Asterisk to IPO

[IamaSherpa]

The connection has been made via SIP Trunks, the end user now needs engaging external trunks (SIP) on the IP Office side but coming from the Asterisk users.
For my understanding the SIP to SIP Trunk connection (IN/OUT) is unavailable because anti-fraud policies, any suggestion ?

 

[amriddle01]

You can connect trunk to trunk with SIP on IP Office, no problem.
If you mean on the Asterisk, then this is the wrong place to ask

 

[hibroth]

You can't dial out from IPO if you don't have users there, you could try with vmpro if existing

 

[amriddle01]

Yes you can, you can take calls in from Asterisk on a trunk and push whatever is dialled to a normal SIP trunk, No attendant or anything needed

 

[IamaSherpa]

How to @amriddle1, simply by creating the two trunks? In my very old memories IPO did not allow that, who would be charged to those calls?

 

[amriddle01]

I've covered it a few times in the past, search for it, should be there.
Obviously calls will be charged to the IP Office if that's where it breaks out

 

[IamaSherpa]

You have hundreds of messages here @amriddle01, tried a search, no way, could you link me to something specific, please?

 

[janni78]

You need to make sure that a valid CLI is sent to the service provider so they can bill something.

Don't know if there is a step-by-step guide, it differs a bit depending on what your service provider expects.

"Trying is the first step to failure..." - Homer

 

[amriddle01]

So assuming you have/want it as follows:

Asterisk >>> SIP Trunk on LAN >>> IP Office >>> SIP Trunk on WAN >>> Public Tel Network

and not:

Asterisk >>> SIP Trunk on WAN >>> Public Tel Network >>> SIP Trunk on WAN >>> IP Office >>> SIP Trunk on WAN >>> Public Tel Network

Then all you need on the IP Office is a *** URI in the trunk from the Asterisk (to force calls that don't match extns to Incoming Call route) and an Incoming Call route as follows:

All fields blank/empty except the incoming line group, change this to the incoming group number of the SIP line from the Asterisk.
Destination set to be . (just a dot)

This tells the system to check any received digits on that line against system shortcodes and route calls accordingly

Done