Transfering between SIP trunks on the same server

Hello everyone,

I would like to know if it is possible to transfer a caller from one SIP trunk to another on the same FreePBX server. Both SIP trunks are with the same service provider but one is used for incoming conference calls only.
If possble we would like able to transfer the caller coming in on the wrong trunk to the conference SIP trunk

And also does it free up that line on the incoming SIP trunk once the transfer is complete?

Asterisk Ver 11.17.1
FreePBX 2.11

Thank You

maybe routing to other trunk is a better way to handle your case. transfer is for exten-exten.

Thanks James,
Routing, transferring, trunk to trunk transfer, whatever that case may be. The outcome I’m looking for is if a caller is picked up on one trunk can that caller be routed to another trunk on that same pbx by the recipient of that call. I’m not looking for the an option to route to another trunk if busy as that’s all black and white.

Guess after some reading the answer is looking like a “trunk to trunk transfer”
An outside caller needs to be transferred to another outside trunk even though it is initiate as they would do for an inside transfer.

After some thought about this I can now see a little more clearly as to how this may be achieved.

Chances are no.
(assuming a PSTN number assigned to conference trunk, ‘other’ trunk for regular calls, but callers are coming in on the normal line to somehow get to the bridge)

The “transfers” (assuming situation where caller comes in regular line and you transfer them to the conference 7 digit number) sequence happens on your asterisk server. The upstream PBX (SIP provider) would need to be entity signaled to transfer the call to the new trunk. (I’ve never seen that ability in asterisk)

If I understand correctly, you have e.g. trunk A with a main number that routes to a receptionist, IVR, ring group, etc. and trunk B that routes to a conference bridge. I assume that the bridge also has an ‘extension number’ that internal users can call.

If someone calls in on trunk A, you can transfer them to the bridge and the technical aspects of the call will be exactly the same as if they had initially called on trunk B. However, it will occupy a channel on A for the duration of the call. This could potentially result in another caller getting a busy signal if all channels are in use. If trunk A is pay-per-minute, there is usually no cost to obtain additional channels. If you have ‘virtual PRI’ pricing, your (expensive) channels are likely shared between the trunks anyhow. However, if you have unmetered incoming with a limited channel count (and it’s expensive or impossible to get more), that would be a problem.

Asterisk can indeed be set up to request the provider to transfer the call (and in other applications that’s quite useful), but in this situation the result would be worse – for the duration of the call, you would be billed for both the incoming and outgoing legs on trunk A (as well as the incoming leg on trunk B, if applicable).

Some additional details about your trunks and the associated plans would be useful.

Thanks guys, I really appreciate your answers. I’ve been reading a bit in regards to this setup and most are similar to what you say Stewart. Both trunk A and B will occupy a channel. The reason why the conference trunk has been setup is because there are not enough available lines for both callers and conference callers. I think my best option would be transferring the caller to an external number routing the call out and back in. Thank you all for the input its most appreciated.

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