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Summary:ASTERISK-19416: H323 trunking failure.
Reporter:Dimos (dtrich0)Labels:
Date Opened:2012-02-21 08:34:02.000-0600Date Closed:2013-12-18 12:42:01.000-0600
Priority:MajorRegression?
Status:Closed/CompleteComponents:Addons/chan_ooh323
Versions:1.8.7.2 Frequency of
Occurrence		Constant
Related
Issues:
Environment:Elastix 2.2.0 Asterisk 1.8.7.0 CentOS:Linux(x86_64)-2.6.18-238.12.1.el5Attachments:( 0) h323_log
( 1) h323_log
( 2) h323_log
( 3) h323_log_with_v1.6
( 4) issue_19416_full_log
( 5) issue_19416_full_log_with_V1.6
( 6) issue_19416_full_log1
( 7) ooh323.conf
Description:I have installed Asterisk 1.8.7.0 and everything is working perfect except the h323 trunk.
To be more specific. I have installed the ooh323 plug-in successfully(no error when asterisk starts) but when I call a number with outgoing route to the h323 trunk the call fails(please see the attached file).

In the same time, I observed the ethernet port with tcpdump and no packet was sent from the Asterisk to 192.168.200.202.
In Addition the h323_log is empty


Finally I check the configuration in Asterisk version 1.6.2.13 and works.
Comments:By: Matt Jordan (mjordan) 2012-02-21 16:20:50.376-0600

We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information


By: Dimos (dtrich0) 2012-02-22 03:14:42.763-0600

The full log file.
By: Dimos (dtrich0) 2012-02-22 03:54:37.098-0600

Thanks for the help. I have attached the requested log.
By: Alexander Anikin (may213) 2012-02-22 15:40:15.121-0600

Dimos,

could you set tracelevel=6 in ooh3232.conf and attach /var/log/asterisk/ooh323_log here?
By: Dimos (dtrich0) 2012-02-23 04:08:28.356-0600

Hello again,
The h323_log file after tracelevel=6 addition to 00h323.conf, amportal restart and make the same call as in issue_19416_full_log call(the h323_log have not change after the call).
Thanks again for your help.
By: Dimos (dtrich0) 2012-02-23 04:24:29.230-0600

Please see the comments in the h323_log attached file.
Thanks for the help.
By: Alexander Anikin (may213) 2012-02-23 08:57:47.608-0600

Dimos,

ooh323 channel said:

[Feb 22 09:46:03] ERROR[5404] chan_ooh323.c: Call to undefined peer 192.168.200.252:1720

This is because ooh323 driver can't now dial directly to an IP address but only to defined peers or to gatekepeer if is used.

You can solve this by definition peer for 192.168.200.252 in ooh323.conf.

If you wish to have direct ip dialing in the ooh323 channel driver then please describe reasons for it.

I suggest that direct ip dialing may contain many troubles, some secutiry holes at least.
By: Dimos (dtrich0) 2012-02-23 10:53:58.293-0600

My ooh323 configuration file.
By: Dimos (dtrich0) 2012-02-23 11:17:25.493-0600

There is not reason for direct ip dialing.
Could you please correct my ooh323.conf file in order peer for 192.168.200.252 to be defined.
Thanks again for the help
By: Alexander Anikin (may213) 2012-02-23 17:58:43.228-0600

Looks like to your ooh323.conf is correct.
Please change dial string from OOH323/192.168.200.252 to OOH323/LOC because peer name is LOC not 192.168.200.252

I hope it will work ok.

By: Dimos (dtrich0) 2012-02-24 03:12:28.695-0600

YOU ARE CORRECT.
Now the calls from the local peer to remote peer is ok but when the remote peer is call me local phone ringing and when I pick up the call I can hear the remote user but remote user is still hearing the calling tone(looks like the local user(I) not pickup the phone)
Please see the new attached log file.
Thanks again for your help support :-)
By: Dimos (dtrich0) 2012-02-24 15:11:31.074-0600

The second call is ok
The first call (from LOC to 111)has problem
By: Dimos (dtrich0) 2012-02-24 15:13:42.004-0600

please reopen it :-)
Sorry for the mess :-(
By: Alexander Anikin (may213) 2012-02-25 10:28:56.408-0600

Dimos,

I see your new log, will analyze it.
By: Dimos (dtrich0) 2012-02-28 05:46:57.640-0600

I also attach the h323_log file with first call established from local Asterisk to remote PBX successfully and the second call from remote pbx to local Asterisk with sound only in the way from remote to local(remote user hearing the calling tone and local user hearing the remote user).
By: Alexander Anikin (may213) 2012-02-28 15:43:30.019-0600

Dimos, could you explain what device make incoming call to asterisk?

This device require alaw rtp channel with dynamic rtp payload (code 96). I think that trouble is there, calling device require dynamic but asterisk send alaw rtp stream with fixed payload.

You can try to disable dynamic payload per G.711 on that device if it is possible.

By: Dimos (dtrich0) 2012-03-01 09:46:19.542-0600

Dear Alexander,
The remote pbx is a Alcatel OmniPCX Office Medium. There is not a option to disable dynamic payload there.
Also in the last h323_log until the time 12:08:34:995 is a successful call originated by Asterisk(Asterisk user has called Alcatel user) and as you can see in time 12:04:49:925 the Alcatel reports for dynamicRTPPayloadType={96} without problem in call establishment and voice transmission and reception.
The problem start when the Alcatel pbx' user is the originator of the call. In this case the Asterisk user's phone is ringing but when he pickup the phone call the Alcatel user is still hearing the calling tone until the Asterik user hang up the phone. From my point of view is signaling problem. Maybe Asterisk don't send pickup information to Alcatel pbx.  
Don't forget that the same configuration file is working with Asterisk version 1.6. My problems has started when I migrate to version 1.8. I thing something change in version 1.8 and makes the problem.
For another time I want to thanks you for your help :-)

PS: I set-up a test server with Asterisk version 1.8.9.2 with only a H323 trunk and a sip phone just for testing, and the problem remains.
By: Alexander Anikin (may213) 2012-03-01 13:03:23.701-0600

Dimos,

It's could be you're right about signalling problem. I will do further researching.
OOH323 codes for 1.8 is based on OOH323 1.6 but logically another so it's possible that
worked 1.6 config doesn't work for 1.8


By: Dimos (dtrich0) 2012-03-02 11:00:58.132-0600

Hi there,
I also upload the logs for a successful call originated by Alcatel user, using Asterisk vesion 1.6.
I hope to help :-)

By: Alexander Anikin (may213) 2012-03-05 09:29:34.560-0600

Dimos,

pls try to change alaw codec setting in LOC section in ooh323.conf, your pbx call asterisk with 60ms framing but asterisk send 20ms.

Change allow=alaw to allow=alaw:60 in LOC section and try call from OmniPCX to asterisk with logging enabled. If trouble will found again please attach log here.
By: Dimos (dtrich0) 2012-03-06 03:21:21.983-0600

Works!!! but why in version 1.6 the Asterisk don't need this parameter in ooh323.conf file?
Many many thanks for the help/support
By: Alexander Anikin (may213) 2012-03-13 17:30:41.295-0500

Dimos,

OOH323 1.6 isn't accurate about codec framing.
It can lower or grow framing depending of opposite side.
Current OOH323 can accept logical channels that have lower or equal his framing for receive
and logical channels that have greater or equal framing for transmit.

But I think we can introduce option to simplify framing checking if it's not really required.

By: Dimos (dtrich0) 2012-03-14 02:47:28.399-0500

Dear Alexander,

I kindly request to include the option in OOH323 for asterisk version 1.8.

After some successful two-way calls(LOC->ChaniaLS and ChaniaLS->LOC)the state has changed again :-(
Now, when ChaniaLS(Asterisk user) call LOC(Alcatel user) ChaniaLS can hear the LOC perfectly but LOC can't hear the ChaniaLS user(no sound/silence). I believe that for some unknown reason the Alcatel pbx has changed again the framing(please see the new attached h323_log).
So, from my point of view a "adaptive" framing will help in cooperation of ooh323 channel with the non-Asterisk pbxs.
Have a nice day and thanks for the support.      
By: Alexander Anikin (may213) 2012-03-14 14:25:59.143-0500

Dimos,

Last problem isn't same as previous.
There is problem with openning logical channels, asterisk try to open LC after TCS and MSD procedure is complete but PBX reject this. Asterisk try to open logical channel due to PBX don't send fast start proposal, look like to FastStart support is off on PBX now (earlier logs show that FastStart work on PBX). Please check config of PBX about FastStart.

And another thing. It seems to your PBX act as H.323v2 endpoint and want open logical channels after call is answered only not before, but set H.323v4 in packets and asterisk try to operate with PBX as v4 endpoint.


By: Dimos (dtrich0) 2012-03-17 02:10:18.381-0500

You are correct!!!
For unknown reason the remote PBX changed "behavior"
As it is more convenient to me to make change to Asterisk I changed the FASTSTART to OFF in Asterisk. So the problem that I had in calls originated by Asterisk solved. BUT after that I had drop calls in the calls originated by remote PBX. I changed the TUNNELING to OFF and now it works!!! What can I say?
Thanks you very much for your help/support and sorry for the mess :-(

By: Alexander Anikin (may213) 2013-12-18 12:42:01.103-0600

close as it solved on config level