Summary:ASTERISK-17225: [patch] Can't call between Tandberg MPS 200 video bridge and SIP softphone on Asterisk running chan_ooh323
Reporter:Andy Boatman (mrhanman)Labels:
Date Opened:2011-01-11 15:14:15.000-0600Date Closed:2011-04-12 10:35:48
Versions:Frequency of
Environment:Attachments:( 0) full.log
( 1) Good_call_to_bridge_from_Tandberg_Endpoint.log
( 2) h323_log
( 3) h323_log-0018542.log
( 4) issue18599.patch
( 5) issue18599-2.patch
( 6) No_Audio_Call_to_softphone_from_bridge.log
( 7) ooh323.conf
( 8) syslog_from_Tandberg_bridge.log
( 9) Tandberg_bridge_-_Asterisk_2.log
(10) Tandberg_Conference_Bridge_-_Asterisk.log
(11) Tandberg_Conference_Bridge_to_Asterisk.log
Description:When a Tandberg MPS 200 video bridge calls a SIP softphone registered to Asterisk running chan_ooh323, the call is immediately hung up after it is answered.  When calling from the softphone to an active conference on the bridge, I get a busy signal.  Asterisk is patched with patches from bugs ASTERISK-17075 and ASTERISK-17045.  Attached is the h323_log containing both cases.
Comments:By: Andy Boatman (mrhanman) 2011-01-14 14:49:06.000-0600

I just upgraded to 1.8.2, and the problem still exists.  Incidentally, the patch from bug ASTERISK-1815401 is already included with version 1.8.2.  So, the only patch I installed was the one from bug ASTERISK-1818433.

By: Alexander Anikin (may213) 2011-01-16 11:23:15.000-0600


incoming call trouble like to 0018542, please try patch from this issue.
About outgoing calls - i see many call to different numbers from asterisk to
Tandberg device and some calls have normal signalling but hangup from asterisk side. For example, call with ooh323_o_7 id from 1500 to 1002 number. Call to another numbers have different hangup cause codes - user busy or some other.

By: Andy Boatman (mrhanman) 2011-01-17 08:18:00.000-0600

After applying the patch from bug ASTERISK-1829542, incoming calls are still dropped as soon as they are answered.  I've attached a new log of the incoming call being disconnected.

As for the outgoing calls, if I call a conference on the bridge that is not in session, the call appears to work normally.  That is, the call is answered and the bridge notifies me that there is no one in that conference.  This is not really useful, because I need to be able to call a conference in session.  When I attempt that, I get a busy signal.  Calling the same conference from a Tandberg endpoint works as it should - the device is placed in the conference.

By: Andy Boatman (mrhanman) 2011-01-17 10:44:37.000-0600

I've also attached the log from the Tandberg bridge of a call from the bridge to the softphone that is answered, and then immediately disconnected.

By: Alexander Anikin (may213) 2011-01-18 16:44:15.000-0600

MrHarMan, please set h245tunelling=no and faststart=no in your config in tandberg bridge section, retest calls and attach h323 and asterisk console log for these calls and your ooh323.conf file also.

By: Andy Boatman (mrhanman) 2011-01-18 16:55:32.000-0600

OK, the changes to the config did not change anything that I could see.  Attached are the requested logs.

By: Alexander Anikin (may213) 2011-01-19 16:24:03.000-0600

Btw, 1st patch from bug 0018542 isn't applicable to the trunk ;) I was make it from experimental codes which has ipv6 support and i'm surpised if you can apply him to the trunk. I uploaded correct patch here and you can apply him to clean trunk. This patch also contain 2nd patch from bug 0018542 (this issue describe same trouble).
But previous is related to incoming issue only and i haven't idea about outgoing calls to Tandberg bridge trouble. Can you attach here syslog from bridge for working call to him or tcpdump saved file with captured signalling data of working call?

Outgoing calls from asterisk to bridge are normal from signalling point of view, but these calls received hangup with abnormal causes and i'm not understand while what we must to correct for normal calling Tandberg bridge.

By: Andy Boatman (mrhanman) 2011-01-20 09:34:51.000-0600

When I apply the patch to the the source or the SVN trunk, I get this build error:

  [CC] ooh323c/src/ooq931.c -> ooh323c/src/ooq931.o
ooh323c/src/ooq931.c: In function âooAcceptCallâ:
ooh323c/src/ooq931.c:1967: error: expected â)â before â{â token
ooh323c/src/ooq931.c:2009: error: expected expression before â}â token
ooh323c/src/ooq931.c:1809: warning: unused variable âh245IpAddrâ
make[1]: *** [ooh323c/src/ooq931.o] Error 1
make: *** [addons] Error 2

It compiles fine without the patch.

By: Alexander Anikin (may213) 2011-01-20 10:12:14.000-0600

Sorry for mistake in prevoius patch, patch is replaced, you can apply it to clean trunk or simple add one ")" at end of string number 1966 in ooq931.c.

By: Andy Boatman (mrhanman) 2011-01-20 15:39:41.000-0600

OK, the call from the bridge is answered by the softphone successfully.  However, the softphone doesn't seem to be sending any audio.  It doesn't receive any audio, either, execpt for when the call is connected the bridge announces that you have joined a conference.  I've attached a log from the bridge of the transaction, as well at the h323_log of the session.  I also attached a log from the bridge of a successful call from a Tandberg Endpoint to the bridge.

By: Alexander Anikin (may213) 2011-01-20 17:31:26.000-0600

I see that signalling is good for call from bridge to asterisk and there is not a essential difference in call signalling between tandberg endpoint and asterisk except one thing - tb endpoint setup transfer capabilities in setup and connect packet same as tb bridge (unresticted digital and 768k rate) and asterisk setup caps to speech at 64Kbps. I suggest tb bridge don't understand these caps or can't work properly with them. Probably it concern the outgoing from asterisk to tb bridge issue also.
I'll try to implement right setup of call capabilities for chan_ooh323 other than speech at 64K.

By: Andy Boatman (mrhanman) 2011-01-21 15:31:16.000-0600

Thanks, may.  Let me know what else you need me to do.

By: Alexander Anikin (may213) 2011-02-10 07:52:53.000-0600


As issue 0018542 is closed now, please try asterisk patched by
patch from this issue. I think that part of this issue was solved by
this patch

By: Andy Boatman (mrhanman) 2011-02-11 09:49:35.000-0600

I recompiled Asterisk and applied the patch from bug ASTERISK-1829542.  The behavior remains the same.  I'll attach the h323_log.

By: Alexander Anikin (may213) 2011-02-26 17:40:57.000-0600


Please test calls with current 1.8 branch from svn and if it will be failed try test with patch number 2 uploaded in this issue

By: Leif Madsen (lmadsen) 2011-04-12 08:48:34

No response from the reporter. Close this issue?

By: Alexander Anikin (may213) 2011-04-12 10:35:48

Closed due to lack of response.
Feel free to reopen if necessary.