Hello,
I have a Nortel CICS with some M7310 sets. Great system but I am wanting to integrate VoIP.ms trunks rather than using the POTS lines. My first idea was to simply use an ATA (one end in POTS lines, and the other in our network registered to VoIP.ms). As expected this worked fine, but we were wanting more that two simultaneous outside calls (inbound or outbound). Our SIP trunk support way more than two channels but since you can only have one channel per POTS line, it’s not practical because you’d have to buy lots of ATA’s or one with a lot of lines (our CICS only has 4 lines). I then came up with the idea to use FreePBX as a “gateway” between the CICS and the outside world. All SIP trunks are connected to FreePBX with the ATA registered as an extension. While this still limits us to two channels, it gave us lots of other features (like different prefix’s for different CID’s). We also have some sip phone’s registered to FreePBX. We want to be able to dial between extensions (like 221 on nortel system, to 521 on SIP as an example). Triggering an outside line on the CICS is still internal (but external for CICS) until FreePBX dials out on the sip trunk. Our inbound routing is to just always connect all inbound calls to the CICS line’s extension (511 and 512). We have a call pilot 100 as our Auto attendant witch is a standard dial the extension or push one for a hunt group.
This makes it possible for the nortel phones to call the SIP phones, but I want the sip phones to be able to dial the nortel phones. This means they need to be able to call the auto attendant of the CICS and enter the correct DTMF tone’s for the extension, and then connect the two.
Does anyone know how to set up asterisk and FreePBX so that when I am on the SIP phones, and I enter example 221 (nortel set), it will call 511 and enter 221, then connect???
I would appreciate any tips!
I want to use the two systems like they’re one.
I have tried making custom destinations like: exten => 221,1,Dial(PJSIP/511,D(ww221))