[App_rpt-users] Autopatch

[Jeff Lehman]

Oh awesome I will take a look at it and see what you have.

Don't bang your head on the GVSIP wall however. Sadly Google has stated
that asterisk connections to GVSIP are a violation of TOS and has turned
off GVSIP access to all but OBIHAI devices....

http://nerdvittles.com/?p=27274

I had 2 GVSIP trunks coming into the PBX and neither of them work today.

Jeff

On Fri, Nov 16, 2018 at 2:12 PM JJC <cummingsj at gmail.com> wrote:

> 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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-- 
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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