29 jan 2009 kl. 10.02 skrev Iñaki Baz Castillo: > El Jueves, 29 de Enero de 2009, Klaus Darilion escribió: > >> The enterprise has many assigned phone numbers (DIDs) which will be >> forwarded by the service provider to the contact registered by the >> PBX. >> >> Problem: How to signal the called phone number? Usually the called >> phone >> number is in the RURI, but the service provider retargets the RURI >> with >> the REGISTER contact, thus this information is lost. >> >> We found this workarounds: >> >> * To: header: the called number is placed in the to header: Ugly, PBX >> have to route based on To header, ..... > > Never, the TO header could be the original destination of the call. > If a > diversion occured then the TO still would point to the original > target. > Also the TO can arrive in very exotic ways: > > To: sip:001234...@xxxxx > To: sip:+1234...@xxxxx > To: sip:1234...@xxxxx > To: sip:01234...@xxxxx > ... > > >> * RURI: The service provider uses only the host part of the REGISTER >> contact and puts the phone number into userpart of the RURI when >> retargeting. Not standard conform, ..... (I think this is was >> SIPConnect >> suggests) > > It would break registration from normal phones which expect the same > RURI > username as the user they used in the registration. > > >> * RFC 3455: P-Called-Party-ID > > No idea. > > >> * routing instead of retargeting: called number in RURI, REGISTER >> contact in Route header > > I don't like. > > >> Any other workarounds left? > > Yes: > > a) The PBX could register as many times as DID has assigned, using the > appropiate DID for each registration. > > > b) The provider could add a custom header containing the AoR for > which the > user location was performed. I wrote sometime ago about it: > > https://lists.cs.columbia.edu/pipermail/sip-implementors/2008-August/020215.html > > I deployed a small SIP provider with OpenSer using this custom > mechanism to > allow multiple DID per PBX (so the PBX needs to extract the custom > header > value when receiving an INVITE from the SIP provider).
Why a custom header when RPID with party=called already exists? Your A option would require many changes in many service providers networks. /O _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
