Hi Roger Cisco PBX behavior seems correct here .Also after transfer is complete in step 3 , Phone-A is out of picture after having BYE exchange with Phone-B and PBX. So during step 4, PBX should use PAI of Phone-B.
Regards Ankur Bansal On Tue, May 12, 2015 at 4:55 PM, Roger Wiklund <roger.wikl...@gmail.com> wrote: > Hi folks! > > Scenario: > > ITSP--------------PBX--------------PHONE-A--------------PHONE-B > > 1. Call from PSTN via ITSP to PHONE-A (via B2BUA PBX) > 2. PHONE-A answers the call > 3. PHONE-A makes a supervised transfer to PHONE-B (REFER within the PBX) > 4. PBX sends UPDATE/Re-INVITE to ITSP with updated SDP containing > connection details to PHONE-B. > 5. PHONE-B is now talking to the PSTN > > In the above scenario using a Cisco Callmanager PBX, during step 4. > P-Asserted-Identity containing PHONE-B is included in the UPDATE/Re-INVITE > to the ITSP. > > In the same scenario using a Mitel MX-One PBX, during step 4, > P-Asserted-Identity is included containing PHONE-A in the UPDATE/Re-Invite > to the ITSP. > > I'm struggling with this as we are using a recording solution and with > Mitel MX-One the SRS cannot tell that the call has been transferred. > > Section 3.2 describes they way Cisco does it (my interpretation at least) > https://tools.ietf.org/html/rfc5876#section-3.2 > > What's the correct way to implement this? The Cisco way or the Mitel way? > What's your take on this? > > Thanks > /Roger > _______________________________________________ > Sip-implementors mailing list > Sip-implementors@lists.cs.columbia.edu > https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors > _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
