Hi Michel,<br><br>Do you have the extension 442033553 on the context users on your extensions.conf ?<br><br>I
have some DIDs from Voxbone (Best DID provider, by the way), and they
have a config section on their support page ... Take a look when you
get a chance ...<br><br>First, you should create sections on your
sip.conf or iax.conf (They'll descontinue iax support by january) by
their originating gateways, not to the extension ... e.g:<br><br>[81.201.84.21] (I renamed to their IP because you should authorize each one)<br>host=81.201.84.21 (Host the incoming call is arriving from, check Voxbone IPs list)<br>type=friend // Must be friend or user (user will phase out in Asterisk future versions)<br>insecure=very<br>context=incoming (context in your extensions.conf that calls from this gateway will be routed to)<br>disallow=all<br>allow=ulaw<br>
allow=gsm // Allowed codecs<br>
allow=g729<br><br><br>On
your DID config page, you'll choose the originating POP, each POP has
several IPs that you should create sections like the above on you
sip.conf. But from the errors you reported your Asterisk server
accepted the call but didn't know how to route it ... You sip.conf must
be with Autocreatepeer=yes but check the context setting ... This is
your default context, in case you don't want to create sections for all
Voxbone IPs (less secure)<br>
<br><br>So, from the errors you reported ... your configuration should look like this ....<br><br>Voxbone config page<br>sip:442033553@yourip<br><br>sip.conf<br>like the above description (Modify the IPs) or create a default context, in [general] context=whatever<br><br>extensions.conf<br>[whatever]<br>exten => 442033553,1,Answer<br>exten => 442033553,n,Backgroud(soundfile)<br><br>That should work ...<br><br>Bye ...<br><br>
<br>
<br>
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<strong>----- Original Message -----</strong><br><strong>From:</strong> michofr@gmail.com<br><strong>Sent:</strong> Wed, September 3, 2008 17:10<br><strong>Subject:</strong> Re: [asterisk-users] DID number<br><br>
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<div class="gmail_quote">On Wed, Sep 3, 2008 at 11:56 PM, michel freiha <span dir="ltr"><<a href="mailto:michofr@gmail.com" _fcksavedurl="mailto:michofr@gmail.com" target="_new">michofr@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">
<div dir="ltr">
<div>Hello Air,Hi,</div></div></blockquote>
<div> </div>
<div>I created an extension like ths:</div>
<div> </div>
<div><br>[442033553]<br>user=442033553<br>type=pusers<br>secret=1234<br>host=dynamic<br>context=users<br>nat=yes</div>
<div> </div>
<div>when calling the DID number from an extension registered on asterisk server everything looks fine...When dilaing the number fromPSTN number I'm still getting the the below erroe:</div>
<div> </div>
<div>[Sep 3 20:56:00] NOTICE[18440]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '442033553' rejected because extension not found.</div>
<div> </div>
<div>Do you think i should define a context to receive calls from outside the asterisk server?If yes do you have any context sample definition?</div>
<div> </div>
<div>Regards</div>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">
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<div><span id=""></span></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>
<div></div>
<div class="Wj3C7c">
<div class="gmail_quote">On Wed, Sep 3, 2008 at 11:36 PM, Igor Hernandez <span dir="ltr"><<a href="mailto:emistz@gmail.com" _fcksavedurl="mailto:emistz@gmail.com" target="_new">emistz@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">
<div>
<div></div>
<div>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/" _fcksavedurl="http://www.api-digital.com/" target="_new">http://www.api-digital.com</a> --<br>
><br>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona<br>> Register Now: <a href="http://www.astricon.net/" _fcksavedurl="http://www.astricon.net/" target="_new">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" _fcksavedurl="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_new">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/" _fcksavedurl="http://www.escapetel.com/" target="_new">http://www.escapetel.com</a><br>
<br>_______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" _fcksavedurl="http://www.api-digital.com/" target="_new">http://www.api-digital.com</a> --<br><br>AstriCon 2008 - September 22 - 25 Phoenix, Arizona<br>
Register Now: <a href="http://www.astricon.net/" _fcksavedurl="http://www.astricon.net/" target="_new">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" _fcksavedurl="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_new">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</blockquote></div><br></div></div></div></blockquote></div><br></div>
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<hr noshade="noshade" size="1"><pre>_______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" _fcksavedurl="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" _fcksavedurl="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" _fcksavedurl="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a></pre></div>
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