<div>Dear Sir,</div>
<div> </div>
<div>Please find below the error that we are getting when enabling 'sip set debug'.</div>
<div> </div>
<p>localhost*CLI> <br><--- Reliably Transmitting (no NAT) to <a href="http://83.202.82.39:5060">83.202.82.39:5060</a> ---><br>SIP/2.0 404 Not Found<br>Via: SIP/2.0/UDP 83.202.82.39:5060;branch=z9hG4bK5ac79f249887f915005b5d34415b1a56;received=<a href="http://83.202.82.39">83.202.82.39</a><br>
From: "961555555" <<a href="mailto:sip%3A961555555@voxbone.com">sip:961555555@voxbone.com</a>>;tag=48201<br>To: <<a href="mailto:sip%3ADID_Number@87.236.144.14">sip:DID_Number@87.236.144.14</a>>;tag=as3fc2e680<br>
Call-ID: <a href="mailto:8b40a2a5a4f6082b5d33a47d01bd111d@83.202.82.39">8b40a2a5a4f6082b5d33a47d01bd111d@83.202.82.39</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>Content-Length: 0</p>
<p><br><------------><br>[Sep 4 11:13:05] NOTICE[10388]: chan_sip.c:14035 handle_request_invite: Call from 'sip_proxy' to extension 'DID_Number' rejected because extension not found.<br>Scheduling destruction of SIP dialog <a href="mailto:'8b40a2a5a4f6082b5d33a47d01bd111d@83.202.82.39'">'8b40a2a5a4f6082b5d33a47d01bd111d@83.202.82.39'</a> in 32000 ms (Method: INVITE)<br>
localhost*CLI> <br><--- SIP read from <a href="http://83.202.82.39:5060">83.202.82.39:5060</a> ---><br>ACK <a href="mailto:sip%3ADID_Number@87.236.144.14">sip:DID_Number@87.236.144.14</a> SIP/2.0<br>Call-ID: <a href="mailto:8b40a2a5a4f6082b5d33a47d01bd111d@83.202.82.39">8b40a2a5a4f6082b5d33a47d01bd111d@83.202.82.39</a><br>
CSeq: 102 ACK<br>From: "961555555" <<a href="mailto:sip%3A961555555@voxbone.com">sip:961555555@voxbone.com</a>>;tag=48201<br>To: <<a href="mailto:sip%3ADID_Number@87.236.144.14">sip:DID_Number@87.236.144.14</a>>;tag=as3fc2e680<br>
Via: SIP/2.0/UDP 83.202.82.39:5060;branch=z9hG4bK5ac79f249887f915005b5d34415b1a56<br>Max-Forwards: 69<br>User-Agent: Vox Callcontrol<br>Content-Length: 0<br></p>
<p> </p>
<p>Regards</p>
<div><br><br> </div>
<div><span class="gmail_quote">On 9/4/08, <b class="gmail_sendername">Jaswinder Singh</b> <<a href="mailto:vicky.r@gmail.com">vicky.r@gmail.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">[442033553]<br>user=442033553<br>type=pusers<br>secret=1234<br>host=dynamic<br>context=users<br>nat=yes<br>
<br><br>make it context=stations , i am assuming this is how your DID provider<br>is sending u calls ?<br><br>Let us know if your DID provider is just sending calls to your ip<br>address or you are registering asterisk server with the, . Keep<br>
context=stations in extensions.conf global section .<br><br>On Thu, Sep 4, 2008 at 2:41 AM, Igor Hernandez <<a href="mailto:emistz@gmail.com">emistz@gmail.com</a>> wrote:<br>> Hey,<br>><br>> Did you reload asterisk after changing the extensions.conf?<br>
><br>> Also, if you try it with "sip set debug" on the console what do you see?<br>><br>><br>> michel freiha wrote:<br>>> Hello Air,<br>>><br>>> I did what you asked for but I got the following error:<br>
>><br>>> extensions.conf:<br>>><br>>> [stations]<br>>> exten => 442033553,1,Answer<br>>> exten => 442033553,n,Playback(demo-nogo)<br>>><br>>> Error message:<br>>> [Sep 3 20:43:02] NOTICE[14092]: chan_sip.c:14035 handle_request_invite:<br>
>> Call from '' to extension '442033553' rejected because extension not found.<br>>> Regards<br>>> On Wed, Sep 3, 2008 at 11:36 PM, Igor Hernandez <<a href="mailto:emistz@gmail.com">emistz@gmail.com</a><br>
>> <mailto:<a href="mailto:emistz@gmail.com">emistz@gmail.com</a>>> wrote:<br>>><br>>> 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<br>>> 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>>> ><br>>> ------------------------------------------------------------------------<br>>> ><br>>> > _______________________________________________<br>
>> > -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a><br>>> <<a href="http://www.api-digital.com/">http://www.api-digital.com/</a>> --<br>
>> ><br>>> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona<br>>> > Register Now: <a href="http://www.astricon.net">http://www.astricon.net</a> <<a href="http://www.astricon.net/">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">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">http://www.escapetel.com</a> <<a href="http://www.escapetel.com/">http://www.escapetel.com/</a>><br>
>><br>>> _______________________________________________<br>>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a><br>>> <<a href="http://www.api-digital.com/">http://www.api-digital.com/</a>> --<br>
>><br>>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona<br>>> Register Now: <a href="http://www.astricon.net">http://www.astricon.net</a> <<a href="http://www.astricon.net/">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">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
>><br>>><br>>><br>>> ------------------------------------------------------------------------<br>>><br>>> _______________________________________________<br>>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br>
>><br>>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona<br>>> Register Now: <a href="http://www.astricon.net">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">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>><br>><br>><br>> _______________________________________________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br>><br>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona<br>> Register Now: <a href="http://www.astricon.net">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">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
><br><br>_______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br><br>AstriCon 2008 - September 22 - 25 Phoenix, Arizona<br>
Register Now: <a href="http://www.astricon.net">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">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</blockquote></div><br>