<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">If I understand you, then yes you can. I do this now. All our telco lines come through our analog NEC phone switch and then through FXO/FXS ports to my Asterisk. Asterisk handles voicemail and the menu system so when somebody dials "6" to get my extension the asterisk does the following:<div><br></div><div>Flash();</div><div>Wait(0.4);</div><div>SendDTMF(268);</div><div>Hangup();</div><div><br></div><div>I added the Wait(0.4) as I found that under heavy load the NEC would not catch the first DTMF digit after the Flash. This solution has worked for us for over a year now.</div><div><br></div><div>Some "bonus" information that may or may not be relevant to what you are doing:</div><div><br></div><div>We have a few SIP phones that we needed to be able to do the same kind of thing. We couldn't flash transfer to the Asterisk, but in the NEC I setup a outgoing trunk line (dial 8 to access) that goes to the Asterisk box. Then I setup a "forward all calls" on extension 268 (when I have my SIP phone active) to "dial out" to 8268. That way when somebody calls my extension it automatically forwards then to extension 268 on the Asterisk box.</div><div><br></div><div>Daniel</div><div><br></div><div><br><div><div>On Jul 23, 2008, at 3:57 PM, Ricardo Melendez wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0; "><div lang="ES" link="blue" vlink="purple"><div class="Section1"><div style="margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-US">Hi to All, I have a PBX (MAINPBX) from a Telecomm Provider, which have the feature to transfer calls (Incoming call -> Answer -> FLASH -> Dial Number to transfer -> Answer -> FLASH+4) and the call is transferred, but I have the need to implement an internal ACD using Asterisk as the PBX, the trunks connected to my Asterisk FXO ports are Extensions of my MAINPBX (ex., 5437, 5440 etc), all features work fine, but I have the need to make asterisk act as a normal telephone when transferring calls, I need to release the line (FXO port in my Asterisk) and make the transfer via the MAINPBX feature.<o:p></o:p></span></div><div style="margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-US">Otherwise I will use 2 lines of my Asterisk PBX to make the transfer and it reduce the incoming lines available for my ACD.<o:p></o:p></span></div><div style="margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-US"><o:p> </o:p></span></div><div style="margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-US">It’s possible send the commands FLASH, FLASH+4 using the incoming line to my MAINPBX via Asterisk like a normal telephone?<o:p></o:p></span></div><div style="margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-US"><o:p> </o:p></span></div><div style="margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-US">Thanks in Advance.<o:p></o:p></span></div><div style="margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-US"><br>Ricardo Melendez<o:p></o:p></span></div><div style="margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-US"> <o:p></o:p></span></div><div style="margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div></div>_______________________________________________<br>-- Bandwidth and Colocation Provided by<span class="Apple-converted-space"> </span><a href="http://www.api-digital.com" style="color: blue; text-decoration: underline; ">http://www.api-digital.com</a><span class="Apple-converted-space"> </span>--<br><br>AstriCon 2008 - September 22 - 25 Phoenix, Arizona<br>Register Now:<span class="Apple-converted-space"> </span><a href="http://www.astricon.net" style="color: blue; text-decoration: underline; ">http://www.astricon.net</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" style="color: blue; text-decoration: underline; ">http://lists.digium.com/mailman/listinfo/asterisk-users</a></div></span></blockquote></div><br></div></body></html>