Jayesh Nambiar
2012-Mar-21 15:36 UTC
[asterisk-users] Bridging an Answered call in Asterisk with another call
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120321/7338c74e/attachment.htm>
Satish Barot
2012-Mar-22 05:03 UTC
[asterisk-users] Bridging an Answered call in Asterisk with another call
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 <jayesh.voip at gmail.com>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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120322/53dd65de/attachment.htm>
Jayesh Nambiar
2012-Mar-22 05:34 UTC
[asterisk-users] Bridging an Answered call in Asterisk with another call
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 <satish4asterisk at gmail.com>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 <jayesh.voip at gmail.com>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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120322/962bb52e/attachment.htm>