<p dir="ltr">I built a headless node (no radios) and I relay all outbound calls on our groups 6 private nodes through that one.</p>
<p dir="ltr">Inbound callers get an IVR sort of configuration that presents the caller with "please wait, someone will be with you shortly. .." unless you know the code to dial the proper node, it will disconnect you after a time telling you "I'm sorry, but no one is able to take your call right now". If you do have the right code, it puts you in receive mode and unless you now know the proper code again, you cannot transmit. </p>
<p dir="ltr">Besides the codes for dialing nodes, there are also codes for ARRL news, Newsline, weather, and building temperature at our repeater sites.</p>
<p dir="ltr">Asterisk is fun!</p>
<p dir="ltr">Sam.. K8SN</p>
<div class="gmail_quote">On Jan 19, 2016 23:57, "Doug Crompton" <<a href="mailto:doug@crompton.com">doug@crompton.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div dir="ltr"><font color="#000000" face="Tahoma,sans-serif">Bob,<br><br> The other thing I thought of which is something I do here is allow a local extension to call into a node. So a SIP phone or any phone on your local Asterisk could call an extension and be connected to allstar. In that case you would use the local extension number to dial into Allstar instead of the incoming DID. <br><br>So if you had Asterisk running your local phone system you could just pick up any phone, dial the designated extension and be connected to Allstar. I do it on two different boards and the Asterisks talk to each other via iax. <br></font><br><b><font style="font-size:16pt" size="4">73 Doug</font><font style="font-size:16pt" size="4"><br></font><font style="font-size:16pt" size="4">WA3DSP</font><font style="font-size:16pt" size="4"><br></font><font style="font-size:16pt" size="4"><a href="http://www.crompton.com/hamradio" target="_blank">http://www.crompton.com/hamradio</a></font></b><font style="font-size:16pt" size="4"><br></font><br><br><div><hr>From: <a href="mailto:w2ymm1@gmail.com" target="_blank">w2ymm1@gmail.com</a><br>Date: Tue, 19 Jan 2016 23:29:31 -0500<br>To: <a href="mailto:arm-allstar@hamvoip.org" target="_blank">arm-allstar@hamvoip.org</a><br>Subject: Re: [arm-allstar] Autopatch Inbound question<br><br><div>Thanks Doug. Once again I was just experimenting with the Asterisk functions and got ahead of myself. :-)</div><div>I do know about the built in allstar way, and it works great. I was just curious. </div><div><br></div><div>Thanks for taking the time to reply. </div><div><br></div><div>73. Bob W2YMM</div><div><br><br>Sent from my iPhone<div>Bob Prybyzerski</div><div><a href="tel:631-960-1051" value="+16319601051" target="_blank">631-960-1051</a></div></div><div><br>On Jan 19, 2016, at 10:34 PM, Doug Crompton <<a href="mailto:doug@crompton.com" target="_blank">doug@crompton.com</a>> wrote:<br><br></div><blockquote><div>
<div dir="ltr"><font color="#000000" face="Tahoma,sans-serif">Bob,<br><br> First of I would be careful about the legality of this. Anyone dialing that number, ham or not, would be connected to your node.<br><br>You do know there is already a public phone dial in for Allstar right? One where you put a pin number in and select a node? Your pin number is at the Allstarlink site. Technically to be legal you would have to do something similar. At least to verify it is you or someone you gave authority to who is a ham.<br><br>Patches in general have fallen out of vogue with cellphones.<br><br>This could be done in a dial plan.<br><br><a href="http://voip.ms" target="_blank">voip.ms</a> has examples at their site for caturing the incoming call. You would basically take that and add in what you want to do with the call. For the DID incoming context in sip.conf you would put a 'context=xxxx' as one of the statements and then in extensions.conf you would define that context. It would look roughly like this....<br><br>In sip.conf<br><br>[voipms]<br>canreinvite=no<br>context=dialin-radio<br>host=<a href="http://washington.voip.ms" target="_blank">washington.voip.ms</a> ; or your selected host<br>secret= ;your password<br>type=peer<br>username= ;your account<br>disallow=all<br>allow=ulaw<br>fromuser= ;your account<br>trustrpid=yes<br>sendrpid=yes<br>insecure=invite<br>nat=yes<br>qualify=2000<br><br>Add account and PW. Some of these commands may be different based on your system.<br><br>In extensions.conf -<br><br>[dialin-radio]<br>exten => xxxxxxxxxx,1,Rpt,node|S<br>exten => xxxxxxxxxx,n,Hangup <br><br>; where xxxxxxxxxx is your did number and node is the node on this server you want to connect to.<br><br>This is a guess for me as I have not done this and it always requires testing. Note this does not include any qualification of the caller which could be done in a more extensive dial plan.<br></font><br><b><font style="font-size:16pt" size="4">73 Doug</font><font style="font-size:16pt" size="4"><br></font><font style="font-size:16pt" size="4">WA3DSP</font><font style="font-size:16pt" size="4"><br></font><font style="font-size:16pt" size="4"><a href="http://www.crompton.com/hamradio" target="_blank">http://www.crompton.com/hamradio</a></font></b><font style="font-size:16pt" size="4"><br></font><br><br><div><hr>Date: Tue, 19 Jan 2016 21:19:17 -0500<br>From: <a href="mailto:w2ymm1@gmail.com" target="_blank">w2ymm1@gmail.com</a><br>To: <a href="mailto:arm-allstar@hamvoip.org" target="_blank">arm-allstar@hamvoip.org</a><br>Subject: [arm-allstar] Autopatch Inbound question<br><br><div dir="ltr"><div><div><div>Hi All, I saw some discussion a few weeks ago about Autopatch and some were using <a href="http://voip.ms" target="_blank">voip.ms</a> <br></div>I Currently have and Autopatch on one of my servers for Outbound calls only using a different service, but I am curious about setting up and receiving an Inbound Call on my 2nd node using <a href="http://voip.ms" target="_blank">voip.ms</a>.<br></div><div>I had no problem setting it up to make an outbound call, So I ordered a DID Number and have configured my sip.conf, extensions.conf. and rpt.conf.<br></div><div>I seem to be stuck on the proper way to setup the dialplan for [voipms-inbound]<br><br></div><div>Any chance anyone else has and Inbound autopatch example I could look at ?<br><br></div><div>Bob W2YMM<br></div><div><br></div><div><br></div></div><div><div><div><div><div><div><div dir="ltr">Robert Prybyzerski Jr<br>W2YMM<br><a href="tel:631-960-1051" value="+16319601051" target="_blank">631-960-1051</a><br><a href="http://www.w2ymm.com" target="_blank">http://www.w2ymm.com</a></div></div></div>
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