[asterisk-users] MixMonitor not recording through transfer
I have the following scenario: Agent calls external number Mixmonitor starts recording call After agent speaks with customer they need to transfer them to an extension that will simply play a message Customer hangs up The problem is that the recording stops the moment the agent transfers the call to the other extension. We need the recording to include the message from the other extension. We use Asterisk 16 on this server. I know that the AUDIOHOOK_INHERIT function was deprecated long ago so I should not need anything extra to keep recording through transfers, or am I wrong? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Handling SIP refers when using a SIP Proxy
Hi,, When using a SIP proxy to load balance calls how do you make it that a call on an attended transfer reaches the same Asterisk box every time? I was told that in later versions of Asterisk there is some "magic" to make it work correctly when load balancing. TIA. Dovid -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
