[asterisk-bugs] [Asterisk 0014188]: [patch] add option to configure the SIP type in users.conf
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Jan 23 11:17:15 CST 2009
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=14188
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Reported By: klaus3000
Assigned To:
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Project: Asterisk
Issue ID: 14188
Category: Channels/chan_sip/General
Reproducibility: always
Severity: feature
Priority: normal
Status: new
Asterisk Version: SVN
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-01-07 09:34 CST
Last Modified: 2009-01-23 11:17 CST
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Summary: [patch] add option to configure the SIP type in
users.conf
Description:
Currently, users provisioned in users.conf are added to chan_sip as
"friend". The attached patch allows to specify the the type in users.conf
using the directive siptype=...
Thus it does not influence with other channel types (in chan_h323 and
chan_iax)
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(0098585) klaus3000 (reporter) - 2009-01-23 11:17
http://bugs.digium.com/view.php?id=14188#c98585
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Nevertheless I think this is a useful addition, for example the simple
scenario:
users.conf:
user1: +43111111
user2: +43222222
user 1 calls a mobile phone. The mobile phone forwards to user 2.
user1 user2 Asterisk GW
-------INVITE----->
Asterisk challenges
the SIP friend
----INVITE--->
ISDN......forwarding to +4322222
<----INVITE---
This INVITE has a From: +431111111
although the GW is defined as peer, Asterisk
challenges the GW because it thinks
that the call is coming from user1 as the
"user" of user1 is stronger than the "peer" of the
gateway.
Thus, defining the users only as "peer" instead of "friend" is really
useful (problem exists in 1.4, not sure about trunk)
Issue History
Date Modified Username Field Change
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2009-01-23 11:17 klaus3000 Note Added: 0098585
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