Scott Gifford
2009-Jul-16 21:45 UTC
[asterisk-users] Stop recording on SIP attended transfer
Hello, We have an application where operators will sometimes take an incoming call from a queue, then contact an outside line, do a consultation, and finally do a SIP attended transfer to join the two parties together. We'd like to record the incoming caller's conversation with the operator and the attended part of the outgoing call, but not the unattended part, after the transfer has completed and the incoming caller is talking on the outgoing line (this part of the call may be confidential). We start the recording with a MixMonitor command when the outgoing call is placed. However, I don't see anything in the dialplan that gets run when the SIP attended transfer happens, where I could issue a command to stop recording. Any suggestions? Thanks! ----Scott.
Danny Nicholas
2009-Jul-16 21:52 UTC
[asterisk-users] Stop recording on SIP attended transfer
I don't know the full details, but I think if the Dial command(s) have the W and/or w options on them, you can activate/deactivate recording via DTMF. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Scott Gifford Sent: Thursday, July 16, 2009 4:45 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Stop recording on SIP attended transfer Hello, We have an application where operators will sometimes take an incoming call from a queue, then contact an outside line, do a consultation, and finally do a SIP attended transfer to join the two parties together. We'd like to record the incoming caller's conversation with the operator and the attended part of the outgoing call, but not the unattended part, after the transfer has completed and the incoming caller is talking on the outgoing line (this part of the call may be confidential). We start the recording with a MixMonitor command when the outgoing call is placed. However, I don't see anything in the dialplan that gets run when the SIP attended transfer happens, where I could issue a command to stop recording. Any suggestions? Thanks! ----Scott. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users