Summary:ASTERISK-00950: SIP channel calling unregistred hosts at
Reporter:Olle Johansson (oej)Labels:
Date Opened:2004-01-29 14:22:22.000-0600Date Closed:2008-01-15 14:42:38.000-0600
Versions:Frequency of
Environment:Attachments:( 0) chansipfix.txt
( 1) chansipfix.txt
Description:The CVS version of the SIP channel (chan_sip.c) have lost control of if a peer is registred or not. If you have a dynamic peer that is not registred, calling that with Dial() generates packet to IP


Trying to track this, help appreciated. COnfirmations if this happens to you also appreciated. I can repeat this error and have numerous packets going to in my server that I want to get rid of
Comments:By: Olle Johansson (oej) 2004-01-29 15:51:16.000-0600

Latest file upload fixes this on my server. Please check. Also fixed some intendation in create_addr() function.

By: dpackham (dpackham) 2004-01-29 17:32:58.000-0600

This happens to me too.   latest CVS some phones get packets to but most SIP phones work just great....

your patch killed CVS SIP so it did not work for me to test.

By: Olle Johansson (oej) 2004-01-30 01:34:07.000-0600

"killed CVS SIP" ? Please be more specific.

By: geertn (geertn) 2004-01-30 04:31:34.000-0600

I second this. With this patch sip does not work anymore, my phones do not register and if I do a:
sip reload

I get a deadlock.

By: Olle Johansson (oej) 2004-01-30 04:54:44.000-0600

Ugly, back to the drawingboards.

The original problem appears when I dial two SIP hosts
One is registred and one is not. Both are called and thet one not registred
is called at IP which is a BUG. We tried to fix this with this patch, but obviously something else happens.

By: Olle Johansson (oej) 2004-01-30 05:07:40.000-0600

SIP reload works like a charm on my system... Strange. What happens with SIP register? Can you attach a SIP DEBUG or doesn't Asterisk answer at all?

By: geertn (geertn) 2004-01-30 06:06:43.000-0600

Tried again, with a fresh cvs checkout (this is against original chan_sip in cvs isn't it?).


Some CLI info:
Asterisk Ready.
*CLI> sip debug
SIP Debugging Enabled
*CLI> show version
Asterisk CVS-01/30/04-12:47:42 built by root@audix on a i686 running Linux
*CLI> sip reload


If I start asterisk withouth asterisk -vvvc I can do asterisk -vvvr but even then a deadlock. After that:
Connected to Asterisk CVS-01/30/04-12:47:42 currently running on audix (pid = 11415)
   -- Remote UNIX connection
audix*CLI> sip reload
Previous SIP reload not yet done

Hope this helps.

edited on: 01-30-04 06:07

By: Olle Johansson (oej) 2004-01-31 17:04:03.000-0600

Hmm, maybe we should also check the Qualification status - "UNREACHABLE" or "LAGGED"...

By: Olle Johansson (oej) 2004-01-31 17:33:49.000-0600

This patch didn't work either. I need help on where and how to cancel a SIP call smoothly.

By: Olle Johansson (oej) 2004-02-01 04:12:00.000-0600

A new approach that works on my system without breaking the SIP channel :-)

I'm checking for IP address in sip_new, which seems to be a good place. Please confirm if this works on your system or not.

By: Brian West (bkw918) 2004-02-01 18:33:58.000-0600

Fixed in CVS

By: Digium Subversion (svnbot) 2008-01-15 14:42:38.000-0600

Repository: asterisk
Revision: 2105

U   trunk/channels/chan_sip.c

r2105 | markster | 2008-01-15 14:42:37 -0600 (Tue, 15 Jan 2008) | 2 lines

If unregistered, don't consider it valid (bug ASTERISK-950)