I have two questions regarding a SIP 200 OK response that ATT (SIP trunk provider) is saying they don't like:
Flow: PSTN (ATT SIP) > SBC > SIP proxy (Avaya Session Manager) > SIP server 1) In the 200 OK response, the Contact header includes a number that they're not aware of. Between the Session Manager and the SIP server the number is changed from the PSTN to an internal number that the SIP server recognizes (which is the number in the contact header). Is that supposed to be a show stopper, I must be understanding this incorrectly, but I thought calls were routed back via record-route/via? 2) In the 200 OK, the SDP contains "a=fmtp:100 (null)". ATT is saying that (null) is not valid and is part of the reason the call is failing. I'm not clear on fmtp, is (null) not an allowed value and possibly a bug coming from the far-end SIP server? Thank you all for your time. _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
