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@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@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. _______________________________________________ App_rpt-users mailing list App_rpt-users@ohnosec.org http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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