Summary:ASTERISK-07776: cancellation does not stop ooh323 dialing an empty/no route device
Reporter:Di-Shi Sun (homesick)Labels:
Date Opened:2006-09-19 16:05:52Date Closed:2011-06-07 14:02:40
Versions:Frequency of
Environment:Attachments:( 0) asterisk.log
( 1) config.log
( 2) h323_log
( 3) src.log
( 4) test4.doc
( 5) test4.txt
Description:In cancellation test case, if the destination is an empty IP/no route device, when source gateway cancels the call, ooh323 does not stop the dial application. The dial will be timeout eventually.


asterisk-addons svn 295

I attached the test log including test bed configuration, ooh323.conf, extensions.conf, cli log, and h323_log in one doc file.
Comments:By: jmls (jmls) 2006-11-01 12:43:44.000-0600

nudge nudge

By: Objective Systems (objsys) 2006-11-06 10:37:34.000-0600

I don't know what you mean by empty route.

The non-marked or unavaliable extensions can be handled by extension.conf, Is it related to this?

Avin Patel

By: Di-Shi Sun (homesick) 2006-11-06 10:48:58.000-0600

an empty device means an IP address in the same network segment as Asterisk, but nobody use this IP addres.

no route device means an IP address not in the same network segment as Asterisk, no route to this IP address, such as

If the destination is an IP address without a real device or no route to this IP address, ooh323 does not work correctly.


By: Objective Systems (objsys) 2006-11-06 11:03:53.000-0600

Please provide your extension.conf & ooh323.conf, displaying this problem.

By: Di-Shi Sun (homesick) 2006-11-06 11:32:47.000-0600

extensions.conf and ooh323.conf had been posted in test4.doc and test4.txt.

By: Objective Systems (objsys) 2007-01-23 13:20:57.000-0600

I don't see log getting message to cancel the call from source.

How you are canceling the call at source?
case 1:
By properly hanging up at source, than call from asterisk will be terminated without timeout.
If this is the case, than it should be a problem.

case 2:
If source is crashed/closed (never sends Release Complete message to asterisk, to hangup call), than asterisk will close the call as timeout occurs.
This is correct behavior.

Avin Patel

By: Di-Shi Sun (homesick) 2007-01-24 18:00:40.000-0600

It is case 1. We hung up the call from the source.

By: Objective Systems (objsys) 2007-01-25 13:02:38.000-0600

In this case, source must be sending the release complete message. I have to check the messages exchanged.

1. Stop asterisk server.

2. Please rebuild the package for debugging with following steps:
cd aterisk-addons-1.2/asterisk-ooh323c
make clean
make debug
make install

3. Now run the asterisk server, passing -vvv option will be good.

4. Now run the problem scenario.
  Make a call from source and asterisk doesn't hangup

5. Provide the var/log/asterisk/h323_log file.
  If you able to provide the CLI messages, than also great.
  in addition, If you can provide wireshark capture data, also greate.

Avin Patel

By: Di-Shi Sun (homesick) 2007-05-25 12:26:29

Hi Avin Patel,

I re-run the test for asterisk svn 65312 and addons svn 384. The results are same. I attached the configure in config.log (including ooh323.conf and extensions.conf), the source cisco gateway log (src.log), asterisk cli (asterisk.log) and h323_log for this test.


Di-Shi Sun.

By: Russell Bryant (russell) 2008-01-16 12:05:23.000-0600

This module has been unsupported for a long time.  Now, it is officially marked as unsupported.  So, only bug reports with patches will be accepted at this time.