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iamralphsutton  
#1 Posted : Monday, October 22, 2018 9:34:09 PM(UTC)
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I am trying to figure out a solution for a VOIP call in show.

As in someone can be a call screener in one room - and then send calls to the live studio -

I have 6 lines coming in via VOIP - w my service provider. and currently - we can just answer a call - then when it's a good one - we can send it to the studio - and the host can talk.

The issue is - it's not really a viable solution - because - when the call is LIVE - we cannot answer any of the other calls.

I come from radio - we had a very expensive answer to this called a comrex. That would allow this - but they are thousands of dollars - and to my knowledge don't work on voip.

Wondering if anyone here has figured out a solution to do a live call in show properly?
TuesdaysLive  
#2 Posted : Monday, October 22, 2018 11:17:28 PM(UTC)
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vMix itself will not do this. However, there are ways one can incorporate third-party systems, services, or what you currently use, to do what you want.
tkachuk.alex  
#3 Posted : Monday, October 22, 2018 11:39:11 PM(UTC)
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Hi! We use IP virtual calling service, Binotel. So they make us different users. So we have employers that recieve calls, if it's good, they switch it to another user, in our case to our soundman at studio. He recieve it at laptop and send it to LIVE. At this time, we can recieve another call, and ask peoples to wait, or call back in few minutes
mjgraves  
#4 Posted : Tuesday, October 23, 2018 12:47:56 AM(UTC)
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I've been involved in this sort of thing for many years. The podcast that I co-produce each week invites participants to connect by telephone, with access lines in about 40 countries. We use a telephone conference bridge that is connected to a Hangout.

You can use whatever service you like for IP telephony. What you need from them depends upon what you hope to achieve. Do you want to take their calls and have someone vet them before passing them to the show? If so you need a switchboard like functionality that's common in hosted PBX services. That allows someone to answer each call, put them in a queue, before they get get passed to the show.

You receive each call, put them on hold (or in queue) then transfer them to the connection that corresponds to you live studio. This is just like a receptionist.

We don't do that. We just allow people to join a conference call that's connected to the show. We can manage the conference, allowing people to speak (or not) as appropriate.

You can do this using a hardware solution like the Comrex system you mentioned, or by combining software and services. In my case, we have one sponsor who is in the conference business (http://www.zipdx.info) and another who is in the business of providing international phone numbers (http://www.voxbone.com)
SportsNetUSA.net  
#5 Posted : Tuesday, October 23, 2018 1:32:51 AM(UTC)
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Originally Posted by: iamralphsutton Go to Quoted Post

I come from radio - we had a very expensive answer to this called a comrex. That would allow this - but they are thousands of dollars - and to my knowledge don't work on voip.


Comrex does make units (such as STAC VIP, VH2) that work with VOIP. But as you mentioned, they are very expensive.

DWAM  
#6 Posted : Tuesday, October 23, 2018 2:54:45 AM(UTC)
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Check Asterisk server for VoIP.

I can think to several ways to do this with Asterisk (mainly as Michael said):
- one is to use a conference room as Michael said
- another one is to simply transfer the selected calls one by one to the specific line that is used for the show
- another one uses the Barge feature
- etc...

All these will be full-duplex

Guillaume
mjgraves  
#7 Posted : Tuesday, October 23, 2018 3:04:23 AM(UTC)
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Originally Posted by: DWAM Go to Quoted Post
Check Asterisk server for VoIP.

I can think to several ways to do this with Asterisk (mainly as Michael said):
- one is to use a conference room as Michael said
- another one is to simply transfer the selected calls one by one to the specific line that is used for the show
- another one uses the Barge feature
- etc...

All these will be full-duplex

Guillaume


Asterisk could be part of a solution. Freeswitch is another open source option.

The trouble is that you need to amass other resources around these things. If voip is not your area of expertise, you might be better of using a hosted PBX service to deliver a complete solution. While there are many of these, I use OnSIP which has a great WebRTC implementation as well.

PS - Cluecon 2018, the annual Freeswitch user conference, was live streamed using vMix and a couple of PTZOptics NDI cameras.

PPS - There was rumor that Tom Sinclair and I might get into a discussion of this (telephony integration, Webrtc fundamentals) on one of his Streaming Idiots shows.
iamralphsutton  
#8 Posted : Tuesday, October 23, 2018 10:09:31 PM(UTC)
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I appreciate all the input -

I am going to go thru this and see if any make sense for us.

Thank you.

mgrayeb  
#9 Posted : Wednesday, October 24, 2018 8:52:02 AM(UTC)
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I've used Call In Studio (see www.callinstudio.com) with Skype once and it worked well for me. I had to use virtual audio cables but they're easy to set up. Very affordable and they have an option where you can use a live screener or an automated call screener where your own pre-recorded message instructs callers to record a brief message on what they're calling to talk about and then the system transcribes the message. Meanwhile, a producer can decide which calls to send to broadcast and type messages to the host about each caller.
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