Thanks Sherwood, But how do I send back a 302, once I'm already in the dialplan (hasn't asterisk already sent back a 200 OK by this point??)
Adrian -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: 23 May 2008 17:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfer Adrian Marsh wrote: > > Hi All, > > In my old telco days (SS7), if I was wanting to hand back a call to > the network for transfer to a different PSTN number, there was a > specific SS7 action I could take, which send the call back to the > network, which in turn then routed the call appropriately. It added a > transfer-number into the SS7 headers so that the originating number, > dialed number and transfer number all stayed to specs, and everyone > was happy. > > In SIP/Asterisk, it seems that I have to hairpin/trombone the call (to > have at least the control packets go via my SIP server), and use a > Dial out to the far end. > > So - is there a way of handing the call back to the network in asterisk ? > > My detailed problem is this: When a call comes in, I want to send it > onto users mobiles, if I hairpin the call that's OK, except the CLI > needs to be that of the originator (from the USERS point of view) so > they can decide if they want to accept the call. > > Here in the UK, this is where the issues begin... the carriers here > don't like it if your sending CLI for other countries, that don't > match what they think they should receive from that connecting > carrier. Eg, if a call coming to them is 13 digits, but they only > expect 11 from that carrier, then they cut the digits. This turns a US > originated call into a Southampton UK originated call! > > So I was hoping that handing the call back to the network in the > traditional sense would make it their problem and not mine... lol > > Thanks, > > > Adrian > > ---------------------------------------------------------------------- > -- > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users http://www.google.com/search?hl=en&q=asterisk+302+redirect+sip&btnG=Sear ch I believe you're looking for a 302 Redirect? Sorry if you're not, but that sounds like what you want _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
