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; FreePBX dialplan to create a dialable feature code that will play audio back to the caller telling them if the | |
; pjsip signaling and rtp/srtp audio they are using is encrypted or not | |
; | |
; latest version: https://gist.github.com/lgaetz/3c2bc4709607fdde40a278f980dbe9b2 | |
; | |
; License: GNU GPL3 | |
; | |
; Usage - place the dialplan below in the file /etc/asterisk/extensions_custom.conf and reload the dialplan (apply config) | |
; from a local device dial 4511. The audio played will indicate if signaling and/or media is encrypted. | |
; |
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; FreePBX Dialplan to block robocalls | |
; | |
; License: GNU GPL3+ | |
; Latest version: https://gist.github.com/lgaetz/bdc261b0d12efafeb1eed64651c7d32b | |
; | |
; | |
; Usage: Place the dialplan below in /etc/asterisk/extensions_custom.conf | |
; change the context for the trunk(s) to from-pstn-captcha1 | |
; Apply Config | |
; |
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; FreePBX Dialplan to mimic the old Campon Feature in fpbx (Asterisk CCSS) | |
; | |
; License: GNU GPL3+ | |
; Latest version: https://gist.github.com/lgaetz/40636245b6846f3aab372b22eb310e40 | |
; | |
; | |
; The Goal: | |
; Watcher extension dials a prefix feature code and target extension. Once the target extension ends their current (or next) call, | |
; fpbx automatically sets up a call between the watcher and target extensions. Secondary goal, avoid AGI/ARI if possible | |
; |
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; FreePBX dialplan to schedule a call for future using Asterisk call file. | |
; | |
; License: GNU GPL3+ | |
; latest version: https://gist.github.com/lgaetz/ec3509da8d989a00064337f194b9d58e | |
; | |
; Asterisk diaplan sub requires 3 arguments | |
; ARG1 = dialstring for first leg of call | |
; ARG2 = dialstring for second leg of call | |
; ARG3 = how many minutes in the future to schedule the call | |
; |
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; FreePBX dialplan for a feature code that can be dialed from an idle device that will immediately bounce | |
; any active call from any of the the same user's extensions. Primary use case is when a user has multiple | |
; registered devices they can quickly move the active call between deskphone, to mobile client to desktop client. | |
; | |
; License: GNU GPL3+ | |
; latest version: https://gist.github.com/lgaetz/4c14829c14eb0a739a216016d9f394b2 | |
; | |
; version history 2024-03-22 First commit working | |
; 2024-03-24 Fix hint and change ext away from _. |
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<?php | |
*** *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** | |
* | |
* Script: lgaetz-click2call.php | |
* | |
* Latest version: https://gist.github.com/lgaetz/6ff9aeff59de40df5755c52d85a65c9b | |
* | |
* Usage: Scipt to list/toggle/set FreePBX Call Flow Controls. Developed for FreePBX 16 | |
* |
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[from-internal-custom] | |
; Dialplan for a feature code to add ; an IP address to the FreePBX trusted zone | |
; Dial F-I-R-E-W-A-L-L from an internal extension and enter an IPv4 address using | |
; DTMF with * as the dot character. | |
; | |
; License: GNU GPL3+ | |
; latest version: https://gist.github.com/lgaetz/8a043b7aa39484f622f0f7c138909558 | |
; | |
; version history 2023-01-21 First commit working |
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; FreePBX dialplan to create a dynamic voicemail blast group. | |
; Place this in /etc/asterisk/extensions_custom.conf and reload | |
; GNU GPL3+ | |
; latest version: https://gist.github.com/lgaetz/76006451910295a4625555361c9b5085 | |
; version history 2022-11-04 First commit | |
; 2022-11-05 Added recording playback to toggle | |
[from-internal-custom] | |
; Create feature code previx with hints. Each extension can enable/disable vm group membership by dialing the |
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#!/usr/bin/php | |
<?php | |
// script requires a command - echo help and exit if arg1 missing | |
if (!isset($argv[1])) { | |
echo " | |
*** *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** | |
* | |
* Script: lgaetz-cftoggle.php | |
* |
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#!/usr/bin/env php | |
<?php | |
/************************************************************************ | |
* callshaper-freepbx.php | |
* | |
* Latest version: https://gist.github.com/lgaetz/7e61c464588623662fcfdf1367e97c32 | |
* | |
* AGI script for adding Stir/Shaken signatures to outbound SIP calls from | |
* Freepbx. Modified for FreePBX from callshaper.php as published by tiltx.com | |
* ref: https://help.smartcarrier.io/portal/en/kb/articles/tiltx-call-shaper-module |
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