[asterisk-bugs] [Asterisk 0014055]: "outboundproxy" in "general" section of sip.conf doesn't work

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14055 
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Reported By:                chris-mac
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14055
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.1 
SVN Revision (number only!): 162896 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-12-10 16:36 CST
Last Modified:              2008-12-11 07:00 CST
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Summary:                    "outboundproxy" in "general" section of sip.conf
doesn't work
Description: 
I am having problems with "outboundproxy" in "general" section of sip.conf
file.

when I put SIP proxy address in [general] sip.conf:
[general]
...
outboundproxy=proxyAddress:5060

and try to Dial(SIP/enum-test at sip.nemox.net), I am getting the following
error in the console:

    -- Executing [43780004711 at dialSIP:2]
Dial("SIP/dev-sip.tele500.com-08204da0", "SIP/enum-test at sip.nemox.net") in
new stack
  == Using SIP RTP CoS mark 5
    -- Got SIP response 482 "Loop Detected" back from 0.0.0.0
    -- Called enum-test at sip.nemox.net
    -- Now forwarding SIP/dev-sip.tele500.com-08204da0 to
'Local/enum-test at common' (thanks to SIP/sip.nemox.net-08210e58)

Debug trace attached.
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---------------------------------------------------------------------- 
 (0096183) chris-mac (reporter) - 2008-12-11 07:00
 http://bugs.digium.com/view.php?id=14055#c96183 
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I just checked and the same problem exists in 1.6.0.3-rc.

How to reproduce it?

Install Asterisk 1.6.0.3-rc with default config files, and use
sip.conf-1.6.0.3.txt (attached) as sip.conf and extensions.conf.1.6.0.3.txt
(attached) as extensions.conf.

Dial into Asterisk with any SIP phone. There will be an error in console
as shown in console-1.6.0.3.txt (attached) and Asterisk will try to send
INVITE to itself - as can be seen in sip-trace-1.6.0.3.txt. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-11 07:00 chris-mac      Note Added: 0096183                          
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