The use of Diversion was not adopted by the IETF; however many vendors support it. RFC 4244 is the IETF approved alternative to Diversion.
In addition, I should mention that some vendors use B2BUAs instead proxies to perform the forwarding services that you mentioned. Thus their call flows would be different from what is shown within draft-ietf-sipping-service-examples. > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On > Behalf Of caio > Sent: Friday, September 05, 2008 9:37 AM > To: Manoj Priyankara (NOD) > Cc: sip-implementors@lists.cs.columbia.edu > Subject: Re: [Sip-implementors] call forward on rfc > > Manoj Priyankara (NOD) escribió: > > Hi, > > > > I also tried to find information in the call diversion. Ie, when a > > call is forwarded to a SIP UAC, how the INVITE differs from > a normal INVITE. > > But I think you can find "Diversion" header in the INVITE message > > > > // > > Manoj > > Do you mean Diversion header, that which is based on levy > draft http://tools.ietf.org/id/draft-levy-sip-diversion-08.txt ? > > I don't know if Diversion must be set before a response of > "Moved Temporarily" or another msg., or if our sip proxy > knowing about the forwarding situation must attempt to send > the invite with diversion already included... > > Examples here (posted by Brett Tate) > http://tools.ietf.org/html/draft-ietf-sipping-service-examples > -15#page-76 > are using "181 Call is being forwarded" as if the proxy sip > was a user agent but explains that the forwarding could be > accomplished using redirect "302 Moved Temporarily" response.. > > Seem to be many options, but don't know what is > recommended..or which is absolutely not recommended. > > Thanks.. > claudio > > > > > > > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On > Behalf Of > > caio > > Sent: Friday, September 05, 2008 12:35 AM > > To: sip-implementors@lists.cs.columbia.edu > > Subject: [Sip-implementors] call forward on rfc > > > > Hi, > > Can anybody point me on RFC3261 (or which rfc) about call > forwarding > > on no answer, or on busy, or unconditional? > > > > I do not know how must be an INVITE when the call is forwarded for > > example from a sip proxy to a pstn gw... > > > > My tests indicates that invite uri is final destination and "To:" > > header > > > > field is the number who did the forward (or has the fwd enabled). > > > > Thanks for any info you can provide me.. > > > > -- > > caio _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
