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
