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Summary:ASTERISK-13199: "outboundproxy" in "general" section of sip.conf doesn't work
Reporter:Chris Maciejewski (chris-mac)Labels:
Date Opened:2008-12-10 16:36:42.000-0600Date Closed:2008-12-17 13:54:37.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) console-1.6.0.3.txt
( 1) debug.txt
( 2) extensions.conf.1.6.0.3.txt
( 3) sip.conf-1.6.0.3.txt
( 4) sip-show-settings.txt
( 5) sip-trace-1.6.0.3.txt
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@sip.nemox.net), I am getting the following error in the console:

   -- Executing [43780004711@dialSIP:2] Dial("SIP/dev-sip.tele500.com-08204da0", "SIP/enum-test@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@sip.nemox.net
   -- Now forwarding SIP/dev-sip.tele500.com-08204da0 to 'Local/enum-test@common' (thanks to SIP/sip.nemox.net-08210e58)

Debug trace attached.
Comments:By: Brandon Kruse (bkruse) 2008-12-10 16:42:59.000-0600

What does the cli command "sip show settings" output?

-bk

By: Chris Maciejewski (chris-mac) 2008-12-10 16:45:52.000-0600

Output of my "sip show settings" attached.

By: Brandon Kruse (bkruse) 2008-12-10 16:46:45.000-0600

Outb. proxy:            78.105.1.128

Is that correct chris?

-bk

By: Chris Maciejewski (chris-mac) 2008-12-10 16:49:33.000-0600

Yes. Asterisk listens on 78.105.1.129 and Proxy is on 78.105.1.128.

The whole thing looks like:

UAC (78.105.1.131) => PROXY (78.105.1.128) => Asterisk (78.105.1.129)

now asterisk should make outbound calls like this:

Asterisk (78.105.1.129) => Proxy (78.105.1.128) => Final Destination (xxx)

By: Brandon Kruse (bkruse) 2008-12-10 16:58:04.000-0600

Right, I see. Can you paste a sip debug?

I am not sure if I can solve this issue now :P

-bk

By: Chris Maciejewski (chris-mac) 2008-12-11 01:18:41.000-0600

bkruse: sip debug can be found here http://bugs.digium.com/file_download.php?file_id=20944&type=bug

Also please note, some more info about this (or similar) issue can be found here http://bugs.digium.com/view.php?id=12006

By: Chris Maciejewski (chris-mac) 2008-12-11 07:00:52.000-0600

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.

By: Digium Subversion (svnbot) 2008-12-17 13:52:36.000-0600

Repository: asterisk
Revision: 165216

U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r165216 | file | 2008-12-17 13:52:36 -0600 (Wed, 17 Dec 2008) | 4 lines

Call proxy_update so that the IP address gets populated. Sending stuff to 0.0.0.0 is silly!
(closes issue ASTERISK-13199)
Reported by: chris-mac

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=165216

By: Digium Subversion (svnbot) 2008-12-17 13:53:38.000-0600

Repository: asterisk
Revision: 165217

_U  branches/1.6.0/
U   branches/1.6.0/channels/chan_sip.c

------------------------------------------------------------------------
r165217 | file | 2008-12-17 13:53:38 -0600 (Wed, 17 Dec 2008) | 11 lines

Merged revisions 165216 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
 r165216 | file | 2008-12-17 15:52:40 -0400 (Wed, 17 Dec 2008) | 4 lines
 
 Call proxy_update so that the IP address gets populated. Sending stuff to 0.0.0.0 is silly!
 (closes issue ASTERISK-13199)
 Reported by: chris-mac
........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=165217

By: Digium Subversion (svnbot) 2008-12-17 13:54:36.000-0600

Repository: asterisk
Revision: 165218

_U  branches/1.6.1/
U   branches/1.6.1/channels/chan_sip.c

------------------------------------------------------------------------
r165218 | file | 2008-12-17 13:54:36 -0600 (Wed, 17 Dec 2008) | 11 lines

Merged revisions 165216 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
 r165216 | file | 2008-12-17 15:52:40 -0400 (Wed, 17 Dec 2008) | 4 lines
 
 Call proxy_update so that the IP address gets populated. Sending stuff to 0.0.0.0 is silly!
 (closes issue ASTERISK-13199)
 Reported by: chris-mac
........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=165218