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GoGo  
#1 Posted : Friday, June 23, 2017 12:30:35 PM(UTC)
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Is there anyone who can point me to which voice phone service I can use with Vmix, that have the capabilities of receiving multiple calls.

I already use Skype, but if I already have a caller and then someone calls in, unfortunately I am not able to answer and have them on hold (but still be able to listen to the show) without answering their call and put them through as a conference call.

I would really like to be able to have multiple calls on separate channels going through Vmix.
DWAM  
#2 Posted : Friday, June 23, 2017 12:51:50 PM(UTC)
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Hi Gogo

this has been discussed many times already. Did you search for similar posts?

Basically there are many options:

- using a voip service/provider or your own ipbx (I use Asterisk)
- using dedicated software solutions for AV like:

http://www.gnuralnet.com/

Search the forum to get more details.

Guillaume
GoGo  
#3 Posted : Friday, June 23, 2017 1:08:31 PM(UTC)
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DWAM wrote:
Hi Gogo

this has been discussed many times already. Did you search for similar posts?

Basically there are many options:

- using a voip service/provider or your own ipbx (I use Asterisk)
- using dedicated software solutions for AV like:

http://www.gnuralnet.com/

Search the forum to get more details.

Guillaume


Hello Guillaume, yes I have searched, but haven't found anything.

I clicked on the link you provided and they do not offer "Audio Call in only" Thank you :) And I looked into Asterisk, so right now I am looking to see if this will work for me.
DWAM  
#4 Posted : Friday, June 23, 2017 1:53:05 PM(UTC)
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Turns out I as remembering another recent post that... you already started ! ;o)

A lot of options were given then...

https://forums.vmix.com/....aspx?g=posts&t=9835
GoGo  
#5 Posted : Friday, June 23, 2017 2:17:46 PM(UTC)
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DWAM wrote:
Turns out I as remembering another recent post that... you already started ! ;o)

A lot of options were given then...

https://forums.vmix.com/....aspx?g=posts&t=9835



This is true, but unfortunately, none of those worked for what I need... Thank you :)
mjgraves  
#6 Posted : Friday, June 23, 2017 4:10:04 PM(UTC)
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Perhaps I can help. My partner and I use the ZipDX conference bridge to allow people to participate a weekly podcast/hangout. We've done this since 2008. Prior to Hangouts, the show was hosted on Talkshoe. The show is about telecom topics, so we have a strong background in telephony.

A ZipDX conference is accessible via access lines in 60 countries. Also, the service can dial-out to anyone, anywhere.

Further, the service has a WebRTC-based WebPhone that allows HDVoice connectivity from computers running Chrome or Firefox.

Finally, the service is available via SIP. That means that it can be interfaced to vMix using a standard SIP soft phone. I happen to use Bria.

All audio processing happens using 16 KHz sampling, delivering an 8 KHz audio channel. The service supports G.722 and Opus. It's not production grade, but vastly superior to a traditional telephone call.

Here's why ZipDX is especially useful in this regard. A ZipDX conference has multiple "breakout rooms." These are essentially private side-rooms, separate from the main conference floor. You can use this mechanism to queue calls into your production.

You can use one of these rooms as the the "broadcast booth" so that anyone in the room can be heard in your production.

Then use the main conference floor as a "Green Room." A place to hangout until the guest is needed in the live production.

One of the rooms has what we call "one-way glass. Anyone in the room can hear the conversation on the floor, but cannot be heard on the floor. If participants in the room have a local conversation the discussion on the floor is automatically ducked. When the local conversation is quite thy return to hearing the discussion on the floor.

You use our web-based dashboard to move people between rooms. It also allows you to mute/unmute people. Most importantly, it allows you to see who is connected.

ZipDX is not free. It's priced per-minute, per-connection. SIP and webphone connections are the lowest rates, since we don't have to pay intermediate networks for the transit.

Disclosure - I work for ZipDX and they sponsor said podcast. Their sponsorship extends solely to the use of the service. They're not pursuing this sport of business, but I'm happy to demo it to anyone who has a real project.
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GoGo on 6/23/2017(UTC)
sinc747  
#7 Posted : Friday, June 23, 2017 8:57:51 PM(UTC)
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What about using the call screen service in Blog Talk Radio?

- Tom
thanks 1 user thanked sinc747 for this useful post.
GoGo on 6/23/2017(UTC)
GoGo  
#8 Posted : Friday, June 23, 2017 10:00:29 PM(UTC)
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mjgraves wrote:
Perhaps I can help. My partner and I use the ZipDX conference bridge to allow people to participate a weekly podcast/hangout. We've done this since 2008. Prior to Hangouts, the show was hosted on Talkshoe. The show is about telecom topics, so we have a strong background in telephony.

A ZipDX conference is accessible via access lines in 60 countries. Also, the service can dial-out to anyone, anywhere.

Further, the service has a WebRTC-based WebPhone that allows HDVoice connectivity from computers running Chrome or Firefox.

Finally, the service is available via SIP. That means that it can be interfaced to vMix using a standard SIP soft phone. I happen to use Bria.

All audio processing happens using 16 KHz sampling, delivering an 8 KHz audio channel. The service supports G.722 and Opus. It's not production grade, but vastly superior to a traditional telephone call.

Here's why ZipDX is especially useful in this regard. A ZipDX conference has multiple "breakout rooms." These are essentially private side-rooms, separate from the main conference floor. You can use this mechanism to queue calls into your production.

You can use one of these rooms as the the "broadcast booth" so that anyone in the room can be heard in your production.

Then use the main conference floor as a "Green Room." A place to hangout until the guest is needed in the live production.

One of the rooms has what we call "one-way glass. Anyone in the room can hear the conversation on the floor, but cannot be heard on the floor. If participants in the room have a local conversation the discussion on the floor is automatically ducked. When the local conversation is quite thy return to hearing the discussion on the floor.

You use our web-based dashboard to move people between rooms. It also allows you to mute/unmute people. Most importantly, it allows you to see who is connected.

ZipDX is not free. It's priced per-minute, per-connection. SIP and webphone connections are the lowest rates, since we don't have to pay intermediate networks for the transit.

Disclosure - I work for ZipDX and they sponsor said podcast. Their sponsorship extends solely to the use of the service. They're not pursuing this sport of business, but I'm happy to demo it to anyone who has a real project.



If this system does what you have listed, then I would like to talk to you about getting it for my show. I got a chance to look at the ZipDX website and I read that it offers an even per minute discount if consumer is using SIP.
mjgraves  
#9 Posted : Saturday, June 24, 2017 10:15:58 AM(UTC)
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GoGo wrote:

If this system does what you have listed, then I would like to talk to you about getting it for my show. I got a chance to look at the ZipDX website and I read that it offers an even per minute discount if consumer is using SIP.


If you call the contact number given at https://www.zipdx.info/about/contact/ there's a good chance that I'll answer.

Or PM me and we can exchange contact details.
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