Really? Mmhh seems you got working what I want and I what you want.. Hehehe try using monitor instead of mixmonitor.. Maybe there is a difference in apps.
|-----Original Message----- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |John Daragon |Sent: Monday, March 27, 2006 4:56 AM |Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: Re: [Asterisk-Users] stop monitor on transfer | |Anton Krall wrote: |> Hi John, yes, Im using native transfer. What I do is use Monitor on |> the dialplan of the extension that picks up the call coming |from PSTN, |> so after that, if the extension forward or transfers the |call, monitor |> keeps recording all thru the end of the call no matter where it is |> been transferred to. | | |Hmmm. This is what I do: | |XXXXXX,1,NoOp() |XXXXXX,2,MixMonitor(${UNIQUEID}.wav) |XXXXXX,3,Dial(SIP/201,15,jTt) |.. | |The call is then SIP transferred by the receptionist, and |that's when the recording ends. | |I'll have a look at native transfer and see if that changes things ! | |jd | |_______________________________________________ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users