GSM Trunk /SIP Extension Outbound Routes

Hi,

I have a GSM unit registed as a SIP extension (1001)
I have a normal extension as 1004

I would like to use this GSM unit as an outbound and inbound route, but as I am a beginner I an not sure how to configure FreePBX. I think I might need to change it from an extension to an outbound trunk.

Details:
Currently I have internal calls working, and some outbound routes via an online provider.

I would like cell pohone(GSM) calls to go via a GSM module with a SIM card. In reality this is connected to the network using a SIP ATA, and a short length of phone wire connects the SIP ATA to the GSM unit. The GSM unit (a nokia DTX-3) is configured to act like a telephone instrument.

On outgoing calls, I can currently dial 1001, then it connects and I hear a dialtone from the GSM unit. Then I dial the cell number to be called, and it connects fine.

For incoming calls via the cell, the unit answers, and I have configured it to immediately dial 1004 so an extension rings.

So basically I can call in and out to some extent. The hardware and SIP registration is working. Now I need call routing etc.

I would like the following prefixes to dial via GSM

01xxxxxxx , 083xxxx , 085xxx, 086xxx , 087xxx, 088xxx, 089xxx

I would NOT like to have to dial 9 for an outside line, just it to detect the numbers dialled. I will assume every call including local calls start with a zero 0,

I probably need to change my setup and set up new outbound routes. I am a real beginnner so ideally post setups I can try.

Thanks a million,

Dan in Ireland

Instead of an extension, use the same parameters and setup a trunk (use context from-trunk).

Then you will be able to use the outbound route module to decide what patterns go to that trunk.

For outgoing settings I need values for:
host=provider ip address
username=userid
secret=password
type=peer

but the SIP ATA I have is registered as an extension. It has a secret to send to the PBX, rather than expecting a secret to be sent to it?

Those are just the few examples. You don’t need anything but a host and a type. Everything else (all 140+) are options.

Suggest looking at Asterisk sample sip.conf to understand how to configure SIP trunks.

OK I have made some progress.

I have set up a custom trunk containing:
Local/[email protected]

When the route chooses this trunk, it connects to extension 1001 (which is the GSM unit, and the GSM unit gives a second dialtone; however the number to be dialled does not seem to be passed onwards.

I think I need to add $OUTNUM$ to the string above, but I am not sure where?

Again in summary, I would like the trunk to dial extension 1001, the extension will automatically answer and give a dialtone; then after a delay I would like the PBX to dial the outgoing number automatically.

Thanks in advance,

Dan in Ireland

The overdial is unnecessary. You should just be able to create a SIP peer and send the digits to dial to the ATA