Hi.

I'm working on updating the Asterisk documentation for the upcoming 1.8 
release, and wanted to collect some information on how (in practice) various 
bits of functionality are implemented.

Right now, I'm working on find-me/follow-me, and have a couple of questions 
about that.

(1) In the case where a call is being "hairpinned" through a PBX from the PSTN 
back to the PSTN (with SIP trunking, of course), and the carrier requires that 
calls originating from the PBX (in this case, the 2nd leg of the hairpin, but 
the carrier doesn't know this) bear the PBX's ANI/CID information... how do you 
indicate the CID from the 1st leg onto the 2nd leg?

The scenario is the following.  Call comes in, rings on a desk phone, person 
doesn't answer, so it starts ringing out to his cell phone via the PSTN.

We can't simply redirect the call back into the network, because we want to 
retain control over it (for soft transfers, hold/park, recording, voicemail, 
etc).  But we want the person who's cell phone is being run to indicate the 
original calling party, not the PBX.

This is shown as:

cell            PSTN            PBX             deskphone
                  ============> =============>          call rings in as 
555-5678

... no answer, so PBX starts ringing out to cell after 3rd ring, bridging 
outside caller,
deskphone, and cell phone into a "conference" ...

                                ||
    <===========        <=============                          call rings out 
as XXX-XXXX, but with
                                                                diversion or 
identify info of 555-5678


And we can't generate a From: header with the caller's information (555-5678) 
copied over from the first call, because the PSTN might consider that a 
"swatting" or "spoofing" attempt and generate a "405 Method not allowed".

So how else do we legitimately indicate to the PSTN that the call isn't being 
originated by us?  Can we use Diversion: or P-Asserted-Identity: headers?  And 
if so, of the SIP/SS7 gateways on the PSTN borders, which are more widely 
implemented?

(2) When the call rings out to the cell phone, is there a header that gets 
translated into SS7 that tells cell carrier being rung to not do a 
forward-on-no-answer to voicemail?  In the above scenario, we want the 
voicemail on the PBX to answer, not the voicemail on the cell's carrier to pick 
up if there's no answer.

Suppressing custom rings, etc. would also be nice.

Thanks,

-Philip



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