Given an Asterisk PBX system version 20 or above. Given that a user calls another user, the call is missed and is sent to voicemail. While the caller is directed to voicemail, the caller receives another call and answers it, putting the first call on hold.
The client implements call holding by sending a SIP INVITE with SDP option a=sendonly. This means the caller audio will still be streamed to Asterisk by default.
Normally, if the peer was another SIP client, the audio wouldn't be heard by the peer user, which means another implementation decision is taken somewhere to mute the stream of the peer on hold?
What is the expected behavior when the peer is a voicemail program? How can the voicemail application be configured to handle call holding, and not record the audio stream when the call is put on hold?
Thanks!
