[asterisk-bugs] [Asterisk 0012957]: Call rejected with 403 when sending a call between two SIP gateways

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12957 
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Reported By:                asenci
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12957
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.21 
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:             2008-06-30 12:46 CDT
Last Modified:              2008-11-20 17:15 CST
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Summary:                    Call rejected with 403 when sending a call between
two SIP gateways
Description: 
I have two Asterisk servers, one acting as SIP/TDM gateway and another as
IVR server.

The issue happens when a SIP/TDM client calls the IVR and the call is
bridged to another SIP/TDM client in the first server.

Call flow:
Client 1 dial to the IVR
Call enters Server A
Server A sends call to Server B
Server B presents the IVR to Client 1
Client chooses Client 2 in the IVR
Server B bridges the call back to the Server A (Client 2)
Call rejected with 403


Setting "fromuser=" in the "Server A" peer at Server B allows the call to
be bridged back to Server A, but scramble Caller ID.

Tested with Asterisk versions 1.4.18, 1.4.19, 1.4.19 and 1.4.21, all with
the same results.
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 (0095236) blitzrage (administrator) - 2008-11-20 17:15
 http://bugs.digium.com/view.php?id=12957#c95236 
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Sorry putnopvut, this seems like something that MAY be simple, and I closed
a bug for you, so you get this one in its place ;) 

Issue History 
Date Modified    Username       Field                    Change               
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2008-11-20 17:15 blitzrage      Note Added: 0095236                          
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