Hi all,
I know that you can record a SIP call to file, but is it possible to ‘record’ direct to a soundcard?
I am trying to work out a way in which the individual people on a conference call could be captured and mixed (to adjust their levels) for a live radio show.
Ideally I would be able to capture just the incoming RTP stream and direct a copy to a sound card, which would then be mixed in the analogue domain outside of the PC.
Is this possible? If not is there some way to split/clone an incoming call to another endpoint/extension? (which could connect to sound card)
you could use the console channel (aka soundcard) and during your show use the * CLI to dial an exten which monitors that channel…
you could also put yourself and an IP phone or softphone (or the console channel) in a meetme room. Then use something like FOP to join callers to it as desired…
[quote=“mungewell”]I am trying to work out a way in which the individual people on a conference call could be captured and mixed (to adjust their levels) for a live radio show.
Ideally I would be able to capture just the incoming RTP stream and direct a copy to a sound card, which would then be mixed in the analogue domain outside of the PC.[/quote]
This is an interesting challenge - using Console channel would only allow the one stream already mixed in MeetMe to be “recorded”, if I understand it correctly. How do you “remix” individual levels? (Not sure how to do this even in analogue domain, because you still get only one channel.)
hmmm, to mix levels you might try meetme with FOP maybe (not sure if it supports changing gains). The analog way to to it is (IMHO at least) put a high impedence tap on each line and hook it to a channel on the sound board. When you want to talk to somebody, have the host answer the phone on that channel and unmute it on the board. Talk, then mute it again and hang up.
This is exactly what I would like to achieve, albeit in the digital domain.
It doesn’t look like Asterisk is going to be of assistance. I’m currently looking to see if something like VoIPong or Oreka can be made to write direct to sound card (or Alsa/Jack shim) directly rather than to a file…
Now I understand that you want to post mix caller(s) with the host in a studio, not individual callers in a conference. I think you can still use Console/dsp. If it’s a single caller (as is often the case with talk shows), you don’t even need Meetme. Simply have the incoming call Dial(Console/dsp), and answer from the console - or enable autoanswer. The entire conversation will be output to the sound card.
In the above scenario, the sound card actually receives voices from both the caller and the host (Mic or line-in). The easiest way to mix is to use your favorite Linux mixer (e.g., amixer, alsamixer) on the Asterisk box to control the PCM level (caller) and Mic/Line-in level (host). This could, however, worsen echo in some cases. You can always mute host in the sound card to output only the caller, then redirect digital output to another machine for mixing.