[App_rpt-users] Autopatch

[JJC]

Jeff - it's funny that you mention this... I have broken down here:
http://enhanced.github.io/2016/04/AllStar_AutoPatch how I got autopatch
working with a SIP service, which is what the freepbx trunk should look
like to ASL...

It's funny that you mention this because I'm hoping to sit down next week
and work through some of this just to get GVSIP working since that's
Googles new incantation of "SIP" and publish some documentation.

JJC
N0PKT

On Fri, Nov 16, 2018 at 12:05 PM Jeff Lehman <kc8qch at gmail.com> wrote:

> So since I am on a roll today, here's another to Chew on.
>
> I have Incredible PBX setup on a Pi here. I use it (at least till Google
> shut off the GVSIP service today) for my home phone as well as Allstar and
> Hamshack Hotline (See hamshackhotline.com for more info).
>
> I want to make one of my allstar nodes dial out through my
> PBX.....Incredible PBX is based on FreePBX and Asterisk sooo...
>
> I have calling from the PBX to ASL working fine. I can dial an extension
> from any of my phones and it connects to my ASL nodes.
>
> I am stumped on From ASL to PBX. Here is what I have done.....
>
> I followed the Directions on this page:
> http://docs.allstarlink.org/drupal/node/133
>
> So first I went into Rpt.conf and set those settings. I used "pstn-out"for
> the context in Rpt.conf for the autopatch. Next I went into extensions.conf
> and set the following in pstn-out:
>
> [pstn-out]
> exten => _XXX,1,Dial,SIP/pbx-autopatch/${Exten}
> exten => _XXX,2, congestion
>
> Note that XXX is for internal extensions. I use 3 digit extensions
> internally for phones. Next I went to sip.conf and set up the following
> stanza:
>
> [pbx-autopatch]
> type=peer
> allowguest=yes
> nat=yes
> autocreatepeer=yes
> insecure=port,invite
> auth=md5
> username=<the username for the extension i want it to dial out on>
> password=nunya
> host=<internal IP address for the PBX with port number>
> qualify=no
> disallow=all
> allow=ulaw
> allowexternaldomains=yes
> context=pstn-out
>
> When I dial *61 + internal extension (203 for example) it waits a few
> seconds then gives me fast busy
>
> When I watch on the asterisk CLI it tries to connect is not able to
> connect and at one point says "no such host: pbx-autopatch"
>
> (I would paste the log but I am not home to be able to copy and paste)
>
> Any initial thoughts? I've been playing with asterisk/freePBX/allstar for
> a little over a year now and I am confident with it, but this one has me
> stumped now...And it's tough to find documentation on this. or I am other
> thinking this and not able to wrap my head around it.
>
> Thanks much for letting me ramble.
>
>    Jeff
>
> --
> Jeff Lehman, KC8QCH
> E-mail: kc8qch at gmail.com
> http://kc8qch.dx.am
> Hamshack Hotline: 4218
>
> Webmaster
> Hamilton County ARPSC
> http://www.hamcoarpsc.org
> E-mail: hamcoarpsc at gmail.com
> Phone: 513-452-6480
>
> Allstar 47374 Administrator
> Project Manager, HHUL
> World Wide Amateur Radio Guild
> http://www.theguildglobal.org
> E-Mail: kc8qch at theguildglobal.net
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