialplan code is from a Realtime switch.  Adding the exact same
line (while keeping the rest in Realtime) in extensions.conf makes it work
fine.

However, all "normal" headers (User-Agent, etc.) are available.
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 jmls - 09-12-07 16:30  
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if pineapple is rotting, should we close this ? 

Issue History 
Date Modified   Username       Field                    Change               
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09-12-07 16:30  jmls           Note Added: 0070434                          
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