[asterisk-bugs] [Asterisk 0012957]: Call rejected with 403 when sending a call between two SIP gateways
noreply at bugs.digium.com
noreply at bugs.digium.com
Tue Jul 1 08:14:55 CDT 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:
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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: 06-30-2008 12:46 CDT
Last Modified: 07-01-2008 08:14 CDT
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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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asenci - 07-01-08 08:14
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Forgot to send the details:
Using Zap channel 104 on server Magnesium (phone at my desk) to call
22229696 (IVR) at server Fluorine.
Choosen option 2 at the IVR.
Server Fluorine calls 22229670 at server Magnesium.
Server Magnesium declines with 403.
Tests done without "fromuser=" set.
Issue History
Date Modified Username Field Change
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07-01-08 08:14 asenci Note Added: 0089483
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