[asterisk-bugs] [Asterisk 0009516]: SIP_HEADER function after re-invite doesn't report headers in initial INVITE
noreply at bugs.digium.com
noreply at bugs.digium.com
Mon Feb 11 09:31:02 CST 2008
The following issue has been RESOLVED.
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http://bugs.digium.com/view.php?id=9516
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Reported By: Marquis
Assigned To: file
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Project: Asterisk
Issue ID: 9516
Category: Core/General
Reproducibility: always
Severity: feature
Priority: normal
Status: resolved
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:
Resolution: suspended
Fixed in Version:
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Date Submitted: 04-10-2007 17:39 CDT
Last Modified: 02-11-2008 09:31 CST
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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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file - 02-11-08 09:31
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I'll suspend this for now since it isn't actively being looked at... if you
want to pursue it oej, I'll leave it alone.
Issue History
Date Modified Username Field Change
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02-11-08 09:31 file Status confirmed => resolved
02-11-08 09:31 file Resolution open => suspended
02-11-08 09:31 file Assigned To => file
02-11-08 09:31 file Note Added: 0082008
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