<font face="trebuchet ms,sans-serif">Be it meetme or confbridge, asterisk 1.2.x or Asterisk 10.x. Is it possible to query a channel and get its conference number in return?<br></font><br><div class="gmail_quote">On Thu, Mar 22, 2012 at 11:09 AM, Satish Barot <span dir="ltr"><<a href="mailto:satish4asterisk@gmail.com" target="_blank">satish4asterisk@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Jayesh, Personally I haven't worked on Congbridge :). <br>Confbridge has evolved a lot in 10.X. So probably you should have no issues using it.<div>
<div></div><div><br><br><div class="gmail_quote">On Thu, Mar 22, 2012 at 11:04 AM, Jayesh Nambiar <span dir="ltr"><<a href="mailto:jayesh.voip@gmail.com" target="_blank">jayesh.voip@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">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 !!<br>
I can definitely use the confbridge application for the same right; as I am working on Asterisk10. What do you suggest??<br>
<br>Thanks again,<br><br>--- Jayesh<div><div><br><br><div class="gmail_quote">On Thu, Mar 22, 2012 at 10:33 AM, Satish Barot <span dir="ltr"><<a href="mailto:satish4asterisk@gmail.com" target="_blank">satish4asterisk@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">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*.<br>
<br>e.g. Assuming your destination is SIP extension, have something like...<br><br>
Action: Originate<br>Channel: SIP/{your_destination_here}<br>Application: MeetMe<br>Data: {your_meetme_number_here}<br><br>Hope this helps. <br>--Satish Barot<br><br><div class="gmail_quote"><div><div>On Wed, Mar 21, 2012 at 9:06 PM, Jayesh Nambiar <span dir="ltr"><<a href="mailto:jayesh.voip@gmail.com" target="_blank">jayesh.voip@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>Hello All,<br>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:<br>
1) User dials 123 from a touch screen Polycom phone.<br>2) Call comes to Asterisk and Asterisk answers the call and asks for PIN number.<br>
3) Once the PIN is validated, Asterisk sends a User Event through AMI which invokes a browser in the Polycom phone.<br>4) The Browser will have a Text-Box to Enter the destination number where the caller wants to be bridged.<br>
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 !!<br>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.<br>
<br>Is there any straightforward application or function to achieve this in Asterisk.<br><br>Any ideas or directions will be of great help !!<br><br>Thanks,<br><br>--- Jayesh<br><br>
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