Jayesh, Personally I haven't worked on Congbridge :). Confbridge has evolved a lot in 10.X. So probably you should have no issues using it.
On Thu, Mar 22, 2012 at 11:04 AM, Jayesh Nambiar <[email protected]>wrote: > Thank you Satish. I was also thinking on similar lines. I was just > wondering if there was any mechanism with which we can bridge a new call > with the existing running call if the Call-ID of the call is known !! > I can definitely use the confbridge application for the same right; as I > am working on Asterisk10. What do you suggest?? > > Thanks again, > > --- Jayesh > > > On Thu, Mar 22, 2012 at 10:33 AM, Satish Barot > <[email protected]>wrote: > >> Make your user wait in a *Meetme* and then call your destination number >> through AMI and once he answers, place him in the same *Meetme*. >> >> e.g. Assuming your destination is SIP extension, have something like... >> >> Action: Originate >> Channel: SIP/{your_destination_here} >> Application: MeetMe >> Data: {your_meetme_number_here} >> >> Hope this helps. >> --Satish Barot >> >> On Wed, Mar 21, 2012 at 9:06 PM, Jayesh Nambiar <[email protected]>wrote: >> >>> Hello All, >>> I need to know a way of connecting an Answered call in Asterisk to >>> another call which was triggered by an AMI. I have a scenario as follows: >>> 1) User dials 123 from a touch screen Polycom phone. >>> 2) Call comes to Asterisk and Asterisk answers the call and asks for PIN >>> number. >>> 3) Once the PIN is validated, Asterisk sends a User Event through AMI >>> which invokes a browser in the Polycom phone. >>> 4) The Browser will have a Text-Box to Enter the destination number >>> where the caller wants to be bridged. >>> 5) The caller enters this number in the browser which is sent as a >>> Originate command to Asterisk through the AMI. Please note Asterisk does >>> not get this number as DTMF events !! >>> 6) Now, I need to BRIDGE this originated call from the AMI with the >>> actual caller who is already present in Answered state in Asterisk probably >>> listening to some music. >>> >>> Is there any straightforward application or function to achieve this in >>> Asterisk. >>> >>> Any ideas or directions will be of great help !! >>> >>> Thanks, >>> >>> --- Jayesh >>> >>> >>> -- >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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