<div dir="ltr"><div>Hello Air,</div>
<div> </div>
<div>I did what you asked for but I got the following error:</div>
<div> </div>
<div>extensions.conf:</div>
<div><br>[stations]<br>exten => 442033553,1,Answer<br>exten => 442033553,n,Playback(demo-nogo)</div>
<div> </div>
<div>Error message:</div>
<div>[Sep 3 20:43:02] NOTICE[14092]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '442033553' rejected because extension not found.<br></div>
<div>Regards<br></div>
<div class="gmail_quote">On Wed, Sep 3, 2008 at 11:36 PM, Igor Hernandez <span dir="ltr"><<a href="mailto:emistz@gmail.com">emistz@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
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<div></div>
<div class="Wj3C7c">michel freiha wrote:<br>> Hi All,<br>> I bought a DID number from VOxbone...this number could be dialed from<br>> any PSTN line and could be forwarded to any SIP server like asterisk<br>> server...Now I need to forward this number to my asterisk server so when<br>
> a customer dial this number from his GSM or Land line PSTN number the<br>> call will be forwarde to my asterisk server and I need to play a wav<br>> file for example..<br>> Can you please give me some tips about how to accomplish this task?<br>
><br>> Regards<br>><br>><br></div></div>> ------------------------------------------------------------------------<br>><br>> _______________________________________________<br>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br>
><br>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona<br>> Register Now: <a href="http://www.astricon.net/" target="_blank">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" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br>Hello,<br><br>I have never used that provider but usually either the provider knows<br>
your switch's ip and routes the did traffic to it or you have asterisk<br>register with the provider so that it knows where to route the calls.<br><br>Once thats done you can do something like<br><br>exten => XXXXXXXXXX,1,Answer<br>
exten => XXXXXXXXXX,n,Playback(file)<br><br>Where the x's are the number that you see coming in from your provider.<br>If you're routed all your dids from what looks like one<br>number(callcentric does this) then you might need to use the sip header<br>
to route your did to the particular extension you want. You shouldn't<br>have to bother with this if you only have one did.<br><br><br>Regards,<br><br>--<br>Igor Hernandez<br>Escape Communications<br><a href="http://www.escapetel.com/" target="_blank">http://www.escapetel.com</a><br>
<br>_______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br><br>AstriCon 2008 - September 22 - 25 Phoenix, Arizona<br>
Register Now: <a href="http://www.astricon.net/" target="_blank">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" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
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