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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style="background-image:
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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>
<div class="PrivateMsgDiv"><!--{12205738117041}-->
<div dir="ltr"><br><br>
<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;">
<div dir="ltr">
<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>
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Arizona<br>
Register Now: <a href="http://www.astricon.net/"
_fcksavedurl="http://www.astricon.net/"
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</blockquote></div><br></div></div></div></blockquote></div><br></div>
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<hr noshade="noshade"
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