Oke that is clear to me, So if i want the transfer function, i have to use a asterisk as a gateway (or any other pbx). But how do you create this new channel? i tried this before, with an asterisk as pbx, that received the outside calls... i've created a dial from asterisk to my opensips extention, but this is clearly wrong because if i try a transfer then, he tries it on the asterisk.. result the new destination not found..
What i understand from your story is that asterisk shouldn't do a dial, but a invite to the opensips extention, am i right? Any ideas on how to do this? Iñaki Baz Castillo wrote: > > El Viernes, 23 de Octubre de 2009, Peter den Hartog escribió: >> yeah i understand that, but why is it sending this refer to the sip >> trunk? >> i mean.. >> >> it's an outside call going to a local extention, i want to transfer from >> 1 >> local extention to another, so why isn't my opensips doing this refer? > > Sorry but you don't seem to understand how a REFER transfer works: > > 1) Your PSTN provider sends an INVITE to your proxy (opensips). > 2) OpenSIPS routes the call to user1. > 3) User1 answers the call and so. > 4) User1 wants to transfer the call to user2. > 5) User1 sends a REFER to *the PSTN provider* (through OpenSIPS as any in- > dialog request). > 6) The PSTN provider accepts the refer so *initiates* a new call to user2. > > Of course point 5 will NEVER work with a PSTN provider (and shouldn't work > at > all!). This is why PBX/B2BUA do exist: to enable PBX features. > > If you just have a proxy and receive calls from a PSTN you could NEVER > transfer that call to other user. > > Having a PBX/B2BUA the sceario changes and allows transference. An example > scenario: > > 1) Your PSTN provider sends an INVITE to your PBX (B2BUA). > 2) The PBX generates a *new* INVITE (a different dialog) to OpenSIPS. > 3) OpenSIPS routes the call to user1. > 4) User1 answers the call. > 5) User1 wants to transfer the call to user2. > 6) User1 sends a REFER to *the PBX* (through OpenSIPS as any in-dialog > request). > 7) The PBX provider accepts the refer so *initiates* a new call to user2. > 8) The PSTN provider didn't realize, at all, about the transference. > > > > > -- > 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-tp3877950p3878529.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