ok so you mean it like this? sip trunk -> opensips -> asterisk every call goes true opensips true asterisk to a extention, so asterisk keep track of all the calls.
so when extention 085* comes in (outside number) i do a dial from opensips to asterisk, asterisk knows it should dial 105, and then i can transfer the 105 call to 103? I think this will work also.. One question tho, what do you mean with: "It should work if 105, 103 and 104 have the same configuration for OpenSIPS and Asterisk." I don't have any asterisk information in the phones now, they all registred on opensips.. Iñaki Baz Castillo wrote: > > 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 > > -- View this message in context: http://n2.nabble.com/Transfer-issue-tp3877950p3892533.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users