SIP call to RTMP Application Help

[Dave Kovalcik]

We have installed your SIP to RTMP Flashphoner application and have successfully implemented it into our application. However there is one feature we need and were wondering how you would handle the situation. We need to place the call on hold and not broadcast out the rtmp encoding while the call is on hold. To explain further - we have a meeting where a person initiates the SIP to RTMP call - but the meeting really has not started yet - and the problem is that when you initiate the call you are instantly on with all the participants. We want to be able to click a button on the web page and put the call in "waiting" mode - I know we can simply put the telephone on mute or on hold, but we want to control the meeting via a web interface. So a work model would look like this -
1. Call is initiated via entering a number in the web page and pressing a button "CALL"
2. SIP line is connected
3. RTMP connection is made
4. (Here's the Change) We want the call to be on hold initially
5. When the person is ready to start the meeting they press "BROADCAST" and audio instantly begins streaming over the web.

I know this function is available in the other SIP demos like phone dialer etc. could we somehow use that method?

Thanks for the help -

UPDATE ----
I searched the forums and it seems there was another post on this - and the response was no go - could you make this change? We would be willing to pay for it -

 

[Max]

I know this function is available in the other SIP demos like phone dialer etc. could we somehow use that method?

Yes this works for Phone-like features. However it does not work for SIP as RTMP forwarding.
We can implement this as a custom feature. Please contact sales@flashphoner.com

 

[Dave Kovalcik]

I sent an email to sales detailing what we want to do - thanks for the help you guys are really responsive...