<div dir="ltr">Thanks Loren, <div>That was a helpful config along with some help from Mike.</div><div><br></div><div>Just a tip for anyone looking for free sip calling. I am using <a href="http://gvsip.com">gvsip.com</a></div><div>You can use your google voice account as a sip account. Pay a one time $5 fee then have unlimited calling.</div><div><br></div><div>Does anyone have an example config for reverse autopatch? I haven't been able to find one yet.</div><div><br></div><div>Jason</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Sat, Jul 4, 2015 at 3:15 PM, Loren Tedford <span dir="ltr"><<a href="mailto:lorentedford@gmail.com" target="_blank">lorentedford@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr">Here is what I did</p><span class="">
<p dir="ltr"><a href="http://www.kc9zhv.com/forums/topic/allstar-link-and-auto-patch/" target="_blank">http://www.kc9zhv.com/forums/topic/allstar-link-and-auto-patch/</a></p>
<p dir="ltr">Loren Tedford (KC9ZHV) <br>
Email: <a href="mailto:lorentedford@gmail.com" target="_blank">lorentedford@gmail.com</a><br>
Phone: <br>
<a href="http://www.lorentedford.com" target="_blank">http://www.lorentedford.com</a><br>
<a href="http://kc9zhv.com" target="_blank">http://kc9zhv.com</a></p>
<p dir="ltr">Sent from Droid RAZR HD from Verizon wireless network</p>
</span><div class="gmail_quote"><div><div class="h5">On Jul 4, 2015 9:46 AM, "Jason Armentrout" <<a href="mailto:jason@atrs.com" target="_blank">jason@atrs.com</a>> wrote:<br type="attribution"></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5"><div dir="ltr">Hi, I'm new to app_rpt (not much of a Linux guru)<div>With some help I've managed to get it setup and repeating and connected to Allstar.</div><div><br></div><div>I'm trying to get Autopatch working and followed instructions at</div><div><a href="http://ohnosec.org/drupal/node/63" target="_blank">http://ohnosec.org/drupal/node/63</a><br></div><div><br></div><div>I'm a little confused about the "context" autopatch mentioned as there is already a context "radio" defined</div><div><br></div><div>It looks like extensions.conf is routing call to pstn out, so I changed the routing under pstn-out and commented out what was there.</div><div><div>[pstn-out]</div><div>exten => _1NXXNXXXXXX,1,Dial,SIP/google/${EXTEN} </div><div>exten => _1NXXNXXXXXX,2,Congestion </div><div>;exten=_NXXNXXXXXX,1,playback(ss-noservice)</div><div>;exten=_NXXNXXXXXX,2,Congestion</div></div><div><br></div><div>When I make a call I hear Call connected and immediately it says call terminated</div><div><br></div><div>I see this in debug</div><div><div>[Jul 4 07:39:03] WARNING[3986]: pbx.c:2470 __ast_pbx_run: Channel 'Zap/pseudo-1 738407999' sent into invalid extension '<a href="tel:5404342200" value="+15404342200" target="_blank">5404342200</a>' in context 'pstn-out', but n o invalid handler</div></div><div>I've defined my sip provider in the sip.conf file.</div><div><br></div><div>I know it is something simple I'm missing, but I'm not an Asterisk expert either and I don't fully understand how all the pieces work. </div><div>I'd also like to setup reverse autopatch as well if someone can provide some simple instructions for this.</div><div><br></div></div>
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