I’ve just installed FreePBX, connected a single SIP trunk (from Flowroute) to a single dedicated extension. I now want to setup a block of DIDs from the same SIP trunk provider to permit a round-robin approach for incoming calls and a first-available approach for outgoing calls. Problem is that I have no clue if this is handled by FreePBX or the trunking provider. The trunking provider has no way to create a “hunt group” (think that’s what they’re called) and I cannot find any way to have FreePBX control how the calls flow before they come into the PBX.
Here’s the scenario. My SIP DIDs are…
800.555.5000 <- one toll-free
407.555.1000 <- multiple local
407.555.1001
407.555.1002
407.555.1003
407.555.1004
407.555.1005
Incoming calls to ALL of these numbers should go to the PBX and be handled by an IVR. Outgoing calls should simply pick the first available (I believe).
Sorry for the newbie question. Hoping someone can point me in the right direction. Not really sure where to begin. I tried the docs / wiki but am not entirely sure what I’m looking for. I’m more green than the FreePBX logo!
Thank you, in advance, for your wisdom… whomever answers.
You are very confused!
First of all outbound calls are not associated with a DID, they go out your SIP trunk and you can send any called ID you want. The carrier may identify the trunk by one of the DID’s but that is administrative, not technical.
Since flowroute bills per minute you can have all the inbound and outbound calls you want so the concept of a hunt group is meaningless. Whatever number the caller dialed will be sent as the DID and you can route the call however you want. Normally you would use the first number as the business line and the rest as direct numbers to extensions.
As far as how the calls are routed after they hit the IVR, you can send to a ring group and select the hunt method or to a queue for more granular control. You can actually route a DID to any module in the IVR, such as a conference, or an announcement.
I suggest you head over to our wiki and hit the getting started guide.
Let us know how you are doing.
Is there more than one wiki? I’ve ready: http://wiki.freepbx.org/
All of the pages off of “Site Spaces,” which sound like an intro discuss the install. The only topic that comes close to this question is “SIPSTATION Trunking” and that area is vague and is written more for someone using their service.
The “Getting Started Guide” briefly mentions trunks and routes (essentially one paragraph each) but does not detail the type of question I asked: http://www.freepbx.org/support/documentation/getting-started
So, all that being said, what exact section or URL do you suggest I read. Posting here was my last resort after NOT finding what I needed in these areas.
I will take a look. Didn’t my explanation make sense?
SIP is SIP so having an understanding of how the protocol works is useful. Keep in mind it’s more of what your provider allows than the protocol itself.