I have the latest Asterisk setup and trunked to an Avaya CM 5.2.1 with a H.323 trunk.
I can make incoming and outgoing calls fine. I have audio from Avaya to the Asterisk, but no audio from the Asterisk (sip client) to the Avaya.
Here is my ooh323.conf config:
[general]
port=1720
bindaddr=10.1.122.7
faststart=yes
h245tunneling=no
gatekeeper=DISABLE
disallow=all
allow=ulaw
dtmfmode=inband
context=from-internal
e164=100
h323id=ObjSysAsterisk
callerid="Asterisk PBX"
progress_setup = 8
progress_alert = 8
[avaya]
type=peer
;type=friend
context=from-internal
host=10.1.105.16
port=1720
disallow=all
allow=ulaw
canreinvite=no
dtmfmode=inband
When I do a packet capture on the trixbox, it is almost like there is no rtp stream from the trixbox to the Avaya. I see from the Avaya, and to/from the sip client. It makes me think that the bridge between the sip phone to the h323 trunk is no existent or broke.
When looking at the h323log on the trixbox this is what it shows during a one-way audio call.
[trixbox1.localdomain asterisk]# tail h323_log
11:38:50:539 ooFindLogicalChannel, checking channel: 1:transmit
11:38:50:539 ooFindLogicalChannel, comparing channel: 1002:transmit
11:38:50:539 Comparing channel with codec type: 4
11:38:50:539 Comparing codecs: current=18, requested=4
11:38:50:539 ooFindLogicalChannel, checking channel: 0:receive
11:38:50:539 ooFindLogicalChannel, checking channel: 0:transmit
11:38:50:539 ERROR: Logical Channel 1004 not found, fasts start answered. (outgoing, ooh323c_o_106)
11:38:50:539 }
11:38:50:539 ERROR: unbalanced structure
11:38:50:539 Error:Invalid Connect message received. (outgoing, ooh323c_o_106)
[trixbox1.localdomain asterisk]#
I've scoured this forum and the internet. Last ditch effort was to post here. Any thoughts?
Thanks in advance.
I can make incoming and outgoing calls fine. I have audio from Avaya to the Asterisk, but no audio from the Asterisk (sip client) to the Avaya.
Here is my ooh323.conf config:
[general]
port=1720
bindaddr=10.1.122.7
faststart=yes
h245tunneling=no
gatekeeper=DISABLE
disallow=all
allow=ulaw
dtmfmode=inband
context=from-internal
e164=100
h323id=ObjSysAsterisk
callerid="Asterisk PBX"
progress_setup = 8
progress_alert = 8
[avaya]
type=peer
;type=friend
context=from-internal
host=10.1.105.16
port=1720
disallow=all
allow=ulaw
canreinvite=no
dtmfmode=inband
When I do a packet capture on the trixbox, it is almost like there is no rtp stream from the trixbox to the Avaya. I see from the Avaya, and to/from the sip client. It makes me think that the bridge between the sip phone to the h323 trunk is no existent or broke.
When looking at the h323log on the trixbox this is what it shows during a one-way audio call.
[trixbox1.localdomain asterisk]# tail h323_log
11:38:50:539 ooFindLogicalChannel, checking channel: 1:transmit
11:38:50:539 ooFindLogicalChannel, comparing channel: 1002:transmit
11:38:50:539 Comparing channel with codec type: 4
11:38:50:539 Comparing codecs: current=18, requested=4
11:38:50:539 ooFindLogicalChannel, checking channel: 0:receive
11:38:50:539 ooFindLogicalChannel, checking channel: 0:transmit
11:38:50:539 ERROR: Logical Channel 1004 not found, fasts start answered. (outgoing, ooh323c_o_106)
11:38:50:539 }
11:38:50:539 ERROR: unbalanced structure
11:38:50:539 Error:Invalid Connect message received. (outgoing, ooh323c_o_106)
[trixbox1.localdomain asterisk]#
I've scoured this forum and the internet. Last ditch effort was to post here. Any thoughts?
Thanks in advance.