<DIV>Hi,</DIV>
<DIV> 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.</DIV>
<DIV> Thank you very much,</DIV>
<DIV>Ron</DIV>
<DIV> </DIV>
<DIV><B><I>albor@ipeya.com</I></B> wrote:</DIV>
<BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid"><BR>I am trying implement two-stage dialing. <BR><BR>Scenario is following: <BR><BR>1. * Dials SIP agent<BR>2. SIP agent answer the phone and provide dial tone<BR>3. * Sends DTMF string<BR>4. "Bridge" channel with calling party <BR><BR>I thought that something like:<BR>exten => _2XX,2,Dial_but_not_connect_(SIP/BYEXTENSION,10)<BR>exten => _2XX,3,Wait,1<BR>exten => _2XX,4,SendDTMF($DTMF_DIGITS) <BR><BR>Should do it. <BR><BR>Thank you,<BR>Alex Fedorov <BR><BR><BR>_______________________________________________<BR>Asterisk-Users mailing list<BR>Asterisk-Users@lists.digium.com<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR>To UNSUBSCRIBE or update options visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-users</BLOCKQUOTE>