[asterisk-dev] bridge two chan
Pavel Troller
patrol at sinus.cz
Sat Feb 13 00:01:36 CST 2010
Hi!
You don't need to develop any special application for this purpose, asterisk
has enough tools for this.
By dialling a callback feature code, just set a database entry, using the
called party as a key and the calling party as a value.
Then implement a 'h' extension, which will be called on hangup of every
connection. In it, query the database, whether the party going onhook was the
ordered one. If not, ignore the event. If yes, run a script, which will either
create a call file with appropriate parties, or use the 'originate' command.
You can even use the Originate() dialplan application in asterisk 1.6.2.
I hope these hints are enough for you to succeed with your task.
Regards, Pavel
> hi,
> call back scenario,
>
> persion1 call to persion2.
> if persion2 busy then
> persion1 enter some predecidef diget to set callback in his dialed phone.
> then he (persion1) hangup phone.
> when persion2 free(ie hangup his call) asterisk dial persion1 phone.
> when persion1 answer both persion1 and persion2 connect with each other
> and start taking.
>
> thanks for your reply,
>
> 2010/2/12 H?kon Nessj?en <haakon at avelia.no>:
> > On Fri, Feb 12, 2010 at 10:50 AM, Bhrugu Mehta <bhrugumehta at gmail.com> wrote:
> >> hi,
> >> actually i am developing feature callback.
> >> for this i am developing app callback.
> >> in this i need this.
> >> what function i have to used to calll to persion1 and when he answerd
> >> call to persion2
> >> and than both bridged together.
> >>
> >
> > I hope you know you can do this with a call file (or AMI Originate)
> > combined with app_dial? How would you execute the 'callback' in
> > app_callback?
> > If the module is calling both parties, something need to "call" the
> > app_callback app. I just don't see what you are trying to accomplish,
> > that isn't already as easy as cake.
> >
> > --
> > H?kon Nessj?en
> >
> > --
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-dev mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
>
>
>
> --
> Bhrugu Mehta
> Sr. S/W Engineer (D&D)
> VOIP,Telephony Team (Asterisk,zaptel etc.)
> India
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
More information about the asterisk-dev
mailing list