I have a call recording (audio) requirement that isn't addressed by local Monitor/Record features.
All signalling and media currently pass through the Asterisk servers, so that won't be an issue. Instead of locally recording audio, for certain calls I need to add what is effectively a 3rd leg to the in progress 2-leg call. This 3rd leg is a SIP dial to a URI and/or PSTN number. I'm thinking I have to do this with a conference bridge config and add a 3rd muted leg to the conference? Suggestions? -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
