Thanks Andy Will try that
Thanks again. Foong ----- Original Message ----- From: "Andy Powell" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 6:56 PM Subject: Re: [Asterisk-Users] Call Transfer > Foong > > Take a look at the sample.call file, modifying the settings in there and copying the file to /var/spool/asterisk/outgoing will cause asterisk to dial the call.. an example config is below > > Channel: SIP/[EMAIL PROTECTED] > MaxRetries: 2 > RetryTime: 60 > WaitTime: 30 > Context: mysipcontext2 > Extension: 2000 > Priority: 1 > > This will make asterisk dial exten 1000 in the context mysipcontext when it's answered it will then call exten 2000 in mysipcontext2.. > > All you need is a script to lookup in the database and generate the script file for you and it's done. > > HTH > > Andy > > > *********** REPLY SEPARATOR *********** > > On 30/07/2003 at 16:30 Chee Foong wrote: > > >Hello Dan, > > > >Thanks for you reply. > > > >Base on you recomendation using the 'T' argument. I manage to do call > >transfer an it works really well. > > > >My problem comes when my boss comes out with a superb idea where the > >transfering process is automated without involving a human :( > > > >Say asterisk get 2 numbers (from database, text file, etc), one belongs > >party A and the other belongs to party B. Asterisk will calls both parties > >and do the tranfer automatically. In another words, asterisk is resposible > >to 'press' the '#' to do the transfer. I don't this can be achieve in the > >extension.conf not matter how you structure you dial plan. > > > >Perhaps, the only way is to write a apps and plug it into asterisk like all > >the asterisk modules such as Meetme. > > > >Any ideas? > > > > > >Foong > > > >----- Original Message ----- > >From: "Dan" <[EMAIL PROTECTED]> > >To: <[EMAIL PROTECTED]> > >Sent: Wednesday, July 30, 2003 3:42 PM > >Subject: Re: [Asterisk-Users] Call Transfer > > > > > >> Hi, > >> > >> It works if you put the 'T' switch in the dial line. > >> > >> You can then transfer the call from the caller. > >> I have tested it in the folllowing configuration and it works: > >> Call from a Cisco 7960 to an ATA 186. > >> Select 'Transfer" on 7960 > >> Call another extension (X-Lite) > >> Select again transfer on 7960. > >> The call remain between ATA and X-Lite. > >> > >> This is what you need? > >> > >> BR, > >> Dan > >> > >> ----- Original Message ----- > >> From: "Chee Foong" <[EMAIL PROTECTED]> > >> To: <[EMAIL PROTECTED]> > >> Sent: Wednesday, July 30, 2003 7:08 AM > >> Subject: [Asterisk-Users] Call Transfer > >> > >> > >> Hello all, > >> > >> I am in a situation where I need to use asterisk to call someone say > >Party > >> A. After the call to Party A got through, asterisk will put Party A on > >hold, > >> then asterisk will call Party B. If call to Party B got through, asterisk > >> will transfer Party A to Party B. > >> > >> I wonder if this features is implemented into asterisk. I have found a > >post > >> in asterisk mailing list: > >> http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html > >> > >> but that doesn't help much. > >> > >> If this features is not implemented, can anyone give me some point on how > >to > >> implement this in asterisk? Do I need to write an app like the Dial apps > >for > >> asterisk to load at start up? > >> > >> > >> thanks > >> > >> Foong > >> > >> > >> _______________________________________________ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > >_______________________________________________ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users