2010/6/16 Philip Prindeville <[email protected]>: > (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?
"Diversion" draft is expired but there is a new RFC for this stuf (History-Info header): http://tools.ietf.org/html/rfc4244 > 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. AFAIK it's not legal for a PSTN provider to accepts outgoing calls wich a CLI number which doesn't belong to the provider, this is, in the case you suggest the CLI of the 2nd leg of the hairpin must be a CLI assoiated to the SIP trunk client (the Asterisk PBX) and cannot be the cell original number. > 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 > > > 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? As said above, "Diversion" is expired. An untrusted client (as any client of a SIP provider) shouldn't send a P-Asserted-Identity header, but a P-Preferred-Identity. The the provider, aftter checking the PPI would append a PAI header with the verified SPI URI. Anyhow this is not important right now in this scenario. A solution could be Asterisk keeping the original From (the cell number 555-5678) and appending a "P-Preferred-Identity: tel:ASTERISK_PBX_VALID_CLI". Usually this would make the provider to display ASTERISK_PBX_VALID_CLI as the calling party number in the PSTN side (this is not what you desire however). Other solution would be Asterisk setting a From with ASTERISK_PBX_VALID_CLI and adding a History-Info (RFC 4244) containing the original all info (555-5678), but for sure the provider will ignore or just discard such information, as there is no way to show such information in the PSTN world. > (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. There is no such specification. Voicemail serves are not a "standard", they are just servers that automatically answer the call to bill it :) -- Iñaki Baz Castillo <[email protected]> _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
