Trixbox CE to Avaya CM

[vaoldschl]

My eyes are burnt out from reading forums trying to get this working and I have tried every configuration that somebody said was working with limited success. My Avaya side is running. I can call from Avaya to Trixbox with no problem at all. On the Trixbox side (using x-lite) I can't even get the call to try and go out. I'm sure I'm missing something fundamental in the setup but have been staring at it for so long it could be 10 feet tall and I would miss it. Here is my Trixbox setup with names changed to protect the innocent:

ooh323.conf

[general]
port = 1720
bindaddr = Trix.ip.add.ress
disallow=all
allow=alaw
dtmfmode=rfc2833

gatekeeper=DISABLE
context=default
progress_setup=8
progress_alert=8
h245tunneling=yes
mediawaitforconnect=yes

[Avaya]
type=friend
context=internal
host=Avaya.ip.add.ress
port=1720
disallow=all
allow=alaw
dtmfmode=rfc2833

extensions.conf

[internal]
exten => 50001,1,Dial(SIP/50001)

exten => _1XXXX,1,Dial(OOH323/${EXTEN}@Avaya)

sip.conf

[general]

context=internal
bindaddr=Trix.ip.add.ress
disallow=all
allow=alaw
srvlookup=yes
canreinvite=no

[50001]
type=friend
secret=38257
qualify=yes
host=dynamic
callerid="Asterisk Testing"
pickupgroup=1

So like I said, I can call from Avaya to the 50001 extension and the call is perfect. When I try dialing 11619 from the x-lite phone I get 'Your call can not be completed as dialed'. I am eternally grateful in advance.

 

[vaoldschl]

Update on this, not that I didn't know the H.323 was working to an extent but, I created an extension matching my desk using the GUI and gave it custom dial option of OOH323/${EXTEN}@avaya.ip.add.ress:1720 and now I can get the call to go across the trunk but it doesn't pass the digits dialed, I get my 'undefined extension' mailbox on the Avaya side. A trace on the the Avaya side shows the calling name and number come across the trunk but the dialed digits aren't coming through. I'll keep working!

 

[busster]

ooh323.conf

[general]
faststart=yes
h245tunneling=yes
gatekeeper = DISABLE

[definity]
type=friend
context=intern
ip=192.168.x.x
port=1720
disallow=all
allow=ulaw
canreinvite=no

extensions.conf

[general]
autofallthrough=yes

[intern]
exten => 2701,1,Dial(SIP/2701)
exten => 2702,1,Dial(SIP/2702)

exten => _1XXX,1,Dial(OOH323/${EXTEN}@definity)
exten => _2XXX,1,Dial(OOH323/${EXTEN}@definity)

sip.conf

[general]
context=intern
disallow=all
allow=ulaw
srvlookup=yes
canreinvite=no

[2701]
type=friend
secret=123123
qualify=yes
host=dynamic
callerid="2701"
pickupgroup=1

[2702]
type=friend
secret=456456
qualify=yes
host=dynamic
callerid="2702"
pickupgroup=1


I don't see a whole lot of difference except the ulaw.

This was a setup that we used about 2 years ago, but ended up scrapping it and using a PRI connection between the two with a cross over cable. This was smoother and more straight forward. Ultimately, this can be done even with older Definity G3's without CLan and Medpro.

 

[vaoldschl]

Thanks for the input, yours is one of the configs I've worked from previously. I appreciate your posting it. I did make the change to ulaw with no change in behaviour. Not that I expected it but I'll try anything at this point. The best I've been able to accomplish trixbox=>avaya is creating the extension with the custom dial string and that just won't cut it because Avaya isn't seeing the dialed digits. The PRI side would be an option but I'm looking for a cheap and dirty way to interface a phone system that will only run SIP and my not too much but a little older avaya that needs a SIP enablement Server to play that game.

 

[vaoldschl]

Got it working! Must be something about the GUI interface v. the .conf files in FreePBX/Trixbox. I added the trunks and routes in the GUI and am able to call back and forth now with no problem. Well, not no problem but at least less problem (I seem to be having some disconnect supervision issues). Now to connect the other sites and see how that goes. When I try and use the connection to route to a hub PBX and then leave the hub via my UDP/AAR tables in Avaya I am getting a BCC compatibility denial event. It's a good thing I like puzzles or this could drive me to drink.

 

[busster]

http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html#links

Check this link.

Once you get it, I would like screen shots if you don't mind.

 

[vaoldschl]

That link looks dead on for the Avaya side.

Once I have the complete solution tested I'll put it into an FAQ. That is after satisfying my boss by actually implementing it, of course, could be a week or so. Thanks again for the help.

 

[busster]

http://rblondon.blogspot.com/2008/02/set-h323-trunk-between-asterisk-and.html

Another link.

What do you mean dead on the Avaya side?

 

[vaoldschl]

Dead on: right on, spot on, correct. His config and mine are the same except for the extensions and ip addresses.

 

[busster]

GUI interface vs editing the files directly.

This will drive you crazy.

If you edit a file directly, save, and apply. This change will stand and work as you think. BUT, if you then go back and make a change through the GUI, save and apply, your first change will be gone.

Unique file changes have to be made in the non-traditional files like XX_addtional.conf, or XX_custom.conf. These will not be over-written.

All of your GUI settings are saved in a database. When GUI settings are applied, the database items are all re-written, any manual changes are gone.

Bottom line, make as many changes as possible through the GUI, things which can not be changed then make in the other files.

 

[busster]

FYI, you can see the GUI database by going into phpMyAdmin.

 

[vaoldschl]

I have calls flowing through the Trixbox now but had to create extensions in the GUI with custom dial of ooh323/12345@avaya.ip.add.ress:1720 to make it work. Not really a great solution as I have thousands of phones and would need an extension for each one this way. When I look at the call reports the channel is coming up OOH323/(null)-xxxx. It appears that the incoming line is not matching the relationships I set in the ooh323.conf file. Any thoughts on that one? I imagine if I can get the channel status fixed then I would be able to edit the extensions_custom.conf file with wildcards that will point to each of my sites.

 

[busster]

Can you not wildcard something like this?

exten => _1XXX,1,Dial(OOH323/${EXTEN}@avaya.ip.add.ress)

 

[vaoldschl]

I've tried that; it doesn't seem to recognize anything I do in the .conf files. It only takes if I use the FreePBX configuration screens.

 

[busster]

okay, try this:

In extensions_custom.conf, add line toward the top.

#include def_ext_custom.conf (or something similar)

Below that find this section "[from-internal-custom]"

and add:

include => from-definity-trixbox (or something similar)


Now use putty to get to Trixbox and create a file called def_ext_custom.conf in /etc/asterisk directory.

start this file with:

[from-definity-trixbox]

Now put your masking below this.

exten => _1XXX,1,Dial(OOH323/${EXTEN}@avaya.ip.add.ress)

 

[busster]

dont forget to restart Asterisk.

 

[busster]

Did this resolve the issue?

 

[vaoldschl]

I never resolved the issue in trixbox. I have removed trix and am now working on * strictly through CLI.

 

[vaoldschl]

So now, working with just * the extensions.conf wildcards are working, the peer relationships built in the ooh323.conf are working, the calls from * to Avaya are working and the calls from Avaya to * are working but the calls routed through * from one Avaya system to another have unidirectional audio (answer, disconnect and all the Q.931 stuff is working great).

BTW I know I don't need the * to connect the Avaya's to each other; I'm only trying it between Avaya's because I don't have programming access to the actual far end system (Siemens HiPath)

 

[busster]

So connection * to avaya1 is h323, what is connection * to avaya2?