H323 trunk with Avaya CM

I configured h323 Avaya CM and Freepbx as following.

[general]
port = 1720
bindaddr = X.X.X.X
disallow=all
allow=alaw
dtmfmode=inband
gatekeeper = DISABLE
context=default
progress_setup = 8
progress_alert = 8
h245tunneling=yes
faststart=yes

[Avaya]
type=friend
context=default
host=X.X.X.X
port=1720
disallow=all
allow=alaw,g729,gsm

I have a “custom” trunk definition in FreePBX gui, it has this in the “custom dial string” OOH323/[email protected]:1720

how can I check if the trunk has established to Avaya CM from the Asterisk side - CLI.

Added a custom trunk in the freepbx gui.

Regards,

I ran H.323 trunks to Avaya CM as well. It is working fine.
Test by making a call.
If it doesn’t work, paste the CM trunk and signaling group settings here.

My custom dial string is: ooh323/[email protected]
And these are my settings in ooh323.conf:
[general]
port=1720
bindaddr=10.1.1.121
h245tunneling=yes
h323id=ObjSysAsterisk
e164=100
gatekeeper = DISABLE
logfile=/var/log/asterisk/h323_log
tracelevel=6
context=default
disallow=all
allow=ulaw
dtmfmode=inband
faxdetect = cng
directmedia=no
directrtpsetup=no
progress_setup=8
progress_alert=8

[avaya-h323]
type=friend
context=from-internal
host=10.1.1.199
port=1720
disallow=all
allow=ulaw
faststart=no
dtmfmode=q931keypad

hi,

thanks for your input. My Avaya CM is generating the below logs.
I’m getting an IVR from Asterisk saying “Your no. cannot be completed as dialled, pls check the no. and dial again”

time data

11:52:09 active station 5457 cid 0x5e
11:52:09 G711A ss:off ps:20
rgn:1 [10.1.6.14]:20272
rgn:1 [10.1.3.6]:2092
11:52:13 dial 400# route:UDP|AAR
11:52:13 term trunk-group 25 cid 0x5e
11:52:13 dial 400# route:UDP|AAR
11:52:13 route-pattern 25 preference 1 cid 0x5e
11:52:13 seize trunk-group 25 member 6 cid 0x5e
11:52:13 Calling Number & Name NO-CPNumber NO-CPName
11:52:13 Setup digits 400
11:52:13 Calling Number & Name 5457 Salahuddin Sa
11:52:13 Proceed trunk-group 25 member 6 cid 0x5e
11:52:13 G711MU ss:off ps:20
rgn:25 [10.1.3.25]:17438
rgn:1 [10.1.3.6]:2052
11:52:13 xoip options: fax:T38 modem:off tty:US uid:0x5014d
xoip ip: [10.1.3.6]:2052
VOIP data from: [10.1.3.6]:2052
11:52:20 Jitter:30 0 0 0 0 0 0 0 0 0: Buff:50 WC:59 Avg:30
11:52:20 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
11:52:21 denial event 1193: Equipment congestion D1=0x928 D2=0x370f2a
11:52:21 idle trunk-group 25 member 6 cid 0x5e
VOIP data from: [10.1.3.6]:2092
11:52:21 Jitter:0 0 0 0 0 0 0 0 0 0: Buff:13 WC:4 Avg:0
11:52:21 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
11:52:22 idle station 5457 cid 0x5e

Regards,

The announcement might be the trunk Intercept Treatment that you are hearing on the Avaya end. When you make that call from Avaya, and you run the Asterisk CLI (asterisk -vvvr), do you get any output?
Do calls fail in both directions?

Here you go

Connected to Asterisk 11.19.0 currently running on freepbx (pid = 1727)
– Executing [[email protected]:1] ResetCDR(“OOH323/avaya-h323-5”, “”) in ne w stack
– Executing [[email protected]:2] NoCDR(“OOH323/avaya-h323-5”, “”) in new s tack
– Executing [[email protected]:3] Progress(“OOH323/avaya-h323-5”, “”) in ne w stack
– Executing [[email protected]:4] Wait(“OOH323/avaya-h323-5”, “1”) in new s tack
– Executing [[email protected]:5] Progress(“OOH323/avaya-h323-5”, “”) in ne w stack
– Executing [[email protected]:6] Playback(“OOH323/avaya-h323-5”, “silence/ 1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
– <OOH323/avaya-h323-5> Playing ‘silence/1.ulaw’ (language ‘en’)
– <OOH323/avaya-h323-5> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
– <OOH323/avaya-h323-5> Playing ‘check-number-dial-again.ulaw’ (language ‘e n’)
== Spawn extension (from-internal, 400, 6) exited non-zero on ‘OOH323/avaya-h3 23-5’
– Executing [[email protected]:1] Hangup(“OOH323/avaya-h323-5”, “”) in new st ack
== Spawn extension (from-internal, h, 1) exited non-zero on 'OOH323/avaya-h323

My requirement is very simple. I just need Conference from Asterisk.

Regards,

Do you have the same problem when dialing an extension that is not a conference?
H.323 into Asterisk has some limitations, e.g. dialing into ring groups won’t work.

I haven’t tried with any extension.

I think, I’m missing something on configuring @freepbx.

Where can I configure 400 as the conference system. Then dial 401 as the Conference no.

Regards,