[asterisk-bugs] [Asterisk 0012006]: [patch] chan_sip fails to set contact, via, and sdp headers correctly with outboundproxy set

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12006 
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Reported By:                mnicholson
Assigned To:                oej
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Project:                    Asterisk
Issue ID:                   12006
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 103725 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-02-15 12:45 CST
Last Modified:              2008-12-10 17:13 CST
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Summary:                    [patch] chan_sip fails to set contact, via, and sdp
headers correctly with outboundproxy set
Description: 
Setting outboundproxy in sip.conf causes asterisk to fail to properly set
contact, via, and sdp headers (they get set to 127.0.0.1).  This is because
when outboundproxy is set, we don't fill in the sa struct of the sip_pvt
for that dialog which is used to generate the ourip member of the sip_pvt
struct which is used to generate the contact, via, and sdp headers that are
faulty.  I don't know what the reasoning was for not populating the
sip_pvt.sa member originally, but it breaks stuff.
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 (0096161) chris-mac (reporter) - 2008-12-10 17:13
 http://bugs.digium.com/view.php?id=12006#c96161 
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Hmmm, I am afraid no packets are hitting my Proxy. There are no aliases
defined, and the _ONLY_ address it listens on is 78.105.1.128

Restarting OpenSER - PID 2772

Listening on 
             udp: 78.105.1.128 [78.105.1.128]:5060
             tcp: 78.105.1.128 [78.105.1.128]:5060
Aliases: 
root at dev2:/tmp#

So looks like the loop is inside of Asterisk only.

Also I changed proxy to 193.108.191.83 (completely different server on
other network) and I am still getting the same error:

    -- Executing [43780004711 at dialSIP:2]
Dial("SIP/dev-sip.tele500.com-08204d10", "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

ngrep shows the same bounce between Asterisk IP => Asterisk IP as before. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-10 17:13 chris-mac      Note Added: 0096161                          
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