This is a pre-install question.
I am considering FreePBX/Asterisk as a adjunct solution but need to confirm how these features work with standard SIP trunks.
During a Follow-Me or VmX call that is placed to an outside E.164 number (cell phone);
Scenario 1 -
Call is presented to system on inbound SIP trunk based DID number. Station is rang, and based on Group call, Follow-Me or VmX feature, second call is routed to outbound trunk. Call appears on external E.164 number (hopefully with original CNAM) and subscriber answers/accepts call.
Does the system perform a network trunk to trunk transfer so the call does not remain hair pinned, and in effect tying up 2 trunks?
Scenario 2 -
Identical to above, only subscriber does not answer call, and end provider (cell company) routes call to their VM.
Does the system have the ability to sense the cell carrier redirect, terminate the call and forward the call to a mailbox? Thus avoiding having to check dual voicemail boxes.
I DO NOT want to terminate call based on time due to the variable call completion time involved with the cell phone carrier(s).
I realize that this may have limitations based on the carriers’ ability/willingness to provide these SIP messages (they are defined, just as they are in the ISDN/Qsig standards as well as in SS7). That is an issue
that I will deal with on the carrier side.