<p dir="ltr">Barry's suggestion was fine.<br>
Can you follow that and try to call 500? Then provide all the kamailio-proxy.log of the call. Thanks.</p>
<p dir="ltr">Daniel<br>
</p>
<div class="quote">On 2 Nov 2014 11:12, Matthew Ogden <matthew@tenacit.net> wrote:<br type='attribution'><blockquote class="quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">ok, so months later, I finally needed this again. And set it up.<div><br></div><div>my point of failure is even though is that the call isn't routed somewhere, presumably its the rewrite rule that isn't working as intended.</div><div><br></div><div>If I look at the kamailio log I see this</div><div>/usr/sbin/kamailio[2238]: INFO: <script>: Setting P-Called-Party-ID '<sip:<null>@<null>>' - R=<null> <br></div><div>and</div><div><br></div><div>/usr/sbin/kamailio[2238]: INFO: <script>: Setting 'sip:<null>:<null>' taken from R-URI as next hop after lb for PSTN call - R=<null> <br></div><div><br></div><div><br></div><div>Of course in working call, that shows valid info.</div><div><br></div><div>I've tried putting my replacement string as 500@voicebox.local, and I've also tried the asterisk port, <a href="http://500@127.0.0.1:5070">500@127.0.0.1:5070</a>. </div><div><br></div><div>Both dont work for me.</div><div><br></div><div>Any ideas what I might be doing wrong?</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Aug 28, 2014 at 10:55 AM, Matthew Ogden <span dir="ltr"><<a href="mailto:matthew@tenacit.net" target="_blank">matthew@tenacit.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Neat!</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">I’ll try this out a bit later</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span></p><p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">From:</span></b><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> Barry Flanagan [mailto:<a href="mailto:barry@flanagan.ie" target="_blank">barry@flanagan.ie</a>] <br>
<b>Sent:</b> 28 August 2014 10:48 AM<br><b>To:</b> Matthew Ogden<br><b>Cc:</b> Jeremy Ward; Daniel Grotti; <a href="mailto:spce-user@lists.sipwise.com" target="_blank">spce-user@lists.sipwise.com</a><br><b>Subject:</b> Re: [Spce-user] voip loopback test</span></p>
<p class="MsoNormal"> </p><div><div><div><p class="MsoNormal">On 27 August 2014 22:24, Matthew Ogden <<a href="mailto:matthew@tenacit.net" target="_blank">matthew@tenacit.net</a>> wrote:</p></div><div><blockquote style="border:none;border-left:solid #cccccc 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Thanks Jeremy, that’s exactly what I mean!</span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Echo test.</span></p></div></div></blockquote><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">
You could quite easily do this on the Asterisk server included with spce. Just make a custom extensions.conf like so:</p></div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">cd /etc/ngcp-config/templates/etc/asterisk/</p>
</div><div><p class="MsoNormal">cp extentions.conf.tt2 extensions.conf.customtt.tt2</p></div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">Then edit extensions.conf.customtt.tt2 and add something like the following at the bottom of the file:</p>
</div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">### Echo Test Context ####</p></div><div><div><p class="MsoNormal">[echo-test]</p></div><div><p class="MsoNormal">exten => s,1,wait(2)</p></div><div>
<p class="MsoNormal">exten => s,n,Playback(demo-echotest) ; Let them know what's going on</p></div><div><p class="MsoNormal">exten => s,n,Answer()</p></div><div><p class="MsoNormal">exten => s,n,Echo() ; Do the echo test</p>
</div><div><p class="MsoNormal">exten => s,n,Playback(demo-echodone) ; Let them know it's over</p></div><div><p class="MsoNormal">exten => s,n,Hangup()</p></div></div><div><p class="MsoNormal">### End of Echo Test ###</p>
</div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">Then, in the existing [sip-in] context, at the top, put in an extension for the number you want to use for the echo test. For example, if you wanted to use 500 as the number someone would dial to do the echo test, then you would do:</p>
</div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">[sip-in] <-- NOTE: this line exists already - do not put it in, just look for it in the file!</p></div><div><p class="MsoNormal">exten => 500,1,Goto(echo-test,s,1)</p>
</div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">You then need to run 'ngcpcfg apply' so that Asterisk picks up the new config.</p></div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">
Then what you need to do in order to get SPCE to actually send this number - 500 - to the asterisk voicemail server, is to put a rule into your subscribers rewrite rule set, under "Inbound Rewrite Rules for Callee":</p>
</div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">Match Pattern: ^500$</p></div><div><p class="MsoNormal">Replace Pattern: <a href="mailto:500@voicebox.local" target="_blank">500@voicebox.local</a></p></div><div><p class="MsoNormal">
Description: Echo Test</p></div><div><p class="MsoNormal">Direction: Inbound</p></div><div><p class="MsoNormal">Field: Callee</p></div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">So now, any call from the subscriber to 500 will get re-written to <a href="mailto:500@voicebox.local" target="_blank">500@voicebox.local</a> and sent to the Asterisk server where it should get the echo test.</p>
</div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">Hope this helps.</p></div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">-Barry Flanagan</p></div><div><p class="MsoNormal"> </p></div><div>
<p class="MsoNormal"> </p></div><blockquote style="border:none;border-left:solid #cccccc 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm"><div><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span></p>
<p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">From:</span></b><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> Jeremy Ward [mailto:<a href="mailto:jward01@gmail.com" target="_blank">jward01@gmail.com</a>] <br>
<b>Sent:</b> 27 August 2014 09:20 PM<br><b>To:</b> Daniel Grotti<br><b>Cc:</b> <a href="mailto:spce-user@lists.sipwise.com" target="_blank">spce-user@lists.sipwise.com</a>; Matthew Ogden<br><b>Subject:</b> Re: [Spce-user] voip loopback test</span></p>
<p class="MsoNormal"> </p><p>Not to stick my nose in something that it doesn't belong, but I think he is referring to what asterisk calls an "echo test". </p><p>Its usually accessed via a star code (*31, if my memory serves me correctly).</p>
<p>Hope this helps.</p><p>Jeremy D. Ward, CWNE<br><a href="tel:%28954%29%20661-4965" value="+19546614965" target="_blank">(954) 661-4965</a></p><div><p class="MsoNormal">On Aug 27, 2014 1:01 PM, "Daniel Grotti" <<a href="mailto:dgrotti@sipwise.com" target="_blank">dgrotti@sipwise.com</a>> wrote:</p>
<blockquote style="border:none;border-left:solid #cccccc 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt"><p class="MsoNormal">Hi,<br>We don't have a specific number.<br>
It depends what you want to achieve, you can call the voicebox number for example or you can call another subscriber with a CF set to a mobile phone.<br>Again it depends what you mean with loopback test.<br><br>Daniel<br>
<br>On 27 Aug 2014 16:43, Matthew Ogden <<a href="mailto:matthew@tenacit.net" target="_blank">matthew@tenacit.net</a>> wrote:<br>><br>> Hi<br>><br>> <br>><br>> Does spce have a number I can dial built-in that is a loopback test?<br>
><br>> <br>><br>> Regards<br>><br>> <br>><br>> <br>><br>> Matthew Ogden<br>><br>> Management<br>><br>> TenacIT<br>><br>> <br>><br>> <br>><br>> <br>><br>
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