Hi All,

I succesfully connected Asterisk both to a bunch of Cisco ATA-186 through SIP and to an Avaya Prologix through H.323 trunks.

When a call is relayed by Avaya (an incoming call from an ISDN PRI that should go to Asterisk’s users) I got an “EndedByTransportFail” from Asterisk’ side. When, however, the call is originated on Avaya everything is fine, no matter is comes from a hardphone, softphone or analog line.

This Avaya also relays to many other H.323 devices, without showing this issue.

This is what I see on Asterisk when relay fail:

[4]WrapH323EndPoint::CreateConnection: Creating a H323Connection [24062]
[2]WrapH323EndPoint::CreateConnection: Incoming call
[2]WrapH323EndPoint::CreateConnection: Incoming connection with no redirecting number in SETUP.
[2]WrapH323Connection::WrapH323Connection: Creation of WrapH323Connection based on user data.
[2]WrapH323Connection::WrapH323Connection: Call is incoming.
[4]WrapH323Connection::WrapH323Connection: WrapH323Connection created.
[2]WrapH323Connection::OnReceivedSignalSetup: Received SETUP message...
[2]WrapH323EndPoint::ClearCall: Request to clear call [ip$]
[2]WrapH323Connection::OnSendReleaseComplete: Sending RELEASE COMPLETE message [ip$]
[2]WrapH323EndPoint::OnConnectionCleared: Connection [ip$] closed.
Nov 16 08:37:47 WARNING[8615]: chan_oh323.c:3554 cleanup_h323_connection: Call 'ip$' not found (clear).
[2]WrapH323EndPoint::OnConnectionCleared: Call with "" ended abnormally
[4]WrapH323Connection::WrapH323Connection: WrapH323Connection deleted.

I am running:

Asterisk: version from CVS (checked out 2005-11-14)
OH323: version 0.7.3

Does anybody already faced this beast before?

Not sure exactly what your issue is. I use OOH323 with Asterisk v1.2rc2 and an Avaya Prologix via H323 trunks without any problems in both directions. I did have some issues initially if I tried to have Asterisk offer more codecs than the ones enabled on the Avaya side, once I narrowed it down to G711/alaw in the ooh323.conf file all worked fine.

It does’nt seems to be codec related, since I am using only one (ulaw) on either side.

The creator of OH323, Jer Jer, disavows it. I would recommend giving OOH323 a try…

Okay, I did try OOH323 (and that H323 in “channels/h323” too). These are my findings:

(BTW, my topology is: A Definity Prologix talking to Asterisk through H.323 and to Ericsson MD-110 through ISDN PRI)

Extensions on Definity can always call Asterisk and extensions on Asterisk can always call Definity, no matter I’m using H323, OH323 or OOH323.

Using either OH323 or H323: Asterisk can always call Ericsson but Ericsson can never call Asterisk.

Using OOH323: Ericsson can always call Asterisk but Asterisk can never call Ericsson.

I did some debug with H323 (the last one I tryed) and found that OpenH323 is droping those calls right in transports.cxx (line 1189 - version 1.17.1) saying that “Signal channel stopped on first PDU”.

Any chance I become lucky now?

This is a workaround I used http://forums.digium.com/viewtopic.php?t=2619