<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Ciao Roand<div><br></div><div>I think you should buy a book and do some reading to build up your knowledge.</div><div><br></div><div>but in the meantime try something like this in the dialplan (extensions.conf)</div><div><br></div><div><div>exten => PSTN,1,Answer() ; Answer inbound calls or internal miss-dials</div><div>exten => PSTN,2,Playback(silence/1)</div><div>exten => PSTN,3,Background(enter-ext-of-person) ; input an extension</div><div>exten => PSTN,n,WaitExten(20) ; Adjust wait, default 5 sec</div><div>exten => PSTN,n,Goto(internal,${EXTEN},1) ; Goto the correct extension</div><div>exten => PSTN,n,Hangup() ; End the call</div><div><br></div><div>where PSTN is your sipura SIP name (1002 i think)</div><div><br></div><div>Ciao</div><div>Roberto</div><div><br></div><div><br></div><div><div>On May 24, 2008, at 3:09 AM, RoLaNd RoLaNd 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 class="hmmessage" style="font-size: 10pt; font-family: Tahoma; ">Hello all,<br><br>ive got the following setup currently:<br><br> <br> __Sipura 3102-----PSTN<br> |<br>Lan |<span class="Apple-converted-space"> </span><br> |<br> |__asterisk<br><br>i configured both asterisk and pstn to be able to receive/make calls through each other using sip of course..<br>but the thing is i want asterisk that when it receives an incoming call from sipura, to answer it, play msg that i recorded and wait for the caller to dial in an extension, where it would transfer the caller to that exntension, and in case the extension owner isnt available to answer it would direct him to his voicemail(tht i dont know how to set yet), and in case the caller didnt dial any extension in a certain amount of time, it automaticly directs it to a specific extensions i'd specify..<br><br>i tried the examples given in lots of forums and so on none of them worked, the phone keeps on ringing with every incomign dial plan ive specified without asterisk answering it..<br>the thing i did is that sipura directs incoming calls to 1002, so ive set the context of 1002 in sip.conf to a dial plan of [incoming-sipura] and ive set the commands i mentioned earlier tht i took out of those forums.. but theyre not working!!!<br><br>anyone has an example i could go on with ?<span class="Apple-converted-space"> </span><br>any help would be apreciated:)<br><br><hr>Discover the new Windows Vista<span class="Apple-converted-space"> </span><a href="http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE" target="_new">Learn more!</a><span class="Apple-converted-space"> </span>_______________________________________________<br>-- Bandwidth and Colocation Provided by<span class="Apple-converted-space"> </span><a href="http://www.api-digital.com">http://www.api-digital.com</a><span class="Apple-converted-space"> </span>--<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></div></span></blockquote></div><br></div></body></html>