gtekcommunications at November 1st, 2010 14:51 — #1
I have a group of DIDs coming in from Bandwidth.com on a trunk that do not have a specific extension. I would like to get these to rollover to each other if one or more are busy.
With our 4 analog lines, the Telco set this up as a Hunt Group. Bandwidth.com says they do not offer any sort of Hunt group or call forward when busy setup.
Is their a way to do this in FreePBX? The only relevant things that I have found are Ring Groups & Follow Me, but these would both produce an endless loop until the timeout is reached.
Basically I would like it as follows:
DID1 is busy, go to DID2
DID2 is busy, go to DID3
DID3 is busy, go to DID4
and so on.
Anyone know of any type of module that supports this?
gtekcommunications at November 1st, 2010 17:29 — #2
If I'm understanding you correctly, then it would not matter.
I have 4 trunks with their own #'s, and I have three DID's coming into the same machine.
I'm not worried about the analog lines because Verizon takes care of the Hunt group.
So you are saying that if you have a catch-all inbound route (which I have), then it would not matter what number it came in on, even if it was busy, that if there are enough trunks, the call will come in?
Sorry if that seems obvious to most, I'm still new to the VoIP world, and even telephony in general.
gtekcommunications at November 1st, 2010 15:58 — #3
Perhaps I should have not included the part about the trunk. We have more than one trunk from Bandwidth.com, not sure how many exactly, but more than enough to accommodated what I am asking.
Sorry for the confusion.
monkeybone at November 1st, 2010 17:17 — #4
that's a new request I've never heard doing and it sounds like that would be outside of Asterisk. I guess the reasoning on why you would do this is different to me .... but I'd like to know the business case anyway.
Asterisk allow you to set up a generic inbound routes and if you don't specify the DID it handles all the incoming calls.
But if you wanted to do something like your want I guess you could use Asterisk queues and limit the number of callers into the queue and send the overflow into queue 2 , queue 3 and so on.
perhaps and guru has a better idea ?
monkeybone at November 1st, 2010 15:49 — #5
Not sure if I understand why you would do this.
One SIP trunk allows you to have a single 2-way conversation. You can attach multiple DIDs to that trunk but you only have only one trunk so you can only handle one caller at a time anyway.
Am i missing something ?
monkeybone at November 1st, 2010 18:03 — #6
you are correct,
You should have an inbound rule that does not have a DID set. This is a catch-all for routing you may forget to program.
gtekcommunications at November 1st, 2010 18:05 — #7
Great, I was not aware of that, but it does make sense after thinking about it.
Thank you very much for the quick responses!