Trixbox and Avaya IP Office 406

I have an Avaya IP Office 406 and I want that it is able to comunicate with a trixbox, particularly that the phone connected to the Avaya could call the Trixbox extensions number and viceversa.

I’ve added a new line in the Avaya pointing to the Trixbox IP address with codec g711u 64k and a funcion code 67 to route the call starting with this two number vs the trixbox.

With Freepbx in the trixbox I’ve created a custom trunk with a custom dial string
OOH323/[email protected]:1720
where XXX.XXX.XXX.XXX is the ip address of Avaya
and Maximum channels = 5

and in the outbound routes I’ve created one named
to-avaya
Dial Patterns= 67|.
Trunk Sequence= AMP:OOH323/[email protected]:1720

Now the Trixbox extensions can call with the Avaya extensions putting 67 before the extension number to call, but on the contrary when I try to call a trixbox extension from an Avaya ip-phone the trixbox extensions ring once and hangs up.

Is there anyone can help me to solve this problem?

My ooh323.conf is :

ooh323.conf
[general]
;Define the asterisk server h323 endpoint
language=it

;The port asterisk should listen for incoming H323 connections.
;Default - 1720
;port=1720

;The IP address, asterisk should listen on for incoming H323
;connections
;Default - 0.0.0.0: tries to find out local ip address on it’s own
;bindaddr=0.0.0.0
bindaddr=YYY.YYY.YYY.YYY

;H.225 channel port range
;Default range is 12030 to 12230, Accepts port range from 1025 to 65500
;h225portrange=12030,12230

;Alias address for for asterisk server
;Default - "Asterisk PBX"
h323id=ObjSysAsterisk
e164=100

;CallerID for the asterisk originated calls
;Default - Same as h323id
callerid=asterisk

;This parameter indicates whether channel driver should register with
;gatekeeper as a gateway or an endpoint.
;Default - no
;gateway=no

;Whether this asterisk server will use gatekeeper.
;Default - DISABLE
;gatekeeper = DISCOVER
;gatekeeper = a.b.c.d
gatekeeper = DISABLE

;Whether asterisk should use fast-start and tunneling for H323 connections.
;Default - yes
;faststart=yes
;h245tunneling=yes
;faststart=no
;h245tunneling=no

;Whether media wait for connect for fast start call
;Default - no
;mediawaitforconnect=no

;Location for H323 log file
;Default - /var/log/asterisk/h323_log
;logfile=/var/log/asterisk/h323_log

;Following values apply to all users/peers/friends defined below, unless
;overridden within their client definition

;Sets default context all clients will be placed in.
;Default - default
context=from-internal

;Sets rtptimeout for all clients, unless overridden
;Default - 60 seconds
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
; when we’re not on hold

;Type of Service
;Default - none (lowdelay, thoughput, reliability, mincost, none)
;tos=lowdelay

;amaflags = default

;The account code used by default for all clients.
;accountcode=h3230101

;The codecs to be used for all clients.Only ulaw and gsm supported as of now.
;Default - ulaw
; ONLY ulaw, gsm, g729 and g7231 supported as of now
disallow=all ;Note order of disallow/allow is important.
allow=ulaw
musiconhold=default

; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad
; h245alphanumeric, h245signal.
;Default - rfc 2833
dtmfmode=rfc2833

; User/peer/friend definitions:
; User config options Peer config options
; ------------------ -------------------
; context
; disallow disallow
; allow allow
; accountcode accountcode
; amaflags amaflags
; dtmfmode dtmfmode
; rtptimeout rtptimeout
; ip
; port
; h323id
; email
; url
; e164
;
;

Did you ever get this going??
I have Asterisk connecting to Avaya IP Office. It works well. There are some differences between my OOH323.conf and yours.
Firstly, g729 is supported. (My IPO is in an alaw country so I need to use g729) Secondly, my dtmfmode is different. see snippets below…

;The codecs to be used for all clients.Only ulaw and gsm supported as of now.
;Default - ulaw
; ONLY ulaw, gsm, g729 and g7231 supported as of now
disallow=all ;Note order of disallow/allow is important.
allow=g729
allow=gsm
allow=ulaw

; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad
; h245alphanumeric, h245signal.
;Default - rfc 2833
;dtmfmode=rfc2833
;dtmfmode=q931keypad
dtmfmode=h245alphanumeric
;dtmfmode=h245signal

I also have some other sections below this. These were missing from you post but may be in you .conf file.

;Define users here
;Section header is extension
[myuser1]
type=user
context=from-trunk
disallow=all
allow=g729
allow=gsm
allow=ulaw

[mypeer1]
type=peer
context=from-trunk
ip=xx.xx.xx.xx ; UPDATE with appropriate ip address
port=1720 ; UPDATE with appropriate port
e164=101

[myfriend1]
type=friend
context=from-trunk
ip=xx.xx.xx.xx ; UPDATE with appropriate ip address
port=1720 ; UPDATE with appropriate port
disallow=all
allow=g729
allow=gsm
allow=ulaw
e164=12345
rtptimeout=60
;dtmfmode=rfc2833
;dtmfmode=h245alphanumeric

Your setting in the IPO are important also…

Compression Mode = G.729(a) 8K CS-ACELP
H450 Support = none
and only check Enable Faststart and Out Of Band DTMF

This works for me. Good Luck.

I am also trying to create a H323 trunk between a legacy PBX and Trixbox. Where did you install the OOH323 from? Thanks.

We have an Avaya IP office. I want to connect a Trixbox. What hardware would I need if any for the trixbox to connect and communicate with the Avaya? Or is there a forum/site that deals with this type of setup.

This evening I used the above ooh323.conf as a template for connecting to an Avaya Open Office System. It works fine.

I sent the bind address to the NIC I wanted to listen for h.323 and made a custom trunk with the dial string ooh323/trunkname/[email protected]

You also have to set the option in the Avaya route for direct IP-IP connections to no or you will have problems with RTP (one way audio.

I am able to dial from the trixbox to the Avaya ok, Voice works fine also.

When I dial from the avaya to the trixbox I see the incoming call via the Asterisk CLI but it dont ring the ext. On the Avaya phone it seems like its connecting also it counts for about 2 seconds and then drops.

what could I be missing?

Ok I got the two systems working

I’ve followed this thread and a blog at https://www.blogger.com/comment.g?blogID=995159643138961238&postID=6947222251176802028&page=1 and we’ve got inbound calls working and outbound calls routing, but with no sound trixbox=>avaya although we can hear the avaya extensions just fine, and if they call the trixbox extensions, both sides can communicate. Are there any other settings we shoud look for/at to make the audio two way?

My ooh323.conf is:

[general]
port=1720
bindaddr=10.1.x.x
faststart=yes
h245tunneling=yes
mediawaitforconnect=yes
h323id=ObjSysAsterisk
e164=100
callerid=asterisk
gatekeeper = DISABLE
context=default
disallow=all ;Note order of disallow/allow is important.
allow=ulaw
allow=gsm
dtmfmode=rfc2833

[avaya]
type=peer
context=default
ip=10.1.x.x
port=1720
disallow=all
allow=ulaw
allow=gsm
allow=alaw
e164=102
amaflags=default


and the configuration is operative as show by: tbpbx*CLI> ooh323 show config tbpbx*CLI> Objective Open H.323 Channel Driver's Config: IP:Port: 10.1.2.28:1720 FastStart yes Tunneling yes CallerId asterisk MediaWaitForConnect yes Gatekeeper: No Gatekeeper H.323 LogFile: /var/log/asterisk/h323_log Context: default Capability: 0x6 (gsm|ulaw) DTMF Mode: rfc2833 AccountCode: ast_h323 AMA flags: Unknown Aliases: 100 ObjSysAsterisk

The Avaya was configured as described in the blog post listed above, with allowances for the differences between the Avaya product in that post and the product we are using. The devil is in the details, though, and substantially the same may not be enough. Any thoughts, feedback, or pointers would be gratefully received.

As I noted in my post, I had to set the direct ip-ip setting on the Avaya to solve the one way audio.

Have you used the Asterisk CLI debug facility to see what is going on?

Without it the Avaya was reinviting the endpoints that could not survive a NAT translation in the core router (the Avaya and Asterisk server where in different VLAN’s).

hi,

could you help me ? I connect * to AVAYA ( s8400 )and the voice is working fine in both directions, but when i call to a external IVR i can’t send dtmf…

tks

Hi.

Is the Avaya you are using a 406 V1? Do you know if it’s possible to create a h323 channel between Asterisk and a 406 V1?

Regards

HI ALL,

Please guide me how do i proceed for integration with Avaya Ip office PBX with third party application. Here we are using PHP Language for this. please any help me or guide if it is have a possible ways and suggestions.