SIP Trunking Between AVAYA IPO500 and Asterisk/Elastix/Freepbx

Sip trunk between Avaya IP Office 500 and Asterisk based pbx.

Prerequisites;

• You must have SIP Trunk license on your AVAYA according to your simultanous call count.
• On AVAYA, all users SIP names must be same as extensions number.
• Asterisk must have a SIP extension for AVAYA registration.

Do the following actions.

AVAYA IP Office:

SIP Line
ITSP Domain: Asterisks IP
IPSP IP : Asterisks IP
Prim Auth / Pass: <SIP Extension's username and password on Asterisk for AVAYA>
Registration Required : checked
In Service : Checked
Everything else : <leave default>

SIP URI
Add a new channel as below.
Local URI : Use User Data
Contact : : Use User Data
Display Name : Use User Data
Incoming Group : <Make sure to set the groups to something unique. I used group 20 for incoming and outgoing.>
Outgoing Group : <same as Incoming group>

Short Code : Code : 2XXX (According your asterisk's extension format, my extentions start with 2000)
Feature : Dial 3K1
Telefon Number : 2N"@Asterisk IP"
Line Group Id : Same as Outgoing Group as above
Rest is default.

Incoming Call Route :
Bearer Capability : Any Voice
Line Group ID : Same as Incoming Group as above

Destinations;
Time Profile : Default Value
Destination : . (dot)


ASTERISK
Create a SIP Trunk like this.
Trunk Name : IPO
Peer Details :
context=from-internal
host=AVAYA's IP
type=friend

Create Outbound Route.
Route Name : IPOffice
Dial Patterns : 2XXX (According your AVAYA's extension format)
Trunk Squence : SIP/IPO

Under General Settings
Set "Allow Anonymous Inbound Sip Calls" to yes

I tested this configuration and works well.

Good Luck