Hi all! We recently discussed the following problem on the asterisk-dev list and are hoping for inputs how to solve it in a standard conform way.
Scenario: There is an enterprise using an IP PBX connected via a SIP trunk to the service provider which handles the PSTN connectivity. The PBX registers with the service provider using a single account (Aor). 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, ..... * 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) * RFC 3455: P-Called-Party-ID * routing instead of retargeting: called number in RURI, REGISTER contact in Route header Any other workarounds left? So what is the preferred mechanism? What do vendors and service providers implement? What should we implement in Asterisk? Thanks Klaus _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
