I am sure most of you are not a novice as I am. I am fairly new to the world of networking. 2 years. Even more “newer” to the world of VoiP.
I have been tasked by my employer to set up a new voip network for my company, replacing the existing one. We currently use a Vega50 ISDN/DSL gateway and a linux sipexchange. They now want this set up to be moved over to freePBX (asterix) but to be still using this vega50 gateway.
If you guys don’t mind I intend to ask some basic questions which most of you probably know like the back of your hand! But I’ve been forever searching online but everything hasn’t clicked into place yet!
So here goes…
Reading on some forums The impression I got is that I first need to set up the Trunk links and from my understanding these are the actually phone lines that come in from our telephone provider to the vega gateway? When adding a trunk in free PBX should I be selecting SIP trunk or Custom trunk for this? I believe it should be custom trunk because the freePBX server would be going directly to the vega50. Correct? If so, what is a custom dial string that it is asking me to enter?
You will probably need to set up a SIP trunk from the gateway to the FreePBX server. The type of trunk you use depends on what you gate will need. If the gateway can provide connectivity via SIP (which most do) you would use SIP. If the gateway does not use SIP you will need to build the trunk using whatever protocol the gateway provides.
To add, since the gateway is already configured you might look at your existing PBX or the gateway itself to determine some if not all of the parameters you will need to enter in FreePBX to create the trunk.
hi. ok Thanks. will have a look into this and report back. except that current sipexcange, name sipxconfig, looks much less simpler. But yes, will have a look.
hmmm still no luck. So i’ve come to realise that what I need to do is set up a SIP trunk and not a custom trunk. But what are peer details and to what device am I peering? Is this the IP phones or the gateway?
