Hello.<br><br>On my Linksys SPA961 I had to change the dialplan string in order for **X to work.<br><br>Greetings,<br><br><div class="gmail_quote">On Fri, May 30, 2008 at 7:12 AM, Henrik Østergaard Madsen <<a href="mailto:Henrik@ostergaard.net">Henrik@ostergaard.net</a>> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div class="Ih2E3d"><br>
> Asterisk <a href="http://1.4.20.1" target="_blank">1.4.20.1</a><br>
> Polycom 501 v <a href="http://3.0.1.0032" target="_blank">3.0.1.0032</a><br>
><br>
> [May 30 00:12:54] NOTICE[26924]: chan_sip.c:14033<br>
> handle_request_invite: Call from 'SIP_PEER' to extension '**1'<br>
> rejected because extension not found.<br>
><br>
><br>
> [May 30 00:14:25] NOTICE[26924]: chan_sip.c:14033<br>
> handle_request_invite: Call from 'SIP_PEER' to extension '**'<br>
> rejected<br>
> because extension not found.<br>
><br>
> I did not even modify the dial plan on the phone, just press **1 and<br>
> dial. Make sure the configuration (dialplan) on the Linksys is<br>
> correct, see <a href="http://spc.pifiu.com" target="_blank">http://spc.pifiu.com</a> for details.<br>
><br>
><br>
<br>
</div>Interresting - I AM sure the dial plan does forward the call on the<br>
Linksys. The TA100 doesnt even have a dialplan - everything is<br>
forwarded raw - and it is the same issue there.<br>
<br>
But the bottom line of this must be that my Asterisk is somehow not<br>
correct. Either the configuration or the Asterisk itself. I cannot<br>
see why a configuration error should come with such a behavior<br>
(without showing something in the debug output). But updating to the<br>
latest Asterisk should not be a big thing.<br>
<br>
Thanks for the help, I will try that over the weekend.<br>
<font color="#888888"><br>
/Henrik<br>
</font><div><div></div><div class="Wj3C7c"><br>
<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Christian Svensson<br>Command Systems