[App_rpt-users] Newbie, autopatch help
Loren Tedford
lorentedford at gmail.com
Sat Jul 4 19:14:39 UTC 2015
How to setup auto patch from a secondary provider.. Hope this helps
http://www.kc9zhv.com/forums/topic/allstar-link-and-auto-patch/
Loren Tedford (KC9ZHV)
Email: lorentedford at gmail.com
Phone:
http://www.lorentedford.com
http://kc9zhv.com
Sent from Droid RAZR HD from Verizon wireless network
On Jul 4, 2015 10:58 AM, <mike at midnighteng.com> wrote:
>
> Unless you have made a change to "context" on the Phone Patch command line,
> you will be in "radio" context.
>
> You can place your dial strings there but...
> Understanding that by default, there is a bunch of stuff in "radio"
> context that you may not understand, it is a "trap" to keep anyone from
> dialing toll numbers. Normally, you would want to place your dial strings
> under/after what is listed there. But you probably need to look that over
> that you are not dialing a string that is being treated/matched as a
> invalid number.
>
> If you look closely at it, you will see that after toll number check, it
> later forwards to new context "autopatch" with the dialed extension. You
> could place your dial strings there.
>
> Keep a back-up copy for reference that you can attempt it again (if
> needed) when/as you understand more.
>
> As a test, you could make the dial string match the "first line in radio
> context" to see if all else is well.
>
> [radio]
> exten => _1NXXNXXXXXX,1,Dial,SIP/google/${EXTEN}
>
>
>
> ...mike/kb8jnm
>
>
>
> > Hi, I'm new to app_rpt (not much of a Linux guru)
> > With some help I've managed to get it setup and repeating and connected
> to
> > Allstar.
> >
> > I'm trying to get Autopatch working and followed instructions at
> > http://ohnosec.org/drupal/node/63
> >
> > I'm a little confused about the "context" autopatch mentioned as there is
> > already a context "radio" defined
> >
> > It looks like extensions.conf is routing call to pstn out, so I changed
> > the
> > routing under pstn-out and commented out what was there.
> > [pstn-out]
> > exten => _1NXXNXXXXXX,1,Dial,SIP/google/${EXTEN}
> > exten => _1NXXNXXXXXX,2,Congestion
> > ;exten=_NXXNXXXXXX,1,playback(ss-noservice)
> > ;exten=_NXXNXXXXXX,2,Congestion
> >
> > When I make a call I hear Call connected and immediately it says call
> > terminated
> >
> > I see this in debug
> > [Jul 4 07:39:03] WARNING[3986]: pbx.c:2470 __ast_pbx_run: Channel
> > 'Zap/pseudo-1
> > 738407999' sent
> into
> > invalid extension '5404342200' in context 'pstn-out', but n
> >
> > o invalid handler
> > I've defined my sip provider in the sip.conf file.
> >
> > 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.
> > I'd also like to setup reverse autopatch as well if someone can provide
> > some simple instructions for this.
> > _______________________________________________
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