Hi,
We have configured our asterisk to work with an Avaya PBX, everything is working perfect, we can call from the SIP Phones to the Avaya Extensions, and viceversa.
The problem is that sometimes, we want to take a line from the asterisk dialing 82 in our Avaya extenions, and sometimes it gave tone and sometimes it doesnt take a line to be able to dial to any SIP extension, we have 32 IP Licences in the Avaya PBX, i dont know if the IP Trunks remain busy for an error or I donw know.
Is the OpenH323 a stable framework for a production enviroment or we should implement a E1 interface between the asterisk and avaya.
We have this configuration in the ooh323.conf
[general]
bindaddr=0.0.0.0
h323id=ObjSysAsterisk
e164=100
callerid=asterisk
gatekeeper = DISABLE
context=avaya
disallow=all
allow=g729
allow=ulaw
allow=gsm
dtmfmode=rfc2833
faststart=yes
h245tunneling=yes
progress_setup = 8
progress_alert = 8
[avayacn]
type=peer
ip=172.21.1.60
port=1720
dtmfmode=inband
context=avaya
disallow=all
allow=ulaw
e164=16
The log h323_log has these messages
---------Date 05/30/08---------
00:15:38:494 No UserUser IE found in ooDecodeUUIE
00:15:38:494 Error:Failed to decode received H.2250 message. (incoming, ooh323c
And the messages log has these messages
May 11 20:06:39 WARNING[4882] src/chan_h323.c: Don’t know how to indicate condition -1 on ooh323c_8
Asterisk version is Asterisk 1.2.22