SIP Trunking DDI Users must have an active Public Number (DDI) on the IPVS Platform. Legacy PBX users/extensions (who do not have a SIP Trunking User account provisioned) must present the SIP Trunk Group number or ‘Bearer Number’ as their outbound CLI for calls to be able to traverse the IPVS platform. Trixbox SIP Trunk Settings & VoIP Configuration Setup. Trixbox, with a lowercase 't', is an IP-PBX software solution designed for small and medium-sized businesses. Trixbox comes in two flavors: the open-source community edition and a hybrid-hosted, commercially-proven solution. From here choose Setup from the menus in the middle of the screen. From the 'Setup' menu choose Trunks and then choose Add SIP Trunk. Create a SIP Trunk. Outgoing Settings. For Trixbox users using SIP protocol, leave the Incoming Settings box and name completely blank.
Avaya IP Office Sip Trunk to Asterisks / TrixBox / FreePBX
Avaya IP Office Sip Trunk to Asterisks / TrixBox / FreePBX
Sip Trunk Setup Trix Box Manual Free
I figured I'd post this HowTo on how to setup a SIP trunk between a IP Office and a Asterisks Phone system for intra extension dialing because I just spent the last 3 days trying to figure this out, and there seams to be plenty of articles on how to do this via H323, but there are very limited docs/HowTo's on doing this via SIP.
At the end of the day, this was a lot simpler that I expected, but I learned alot in the process.
------------------------
Avaya IP Office Side:
------------------------
1) Create a new SIP Trunk ( SIP Licenses are required for This )
2) The only thing you fill in here is the IP address of Asterisks/TrixBox,FreePBX box. Please note that this is being done anonymously, so I assume the two machines are either on the same LAN or connected securly via a VPN> I would not recomened this setup if you are doing this over the internet.
ITSP Domain: <blank>
IPSP IP : Asterisks IP
Prim Auth / Pass: <blank>
Eveything Else: <blank>
Check in Service box
3) Set Up SI URI - ADD a channel and set this to <use user Data> - Make sure to set the channel to something unique. I used group 420 for incoming and outgoing.
4) Other Tabs can be left default settings
5) Create a short code for calling the Asterisk Box. The extension on my asterisks are 4xx. So my SC looks like this:
Code: 4XX ( Change this with your extension format )
Feature: Dial 3k1
Tel Number: 4N'@x.x.x.x' (Replace with IP of asterisks)
Line Group : 420 (Group Id Set in URI)
6) Create an Incoming Call Route Like this:
Bearer: Any Voice
Line Group : 420
<the rest you leave default>
Under Destination
Default Value is . (just a period)
7) Under Each User, make sure to set their Sip Name to their extension number under there SIP Tabs.
-------------------------------
ASTERISK SIDE via FreePBX GUI
-------------------------------
1) Create a SIP Trunk that looks like this:
Trunk Name: IPO
Peer Details:
host=x.x.x.x (IP of IP Office)
type=friend
2) Create an Outbound Route
Route name: IPOffice
Intra Company Route <checked>
Dial Patterns : 2XX ( Replace with the format of your IP Office extension )
Trunk Sequence: SIPIPO
3) Under General Settings
Set 'Allow Anonymous Inbound Sip Calls' to yes
That should be it, you should now be able to call back and forth between the 2 systems as if they are one.
At the end of the day, this was a lot simpler that I expected, but I learned alot in the process.
------------------------
Avaya IP Office Side:
------------------------
1) Create a new SIP Trunk ( SIP Licenses are required for This )
2) The only thing you fill in here is the IP address of Asterisks/TrixBox,FreePBX box. Please note that this is being done anonymously, so I assume the two machines are either on the same LAN or connected securly via a VPN> I would not recomened this setup if you are doing this over the internet.
ITSP Domain: <blank>
IPSP IP : Asterisks IP
Prim Auth / Pass: <blank>
Eveything Else: <blank>
Check in Service box
3) Set Up SI URI - ADD a channel and set this to <use user Data> - Make sure to set the channel to something unique. I used group 420 for incoming and outgoing.
4) Other Tabs can be left default settings
5) Create a short code for calling the Asterisk Box. The extension on my asterisks are 4xx. So my SC looks like this:
Code: 4XX ( Change this with your extension format )
Feature: Dial 3k1
Tel Number: 4N'@x.x.x.x' (Replace with IP of asterisks)
Line Group : 420 (Group Id Set in URI)
6) Create an Incoming Call Route Like this:
Bearer: Any Voice
Line Group : 420
<the rest you leave default>
Under Destination
Default Value is . (just a period)
7) Under Each User, make sure to set their Sip Name to their extension number under there SIP Tabs.
-------------------------------
ASTERISK SIDE via FreePBX GUI
-------------------------------
1) Create a SIP Trunk that looks like this:
Trunk Name: IPO
Peer Details:
host=x.x.x.x (IP of IP Office)
type=friend
2) Create an Outbound Route
Route name: IPOffice
Intra Company Route <checked>
Dial Patterns : 2XX ( Replace with the format of your IP Office extension )
Trunk Sequence: SIPIPO
3) Under General Settings
Set 'Allow Anonymous Inbound Sip Calls' to yes
That should be it, you should now be able to call back and forth between the 2 systems as if they are one.