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?