Bob Pierce
2008-Apr-17 14:45 UTC
[asterisk-users] Sip or IAX device with professional balanced audio out
Hi all, I've been googling for a solution here and haven't really come up with anything yet. We're doing an Asterisk install for a local radio station, and we're looking for a phone that they can use in their control room hooked up to their mixer board for recording calls. So, when you phone in for some contest or to request a song they record it and play it back a few minutes later on the air. They are currently recording calls from a hacked pots phone, but I was hoping for something a little more elegant with their new system. Has anyone run across a solution that might work nice here, or is there some other way of tackling this problem that I may have overlooked? Thank for your suggestions. Bob
Tim H. Panton
2008-Apr-17 14:53 UTC
[asterisk-users] Sip or IAX device with professional balanced audio out
How about avoiding the phone entirely in the playback phase? Have asterisk record the call to disk in MP3 or Slin, then use a pc with decent audio card to read it off the shared disk and feed it to the mixer. Tim. ----- Original Message ----- From: "Bob Pierce" <pierceb at westmancom.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Sent: Thursday, April 17, 2008 3:45:56 PM (GMT) Europe/London Subject: [asterisk-users] Sip or IAX device with professional balanced audio out Hi all, I've been googling for a solution here and haven't really come up with anything yet. We're doing an Asterisk install for a local radio station, and we're looking for a phone that they can use in their control room hooked up to their mixer board for recording calls. So, when you phone in for some contest or to request a song they record it and play it back a few minutes later on the air. They are currently recording calls from a hacked pots phone, but I was hoping for something a little more elegant with their new system. Has anyone run across a solution that might work nice here, or is there some other way of tackling this problem that I may have overlooked? Thank for your suggestions. Bob _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users