Hello!
in Asterisk 11, I need to create a sort of room where a announcer can talk with two clients.
In detail:
I use “sort of room” because the announcer interacts hearing the opinions of two clients but they dont have to hear each others, they should interact only with the announcer.
My solution used is to put the presenter in a room Confbridge, while the two customers are directed to a extension ExtenSpy (on whisper mode) on the same room.
So far so good because , but I also need to record entire audio session using a SIP endpoint that can record an audio flow. If the SIP endpoint enters in the room he will record only the audio of the announcer but nothing regards the questions of the two clients.
Do you have any suggestions on how to proceed for a right audio record?
Nathan
I’ve resolved using MixMonitor and sox utility.
At the beginnig of the call conference I will record three audio calls using Mixmonitor for each channel, when call is finished I’ll use sox utility to merge the three audio in a unique wav file (for example using a sample script) and finally asterisk UA calls the SIP endpoint using an originate using application Playback(audio-file-merged).
This is not a real time recording but for my needs it works!
Bye
Nathan
Why didnt you use the confbridge voice recording feature to record the conference voice ?
wiki.asterisk.org/wiki/display/AST/ConfBridge
record_conference
yes/no
Records the conference call starting when the first user enters the room, and ending when the last user exits the room. The default recorded filename is 'confbridge--.wav and the default format is 8kHz signed linear. By default, this option is disabled. This file will be located in the configured monitoring directory as set in asterisk.conf