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