On Tue, 10 Aug 2004 16:29:57 -0700, Chris Shaw wrote: >----- Original Message ----- >From: "Christopher Jacob" <[EMAIL PROTECTED]> >To: <[EMAIL PROTECTED]> >Sent: Tuesday, August 10, 2004 3:57 PM >Subject: [Asterisk-Users] SIP Transfers (Possibly reinvite) > > >> Hey Folks, >> >> Is it possible to transfer an incoming call back out without a "trombone" >> effect. >> >> For instance; >> >> Caller dials my broadvoice # --> Asterisk Answers and plays a menu --> the >> caller selects an option --> asterisk transfers the call to my cell phone >> via broadvoice and removes itself from the equation so I end up with... >> >> Caller --> Broadvoice --> Cell Phone >> >> Vs. >> >> Caller --> Broadvoice --> Asterisk --> Cell Phone >> >> >> Any ideas on how this could work? I'm thinking it's something to do with >> reinvite. >> >> Thanks >> >> Chris > >Unless you also have a PSTN connection (you didn't mention one) you will >actually be doing something more like this: > >PSTN -> BroadVoice -> Asterisk -> BroadVoice -> Cellphone > >Not sure if re-invite really applies here... Basically what reinvite does is >it uses the SDP information passed from a sip proxy to connect an IP phone >that is connected to * directly to the calling party (in this case, the >BroadWorks server), removing * from the equation... What you want to do is >invite the broadvoice server back on itself which would create a loop and I >don't think that will work... > >If you have enough bandwidth, the diagram shown above would work like a >3-way call with * as the initiator and your cellphone as the 2nd leg... If >you don't have the bandwidth, the other option would be to get an X100P card >and have * dial your cellphone through POTS when someone dials that >extension... This would remove you from a pure VoIP setup which is probably >not what you want...
I used a related setup to have DISA access to my * box for making overseas calls at cheaper rates. At present my setup works like this... cell -> VoicePulse Connect DID -> Asterisk -> VoicePulse Connect -> UK landline Since * jumps out of the way once the call is connected I was hoping to make the arrangement independent of my office bandwidth, etc. However I've been having some reliability problems thus far, primarily around random hangups. I was thinking of trying to use another provider for the outbound leg so that VPC is not handling the call twice. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 "Don't Panic" - from The cover on the Hitchhikers Guide to the Galaxy ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
