Hi I would like to further ask if it is possible to transfer a call from openphone to pstn. i.e. i use openphone and asterisk -oh323 channel driver to make a call to a PSTN number through zap channel connected on that end.Then i wanna transfer that PSTN number to some other openphone extension/alias May i have a look at your extension to conf, as i am not clear with how to implement this.
Rgds Manoj k Gupta ----- Original Message ----- From: "Chee Foong" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 2:00 PM Subject: Re: [Asterisk-Users] Call Transfer > 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 __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users