I have a trunk that has unlimited free calls and can do free REFERs which are handled seemlessly as transfers to another ptsn number with the other parties CID. What I am trying to make is a custom trunk in freepbx which is basically asterisk but with fancy graphs, to have a custom dial line to use some part of the dialplan to do the following.
Call Trunk CID from Trunk Unlimited
REFER Trunk CID from Trunk Unlimited to the actual destination
Bridge the incomind call to Trunk CID to the origin
Wait until destination picks up or unavailable
Close call from the Trunk Unlimited
I tried doing some dialplans but I never got trunk cid to pickup correctly.
It would be interesting to make this work.
This is not accurate. FreePBX is a PABX which uses Asterisk for its lower level processing. Hundreds of lines of dialplan are part of FreePBX. FreePBX cannot do arbitrary configuration of Asterisk; it configures the PABX, not the underlying implementation.
Also this is the wrong forum for FreePBX.
This sequence is not possible. The bridge is formed in the upstream PABX or central office switch.
I get lost here. I’m not sure what Trunk CID means, although I have as supicion htat you may end up with subscriber busy at this point, if you have missed a step, which is that the initial caller calls in presenting “trunk ID” as their caller ID.
You should also make sure you understand your provider’s business model; if they end up charging less for this than someone who charged per leg, in the normal way, you may well be breaching some fair usage constraint.
If you are thinking of replacing the an original incoming call with your outgoing one, I think it is still the case that Asterisk doesn’t support outgoing REFER/Replaces. In any case, I would be surprised if the original caller’s provider would stop charging their customer.
Some providers penalise short calls and unanswered ones, to stop people signalling their presence free of charge.
In that case we expect you to provide those dialplans and logging which shows how they fail.