There could be two types of forwarding. Network initiated and UE initiated. In the network initiated forwarding, the forwarding number is known to the network (User some-how configures this). The network, which is ideally a SIP Application server forwards the call depending on the error response or no answer from the called UE. In another case, the UE itself sends 302 with the forwarding party address in the Contact header and the proxy can send new INVITE to the forwarded party using this Contact received in 302. Both are widely used approaches. Moreover, it also depends on the capability of your network and UE. People have been using non-standard Diversion header to indicated the forwarding history of the call. History-Info header is the standard approach.
On Fri, Sep 5, 2008 at 7:07 PM, caio <[EMAIL PROTECTED]> wrote: > 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 > > _______________________________________________ > Sip-implementors mailing list > Sip-implementors@lists.cs.columbia.edu > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
