Newbie, autopatch help

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.

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
ohnosec.org - ohnosec Resources and Information.

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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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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“Unsubscribe or edit options button”

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confirmation. If you have trouble unsubscribing, please send a message to

the list detailing the problem.


App_rpt-users mailing list

App_rpt-users@ohnosec.org

http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

To unsubscribe from this list please visit http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users and scroll down to the bottom of the page. Enter your email address and press the “Unsubscribe or edit options button”

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Thanks Loren,
That was a helpful config along with some help from Mike.

Just a tip for anyone looking for free sip calling. I am using gvsip.com

You can use your google voice account as a sip account. Pay a one time $5 fee then have unlimited calling.

Does anyone have an example config for reverse autopatch? I haven’t been able to find one yet.

Jason

···

On Sat, Jul 4, 2015 at 3:15 PM, Loren Tedford lorentedford@gmail.com wrote:

Here is what I did

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 9:46 AM, “Jason Armentrout” jason@atrs.com wrote:

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

To unsubscribe from this list please visit http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users and scroll down to the bottom of the page. Enter your email address and press the “Unsubscribe or edit options button”

You do not need a password to unsubscribe, you can do it via email confirmation. If you have trouble unsubscribing, please send a message to the list detailing the problem.