On 4/11/13 3:01 AM, onewhoknows wrote: > 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 >
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