-----Original Message----- From: Ron McMillin [SMTP:[EMAIL PROTECTED] Sent: Saturday, April 03, 2004 6:33 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] two-stage dialing
Hi, This is not gonna work, is it? Is there such thing as Dial_but_not_connect_? I am trying to do the same thing but don't know how to accomplish this. If you've or anyone here figured out, please let me know. Thank you very much, Ron [EMAIL PROTECTED] wrote: I am trying implement two-stage dialing. Scenario is following: 1. * Dials SIP agent 2. SIP agent answer the phone and provide dial tone 3. * Sends DTMF string 4. "Bridge" channel with calling party I thought that something like: exten => _2XX,2,Dial_but_not_connect_(SIP/BYEXTENSION,10) exten => _2XX,3,Wait,1 exten => _2XX,4,SendDTMF($DTMF_DIGITS) Should do it. Thank you, Alex Fedorov _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users << File: ATT00083.htm >> _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
