[asterisk-bugs] [Asterisk 0012957]: Call rejected with 403 when sending a call between two SIP gateways
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Dec 17 14:23:11 CST 2008
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: putnopvut
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Project: Asterisk
Issue ID: 12957
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
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: 2008-12-17 14:23 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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(0096571) putnopvut (administrator) - 2008-12-17 14:23
http://bugs.digium.com/view.php?id=12957#c96571
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19 days with no response makes me think that this is something no longer
affecting the reporter. I'm suspending this, but it may be re-opened if
desired.
Issue History
Date Modified Username Field Change
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2008-12-17 14:23 putnopvut Note Added: 0096571
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