Set H323 trunk between Asterisk and Avaya

Once asterisk and H323 is installed (previous post) follow the below configuration files to have the ip trunk up and running do the following configuration:

Setup h323.conf, sip.conf and extensions.conf as the examples below:

The h323.conf just defines the link between the Avaya and the Asterisk server.

/etc/asterisk/h323.conf
; The NuFone Network's ;
Open H.323 driver configuration ;
[general]
port = 1720
bindaddr = 192.168.58.227
disallow=all
allow=alaw
dtmfmode=inband
gatekeeper = DISABLE
context=default
progress_setup = 8
progress_alert = 8
h245tunneling=yes

[Avaya]
type=friend
context=default
host=192.168.58.216
port=1720
disallow=all
allow=alaw,g729,gsm,slinear

On sip.conf I set two demo extensions 89301 and 89300. This are numbers similar to my Avaya extension range and they also match a DDI range.


/etc/asterisk/sip.conf
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[89300]
type=friend
regexten=89300
context=default
secret=89300
username=89300
callerid="User1" <89300>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
dtmfmode=inband

[89301]
type=friend
regexten=89301
context=default
secret=89301
username=89301
callerid="User2" <89301>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
dtmfmode=inband


On extensions.con I set up a basic dial plan to send sip calls to each sip phone and calls from the sip phones to the Avaya system.


/etc/asterisk/extensions.conf

; extensions.conf - the Asterisk dial plan
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no

[globals]
TRUNKMSD=1

[default]
;Simple direct to sip extensions
exten => 89300,1,Dial(SIP/89300)
exten => 89301,1,Dial(SIP/89301)

;Simple rule to divert all calls from Sip extensions to Avaya extensions and External numbers on PSTN.

exten => _8XXXX,1,Dial(H323/${EXTEN}@Avaya); Avaya Extension
exten => _90ZXXXXXXXXX,1,Dial(H323/${EXTEN}@Avaya); External National UK Number


Configure the Avaya side:

The first thing to check is to see if you have the following:
a. On system-parameters customer-options in the second page you must have enough Maximum Administered H.323 Trunks to use.
b. A CLAN to use.
b. A MEDPRO to use.

Note that my Avaya PBX is a CM version 3.1.2, there might be other requirements that I might be missing.

1) Set up the names of the Avaya CLAN and the Asterisk server on node-names ip
(in the image below lmntel01 would be my Asterisk ip address and 192.168.58.216 my clan address)



2) Below are the details of my CLAN, note the network region defined. In this case 6.



3) The configuration of the CLAN is shown below:



4) The network region 6 has the following configuration:









c) The network region 6 has defined ip-codec 6 taht has the following codecs (NOTE: the use of this codecs is due system configuration/hardware limitiations. You should try to use something else than G729)




d) Now you have to define the IP Trunk and IP Signaling group. You have to create both on sync, below just the final output.






Note that the Supplementary Service Protocol needs to be set to "a" to allow the passing of call info (extension number, name, etc) from and to the two systems. This will be shown at the end of this post in an Avaya to Asterisk extension call.









e) At this stage you should have the H323 IP trunk up and running between the two systems. You might require a busyout of signalig group/trunk to bring it up.






f) The next step is to define the routing off calls from the Avaya to the Asterisk box using the new trunk created.

First I set up on the uniform-dial-plan that all my extensions begining wiht 893xx and 5 digits should use the ARS table. My Avaya dial plan is of 5 digits, and I do an inc-call-handling-trmt to change external DDI to an internal extension. For example 02070189301 is set to the internal number 89301, note that 89301 is NOT an extension that is defined on the Avaya system.



Then on the ARS analysis table I defined that the 893xx range should use the route pattern 66. (not sure which call type I have to use, on my test the last one was natl therefore it was left like that).



Finally the route-pattern 66 is defined using the trunk 66 previously created and that would be all the configuration needed.



One final step that I had to do was to complete my public-unknown-numbering table. I found on the test that while the extension name was sent to the asterisk server the extension name was not. External calls to the DDI worked fine and the caller number was passed but when calling from an internal extension there was a problem.

After googling and checking this I found that by setting the first digits of my Avaya extensions and the trunk all the Avaya extensions sent the number to the Asterisk server and were display on the IPSoftphone being used for testing.



An example of a call using X-Lite sofphone is below. From extension 84562 on my Avaya PBX I placed a call to extension 89300 on the Asterisk server. The configuration of 84562 is below:



You can see on the result that the name and the extension number is sent to Asterisk. This is due using Supplementary Service Protocol a on the second page of the trunk 66 definition as mentioned before.