El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió: > Yes, i just noticed that error myself, that's something else didn't had > that before today :-), but that's the whole issue, i think it's not > sending a new refer, it just creates a new call on line 2, and when i try > to press transfer and hang up the call disappears and in my phone screen i > see "transfer failed"
This is how attended transfer works: - A is speaking with B. - A puts B on hold and sends a *new* INVITE to C (and talks with him). - A sends a REFER to B with "Refer-To: sip:c...@domain;replaces=xxxx". - B then generates an INVITE to C with "Replaces" header. - C accepts the call and *replaces* the previous call (established with A) since the new INVITE contains a "Replaces" header with previous dialog information. > > the situation is like this: 104 is on Asterisk, 105 & 103 are on opensips, > 104 takes all the outside calls (for now i made it like this, so we are > able to transfer the calls announced) > > i call from my mobile, true the sip trunk to 104. I transfer a call from > 104 to 105, this works fine. Then i transfer the same call from 105 to 103, > these last 2 are both opensips extensions.. and that last part, doesn't > work. the ngrep of a call like this is what you can see in my last post It's not easy to guess the issue with this information. However, I describe a flow that should work: - Asterisk receives a call and calls to 104 (through OpenSIPS). - 104 transfers Asterisk to 105, so now Asterisk is speaking with 105. - 105 transfers Asterisk to 103, so now Asterisk is speaking with 103. It should work if 105, 103 and 104 have the same configuration for OpenSIPS and Asterisk. -- Iñaki Baz Castillo <i...@aliax.net> _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users