sending calls from a remote gateway to sip trunk

husaints

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#1
Hi,

We want to be able to send calls from a remote gateway to the sip trunk on our elastix system, which will terminate the call via the sip trunk. When making the call via the remote gateway which sends the call to the elastix IP the call comes in as an incoming call and is terminated to one of the extensions defined in the inbound route. Elastix is set to accept anonymous calls and all calls destination is one of the extensions 810. Hence the call sent from the remote gateway ends up calling the extension and does not dial the phone number which is to be dialed through the sip trunk.

We want to be able to send this call directly to the sip trunk to terminate the call via the trunk, how can we do this, as there is no option in the inbound roue destination to send the call to a trunk. Do I need to create an incoming SIP for accepting calls from the GW IP? How do I route the particular call to the particular SIP Trunk?

Eg.

GW 165.65.123.1(Dials 12121113456)------>ELASTIX SERVER------>SIPTRUNK(USA Termination)------>DIALS 12121113456


Thanks,
 

Patrick_elx

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#2
your should create a trunk in Elastix for your gateway to see it as a trunk.

What are your gateway settings ? What brand/model?

What trunk setting did you create for this gateway?
 

husaints

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#3
Ok I have created a SIP trunk for the gateway in Elastix with the outgoing settings being blank and the the incoming settings with host=202.202.xxx.xxx (which is the gateway address) and type=friend. Now calls from the gateway are being processed by the elastix server but the call is going through the inbound routes and goes to the destination extension defined there. It does not dial out the number as per the outbound routes and through the SIP trunk named TRUNK1.

I need the call to be dialed through the SIP Trunk (TRUNK1). How do I get elastix to do that? the gateway sends the call to be dialed as 1212xxxxxx which comes through the elastix and elastix should dial the number through TRUNK1. Its a simple setup but I am lost.

The gateway sending calls is a portasip server, which sends the calls dialed by sip clients connected to it. How is it possible to route incoming calls from a particular IP to a particular trunk?

Or to put it in another way how do I get the call to hit the "Outbound Routes" instead of the "Inbound Routes" If I can get the call from 202.202.xxx.xxx to hit the "Outbound Routes" rather than the "Inbound" that would sort it out!
 

dicko

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#4
The context on your tie-line trunk needs to be from-internal for it to access outbound routes, you can use "custom-contexts" for a more restricted access
 

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