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/*
****************************************************************
* *
* Always Be Conferencing (ABC) *
* *
* Authors: chris @ Penguin PBX Solutions (PenguinPBX.com) *
* Version: 21n "Niners Shoulda Won #KeepCalm #Kelce Edition" *
* Release: 12 February 2024 *
* License: Creative Commons Zero v1.0 Universal (CC0 1.0) *
* *
* To the extent possible under law, the author(s) have *
* dedicated all copyright and related and neighboring rights *
* to this software to the public domain worldwide. This *
* software is distributed without any warranty. *
* *
* You should have received a copy of the CC0 Public Domain *
* Dedication along with this software. If not, then please *
* see: <http://creativecommons.org/publicdomain/zero/1.0/>. *
* *
****************************************************************
Howdy
=====
ABC starts your Conference Bridge(s) in ASTERISK(R)
by simultaneously dialing the other participants,
making it easier for both you and your Friends
to stay connected using audio and/or video
streaming over vast seas of internet.
ABC also provides various call notification options
and geolocation features that you can configure to
suit your needs -- potentially useful in areas such
as emergency calling scenarios, where the tendency
of the operator to hang on to the phone line might
nudge you to scrap the conferencing and start calling
this file "A"lmost instead of "A"lways Be Conferencing.
"A"nyhow... time to grab some water and coffee!
AND REMEMBER:
ABC IS YOUR FREE GIFT, AS-IS, WITH NO WARRANTY!!!
PREREQUISITE: ASTERISK
======================
ABC requires that you install ASTERISK, first!
Yes that means Linux!
$ sudo apt-get install asterisk
...used to do a pretty nice job on Debian GNU/Linux 10,
but times change, and that seems harder on Debian 12...
To help install Asterisk, you might consider another
offering from Penguin PBX, such as the pngnx23299 Ansible
Role on github -- it installs Asterisk 20 and FreePBX 17
and ABC on top of your existing Debian 12 system.
Or, you can get your free copy of Asterisk from:
https://downloads.asterisk.org/pub/telephony/asterisk/
Then check hashes and run something like:
$ ./configure
$ make -j4
$ sudo make install
Already running ASTERISK or derivatives like FreePBX ?
Great! As long as it is ASTERISK 13 or newer, then you
can probably get ABC up and start testing in under an hour.
Internally, ABC is tested most with ASTERISK 20 and the
PJSIP channel driver using Session Iniation Protocol.
For integration with FreePBX, please see below section titled:
"Quick Start mini-HOWTO demo for first time ABC users on FreePBX".
For the best IVR Demos, please install the Extra Sounds files for
ASTERISK if your distro does not do so (looking at you Debian 10 ;)
from:
https://downloads.asterisk.org/pub/telephony/sounds/
Integrating ABC with your local ASTERISK installation
=====================================================
ABC is almost entirely contained in the single file you are reading right now.
This file is written in ASTERISK Extension Language (AEL version 2) dial plan format.
This is different than standard CONF dial plans used in most PBX installs.
But they basically do the same thing ie. let you express your call flows.
Never used AEL? That's fine -- search this file for the acronym: AEL.
Or scroll to "PREREQUISITE: The ASTERISK Extension Language (AEL)".
That section goes step-by-step through configuring your AEL setup.
Don't panic; for most users, using the minimal defaults, the AEL parts
should be as easy as copying this AEL file and then making sure it loads.
The harder localization *work you need to do* is in the CONF files!
What do you need to do ?
You need to add glue in your "extensions.conf" or "extensions_custom.conf"
files that sticks in calls to ABC, in your main Asterisk configuration
directory (usually /etc/asterisk/ directory.) On your test bench, you can
try adding this line into your local test dial plan context:
include => pngnpbx-abc-easy-include-gps
That will activate the following ABC extensions for various location IVRs:
123*222*0 <-- IT admins: dial this first to set up crypto keys!
123*222*1
123*222*2*701 or 123*222*2*702 etc.
123*222
123#222
You can skip the splats and hashes by instead including:
include => pngnpbx-abc-easy-digits-only-include-gps
To get these things working:
123222 (that's "123 ABC" on the phone keypad -> redirects to 123*222*1)
1232220 (for IT crypto key initializition to encrypt location info)
1232222701 (to check an extension)
Reload and try that out with "dialplan reload" and "ael reload" on the
Asterisk CLI.
Several files in the /etc/asterisk/ directory can affect ABC:
1. abc.conf - You (c)(sh)ould make this file. See the Sample abc.conf below.
2. confbridge.conf - ABC needs at least one line at the top: [general]
3. indications.conf - ABC optionally uses one new pattern for silent ring.
4. voicemail.conf - No edits usually needed as ABC works with most setups.
See Examples 1-19 below for some integration ideas and proposed edits
to your dial plan. Generally you will only need minimal modifications
to get working with ABC.
History
-------
There are many use cases considered by ABC, improving with each release.
15 - initially tested on Asterisk 15
16a - moved to testing on Certified Asterisk 16
16c - add emergency calling nice-things-to-have
16i - add single codec video options
16l - add some Asterisk 13-15 backwards compatibility
20m - support Asterisk 13-20; add parallel paging
21n - support Asterisk 13-21; add CPX/CSX/NGX codes, more headers & clouds
ABC Use Case: Emergency Outbound Calls (EOC)
--------------------------------------------
ABC can be configured to very quickly notify others of your call attempts,
using several different methods, which can be mixed together as you wish:
* Conference Calls
* Phone Calls
* Emails
* Voicemails
* Texts
* Overhead Pages
* Strobe Lights
* External Processes
* Cloud APIs
* and more!
For SIP Call-type notifications, the caller ID info notify-ees see on phones
can show them your name; your phone number; an optional suffix on your caller
ID number (using CSX codes); your X/Y/Z co-ordinates including latitude/longitude
(GPS), Plus Codes, elevation, depth, room, floor, building number, and dozens
of other optional details; and the Remote End destination phone number you
originally dialed eg. 112, 911, 999, etc.
Of course, as this is Always Be Conferencing, the Conference Calls notification
method is perhaps the most interesting to consider...
When your Friends answer their phones, they will hear some short notes (A-B-C),
and (optionally) a Text-To-Speech (TTS) play back of your GPS/PLUSCODE
co-ordinates, and then be connected to talk with you and other Friends in your
"Yurt" conference room.
In Simple & Tin Can Modes, Friends will even be able to talk with you *before the
Remote End answers* -- very helpful if the Remote End *never answers* due to
technical/capacity failures ouside of your network! (What if the far away operators
are all busy? ABC can let you talk to local safety and security teams, instead of
waiting on hold!)
Also in Simple Mode, the conference will continue until all parties disconnect
-- even if the OriGinal caller (you the O.G.) hang up the phone / drop the call
before anybody else "really" answers and starts talking!
But wait, there's more! And more than one Conference Call room available...
Instead of Friends, or in addition to them, if you use Paladins, they will be
placed into a separate conference -- the "Palace" -- all by themselves. They
won't be able to talk with you -- like your Friends in the "Yurt" conference
-- but Paladins will be able to hear you and Remote End talking, and Paladins
can talk to each other to help co-ordinate the local response to your call.
Sometimes Paladins don't listen to the Yurt, instead, they enter the Palace
and talk amongst themselves -- but that is still triggered by you the O.G.
making a phone call (and fully configurable from within ABC dial plan API.)
Related, Trolls are in the Yurt with you the OG caller, the Remote End, and
any Friends; but Trolls can only listen and not speak to anyone. Also Trolls
only get two DTMF menu options (to change their listening volume up or down.)
Sometimes Trolls are not people but blinking lights or overhead speakers that
play pre-recorded/dynamic-text-to-speech back alert messages -- up to you.
For those not wishing to do much conferencing, but get voicemail/email/page
notifications when somebody makes an emergency call, and do so with minimal
impact to their existing dial plan, then check out Parallel Mode, as described
in Examples #13, #14, and #15.
In Parallel Mode of the EOC Use Case, you can share a list of locations
with your telco *in advance* then let ABC change the Caller ID number
on-the-fly at the start of the Emergency Call. This works with many
different types of SIP and PRI deployments. ABC provides methods to
choose Caller ID based on IP subnets, as well as supporting of dynamic allocation
from a pool of existing numbers.
Extended SIP headers such as X-ng911 are also configurable via abc.conf
ABC Use Case: Work From Home (WFH)
AKA Insta Teams Mode (ITM)
----------------------------------------
ABC can be configured in Insta Teams Mode, to help you reach out to your Team
without them having to remember all the conference call details themselves,
making it easier for you to start-up Yurts with your Friends.
With WFH/ITM, you text: "I'll give y'all a call tomorrow morning between 10
and 11." Then tomorrow you call your ABC phone number, authorize the call (PIN)
to start spooling out to your Team, and then join in talking with everyone who
accepts the invitation to your Yurt. Or, check-in more on important projects,
from the road, using just your phone, when the chat/email threads get too long.
There are several differences in this Use Case...
In contrast with emergency calls, in Teams Mode your Friends will *NOT* be
placed into the Yurt right away -- they will hear a short message from you and
then a message asking them to join via "press 1" keypad option. This helps
prevent your Yurt from getting stuck in somebody's cell phone voicemail system.
Also, there are no Paladins or Trolls in this mode. And Friends in Teams Mode
won't hear your GPS/PLUSCODE co-ordinates (like they do in emergency modes).
In-call menu options are a little different, too.
You might think of the Work From Home / Insta Teams Mode solution as a sort of
mini-town hall teleconference, complete with features like *9 for raising your
hand.
Sample abc.conf configuration file
==================================
This (optional) abc.conf file can go into your /etc/asterisk/ directory:
[general] ; Always Be Conferencing by Penguin PBX Solutions.
;
demomode=nope ; Demo mode uses extension numbers 7001-7008.
; There are also some named (not numbered) extensions
; that you might want to try eg. ABCtheOG.
;
keydir=/var/lib/asterisk/keys ; Where to store encryption key files used by ABC.
;
[eoc] ; Emergency Outbound Calling
;
locationheader=X-ng911 ; Optional extra PJSIP header to stuff with more
; NGX info eg. Room 1234, S Floor 5, Red Barn, etc.
;
route=PJSIP/MYTRUNK,60 ; Route tech / name, ring timeout.
route=PJSIP/MYTRUNK2,60,P,S ; Route tech / name, ring timeout, and caller ID prefix/change, and suffix.
route=PJSIP/MYNEWTRUNK,10 ; Up to 10 route variables - this one shows 10 second ring timeout.
route=SIP/MYOLDTRUNK,10 ; Even older technologies okay - these are cumulative variable definitions.
route=IAX/MUHIAX,10 ; See your respective config files for each technology
route=DAHDI/MUHZAPGROUP,120 ; eg. pjsip.conf, sip.conf, dahdi.conf, iax.conf, etc.
;
friend=PJSIP/ABCfriend1 ; Prefill who gets dialed into the conf call by default.
friend=PJSIP/ABCfriend2 ; Up to 10 friends.
;
troll=PJSIP/ABCtroll1 ; Up to 10 trolls.
troll=PJSIP/ABCtroll2 ;
;
paladin=PJSIP/ABCpaladin1 ; Up to 10 paladins.
paladin=PJSIP/ABCpaladin2 ;
;
paravm=ABCfriend1 ; Who gets notified of EOC calls (in Parallel mode.)
paravm=ABCfriend2 ; May be specified up to 10 times.
paravm=ABCtroll1 ; Be sure to include @context if not using 'default'.
paravm=ABCtroll2 ; These entries must exist in voicemail.conf file.
; Consider making new voicemail users in a new
; voicemail context just for EOC -- then you can
; specify a better 'emailsubject' and 'emailbody'
; to clearly represent that these are emergencies.
;
parapg=PJSIP/123 ; Send a Page (in Parallel mode.)
parapg=PJSIP/456 ; Each of these phones will hear call meta data,
parapg=PJSIP/789 ; not the actual audio stream.
;parapg=PJSIP/555 ; Up to 10 page-ees.
;
;pararw=#YOUR_CUSTOM_URI# ; Send a cURL (in Parallel mode.)
pararw=https://PenguinPBX.com/hiabc1
pararw=https://PenguinPBX.com/hiabc2
;pararw=#YOUR_CUSTOM_URI# ; Any additional arrows are fired in parallel.
;pararw=#ANOTHER_CUSTOM_URI# ; Up to 10 arrows.
;pararw=https://api.example.com/key/#YOUR_API_KEY#/hello-world
;pararw=https://api.example.com/key/#YOUR_API_KEY#/sms-to/3035551234/type-alert
;
;
[itm] ; Insta Teams Mode (WFH scenarios)
;
conferenceminutes=120 ; Maximum length for one conference.
monthlyminutes=600 ; Allowed cumulative conference minutes for the month.
vmcontext=default ; Voicemail context used to authenticate users.
numberstyle=us ; Default is 'international' for robo number speech.
TESTING
=======
ABC was first tested with PJSIP on ASTERISK certified version 16.8-cert3.
It should also work okay with other channel techs -- see DAHDI in Example #2.
And chan_sip should mostly work (see PJSIP_DIAL_CONTACTS and PJSIP_HEADERS for
exceptions.) ASTERISK version 13 also mostly works (except for Yurt Corners
in Work From Home Mode) but it uses some call files in /tmp and the System()
application to shuffle files around for asynchronous call spooling.
Extension 922 is used often in the ABC Examples -- see below -- although any
extension number (0-9a-zA-Z) should work. But you should test it yourself.
Really, you should test and verify everything in ABC that you intend to use!
If you are not comfortable with that, or even if you are, please consider talks
with your neighborhood free/libre/open source software folks for separate
support/warranty consulting services.
AGAIN: ABC IS YOUR FREE GIFT, AS-IS, WITH NO WARRANTY!!!
FEATURES
========
~ Multiple notifications at start of call - by vmail, email and/or phone calls.
~ Dynamic call back numbers - helps users on phones that lack dedicated DIDs.
~ Caller ID based on Subnet ID - easier remote office, VPN and VLAN support.
~ Update phone GPS/PLUSCODE/CSX/FRX with included visual-IVR - on-screen phone menus.
~ Encrypted local storage of location info - protects info when data at rest.
~ Dynamic Location Routing per RFC 6442 - full discussion included in this file.
~ Text-To-Speech of GPS/PLUSCODE and added information - ideas for emergencies.
~ Record each leg of the call in separate WAV files - consider all legal issues.
~ Automatically bridge in callees - "Simple Mode".
~ Wait for answer before bridging - "Tin Can Mode".
~ Prompt for key to confirm answer - "Insta Teams Mode".
~ Skip the conferencing and just notify - "Parallel Mode with Voicemail Notify".
~ Easy integration with ASTERISK and FreePBX installs - see mini-HOWTO below!
KNOWN BUGS
==========
~ Callers from the same extension number dialing the same destination
number will end up in the same call path. This could happen when users
dial 922 from their laptop soft phone at the same time as dialing 922
from their desktop phone. The recommended workaround is to use separate
extension numbers for each device registration.
~ DTMF pass through is not activated (except in Parallel Mode). Bug/Feature!
TODO LIST - EOC scenarios
=========================
~ Support telecommunications relay services via MESSAGE functions.
~ Pidgin example using SIMPLE driver to see pop-up notifications on PCs.
~ SMS support for Text-to-Remote end.
~ Friends option to start out muted - on a friend by friend basis,
then press 1 to un-mute at any time.
~ Multicast addresses for paging.
~ Add option 8 for "U"sers - to list out the roll of users in the conf.
~ Finish CLIPS codes for appending to Emergency Caller ID number and/or
dialed number - either way it is for parsing by the telco, to map the codes
they assign for each of your specific physical locations.
~ Add waiting rooms for Yurts to be called into by end users.
The admin should get a "knock-knock-knock" sound that somebody wants to join.
Admin gets 60 seconds to listen to the name and/or number and accept.
~ Always playback any Z-coordinates, in addition to X and Y.
~ Reminder function to check if user did not recently update their
GPS/PLUSCODE/CPX/CSX/FRX/NGX, to be placed just before a "Dial" by the user.
~ Most overhead paging does not actually use the Page() application in ASTERISK.
~ Disconnect Remote End when OG and Friends have left Yurt (using marked users.)
~ Extend Parallel Mode beyond just Voicemail Notifications.
~ Nosy Paladin audio feed from Yurt into Palace sounds terrible! Broadcast()?
~ Consider press 1 to accept call options on some recipients (like WFH mode) in
order to avoid remote cell phone voicemail systems and answering machines.
~ Asterisk added more Geolocation parts upstream. Integrate these with ABC.
TODO LIST - WFH scenarios
=========================
~ Support chat box via MESSAGE functions. Send SMS to users so they
can enter from their phones (eg. audio from call on bluetooth, but
full chat window on your smart phone.)
~ More notifications when recording starts/stops.
Such as when during the chat, admin starts recording,
play message to all on the call.
~ Automatically mute all new entrants after Yurt size exceeds X users.
~ Friendly instructions at the start for callee. "This is a YurtTel call.
Press *6 to unMute yourself." either before or after the temp greeting.
~ Single entry sound, when room is full and/or 60 seconds. "There are
X people in the room, press *8 to redial the Y people still not here."
(Automatic quorum detector -- threshold user selectable eg. 50% or 75%).
~ LEAVE SOUNDS! Option in abc.conf ?? Only when below_min_quorum thresh ?
~ Playback "everyone else has left the call" if you are only one remaining.
~ Allowing adding new phone numbers on-the-fly mid-conference.
TODO LIST - generally
=====================
~ Line up Test numbers with Example numbers.
~ Consider Answering Machine Detection app options on remote legs.
~ Example settings with WebRTC and SIP soft phones.
~ SIP interconnects to other conference engines via SIP or H323.
~ Update to use Broadcast() application introduced in ASTERISK 18. Big job!
~ Eventually app_abc ? Maybe but AEL is fun !
~ Look into HangupCauseClear() for mulitple re-Dial()s.
~ Look into ConfKick() to get everybody out when not in operator services mode.
~ Speech-to-text recognition with Speech...() apps.
Quick Start mini-HOWTO demo for first time ABC users on FreePBX
===============================================================
Up and running in 15 minutes!
This mini-HOWTO assumes all phones are using same channel technology.
This could be all chan_PJSIP (recommended), or all chan_SIP,
or all chan_DAHDI, or all chan_IAX, or others - depends on you.
After completing these steps, you should be able to dial several
new extensions from your desk phone.
1. Start out by test dialing these numbers from your desk phone:
123*222*0
123*222*1
123*222*2
123*222
123#222
123222 (that's "123 ABC" on the phone keypad)
922
933
You'll probably hear some sort of error message on most of
them. This is different for each install. Use your judgement.
You may need to disable some of the routes in your FreePBX
if they conflict with these numbers. If that is not what you want,
then please stop now and try one of the more advanced options
for FreePBX integration (see Example #5 elsewhere in this file.)
2. SSH into your PBX:
ssh muhpbx
3. Download copy of AEL file you are currently reading from GitHub to your PBX:
wget https://raw.githubusercontent.com/chrsmj/always-be-conferencing/master/extensions-pngnpbx-abc-v21n.ael2
4. Copy in the new ABC AEL file:
sudo cp extensions-pngnpbx-abc-v21n.ael2 /etc/asterisk/
5. Check if you already got some other AEL files active in your existing Asterisk install:
ls -last /etc/asterisk/*.ael*
If you only see this ABC AEL file, then jump ahead to Step 7A.
6. If you see other exisiting AEL files, then check the hashes:
md5sum /etc/asterisk/extensions*.ael*
The Asterisk sample AEL file "extensions.ael" MD5 hash value
from versions ~1.8 to 18+ is:
9e8a0e48880f732345eac9ed889b6115
If your hash matches that, great, you've got a default AEL file.
Proceed to Step 7A.
If your hash is not a match, then your AEL file is non-standard.
Either you or someone before changed that file. Possibly it was
configured to load an older version of ABC. You must figure it out
a little more before proceeding.
7. Allow Asterisk to load the new ABC file -- either replace or append
to the "extensions.ael" file, using 7A or 7B (but not both steps):
7A. Overwrite method
echo '#include "extensions-pngnpbx-abc-v21n.ael2"' | sudo tee /etc/asterisk/extensions.ael
7B. Append method
echo '#include "extensions-pngnpbx-abc-v21n.ael2"' | sudo tee -a /etc/asterisk/extensions.ael
8. Choose your integration method with FreePBX -- any or all of the following 8A-8G:
8A. GPS and other geolocation integration
echo -e '[from-internal-custom]\ninclude => pngnpbx-abc-easy-include-gps' | sudo tee -a /etc/asterisk/extensions_custom.conf
8B. Emergency Outbound Calling - 922 testing - TONGA WARNING: Do not enable this 922 just yet in TONGA and maybe more countries.
echo -e '[from-internal-custom]\nexten => 922,1,Goto(pngnpbx-abc-easy-goto-eoc,922,simple)' | sudo tee -a /etc/asterisk/extensions_custom.conf
8C. Emergency Outbound Calling - 933 more testing
echo -e '[from-internal-custom]\nexten => 933,1,Goto(pngnpbx-abc-easy-goto-eoc,933,tincan)' | sudo tee -a /etc/asterisk/extensions_custom.conf
8D. Emergency Outbound Calling - parallel notifications - US AND CANADA AND OTHERS WARNING: DO NOT CHANGE 911 ROUTING
UNTIL AFTER YOU COMPLETE GOOD TESTS WITH 922 AND 933. ONLY THEN DO YOUR 911 CALL TEST AT OFF-PEAK PSAP HOURS.
JUST BE NICE - DO NOT HANG UP ON THE TEST - TELL THEM IT IS A TEST AND CHECK YOUR LOCATION AND CALLER ID INFO!
echo -e '[from-internal-custom]\nexten => 911,1,Goto(pngnpbx-abc-easy-goto-eoc,911,parallel)' | sudo tee -a /etc/asterisk/extensions_custom.conf
8E. Allow full demo integration with FreePBX managed internal phones:
echo -e '[from-internal-custom]\ninclude => pngnpbx-abc-easy-include-demo' | sudo tee -a /etc/asterisk/extensions_custom.conf
8F. Allow video door bell demo:
echo -e '[from-internal-custom]\ninclude => pngnpbx-abc-easy-include-doorbell' | sudo tee -a /etc/asterisk/extensions_custom.conf
8G. Allow Insta Teams Mode:
echo -e '[from-internal-custom]\ninclude => pngnpbx-abc-easy-include-itm' | sudo tee -a /etc/asterisk/extensions_custom.conf
9. Double-check if permissions and just plain existence of files seems reasonable:
ls -last /etc/asterisk/extensions*.{conf,ael}*
10. Press "Apply Changes" button in FreePBX GUI.
If you don't see the button, then make a small change elsewhere in the GUI,
such as changing the name on a test extension, then save it and look again
for the "Apply Changes" button.
11. Go ahead and try dialing the numbers in Step 1 again.
All the numbers starting with 123 should do something.
The 922 and 933 (and 911) will only work if you configure
some additional settings in the "/etc/asterisk/abc.conf" file.
To silence the ring-back you hear from friend's phones, but not the
Remote End, you can run the following command on the ASTERISK CLI:
sirius*CLI> indication add us silent 0/15000
(Or you can make silent ring a permanent modification in indications_custom.conf)
To hear your GPS/PLUSCODE location, any participant can press * at any time
during the call. To stop the bot playback, any participant can press #.
See additional conference bridge menu options below - there are also
different options available, based on the nature of the call.
Thus ends the mini-HOWTO for integration with FreePBX.
See further below for several more Examples with FreePBX.
PREREQUISITE: The ASTERISK Extension Language (AEL)
===================================================
This is not your standard dial plan "/etc/asterisk/extensions.conf" file!
ABC is configured in the ASTERISK Extension Language (AEL). AEL syntax is
converted into standard CONF file format when loaded in to system memory
by ASTERISK. But you should not need to edit this ABC AEL file very often,
if ever. It only needs to be *included* into your dialplan.
Please copy this file into your "/etc/asterisk/" directory and set it to
read-only permission for the user/group your ASTERISK process runs as (usually
"root" or "asterisk".)
To confirm you have AEL working in your install of ASTERISK, you can run this
command on the ASTERISK CLI:
sirius*CLI> core show help ael
...which, if loaded properly, should list "ael reload" and "ael set debug".
Otherwise, if this is your first time using AEL, then you may need to manually
load the module, or take other steps. This may include rebuilding from source
code eg. selecting the AEL module when using "make menuconfig".
After confirming AEL is active and working, edit (or make) this file:
/etc/asterisk/extensions.ael
And add this line (with no spaces at the front):
#include "extensions-pngnpbx-abc-v21n.ael2"
...then save the file. Now you should be able to load the AEL config and view
the in-memory dial plan in CONF format:
sirius*CLI> ael reload
sirius*CLI> dialplan show pngnpbx-abc-dial
If that works, and you see a bunch of dial plan looking output, Congratulations!
You are now able to use ABC from standard dial plan in "extensions.conf" file
-- or in the "extensions_custom.conf" file if you are using FreePBX.
Simple Mode vs. Tin Can Mode vs. Insta Teams Mode vs. Parallel Notification Mode
================================================================================
If you choose Tin Can mode, then ABC will use Dial() instead of Originate() at
start of the call to the Remote End. This is to help satisfy requirements in
some situations where it is deemed desirable to only route the call and not
answer the caller via early bridging. (Particularly this is for emergency
calling situations.)
Please note the following major differences between Modes:
0. In Simple Mode, everyone can always talk to each other.
In Tin Can Mode, *on some phones*, Caller can talk to Friends before
Remote End answers (but it depends on the Friends' type of phone.)
In Teams Mode, Friends do not enter the Yurt until they press 1 to opt-in.
In Parallel Mode, there is no conferencing ;( only notifications.
1. In Simple Mode, everyone always gets access to DTMF menu options, because
they are always in the Yurt.
2. In Teams Mode, Friends must press 1 to join, then it is basically just
like Simple Mode.
3. In Tin Can Mode, Friends will not get access to DTMF menu options until the
Remote End answers. Then, everyone will get access, same as Simple Mode.
4. In Tin Can Mode, there may be a few more frames of feedback immediately
after the Remote End answers.
5. In Tin Can Mode ONLY, when caller hangs up before Remote End answers, then
the call will drop. (MAYBE TODO: Fix this in a future release.)
PRIVACY
=======
The handling of exact X/Y/Z co-ordinates of each phone presents multiple safety,
security, and privacy concerns. (See RFCs 3693, 6280, 6442 and 6772.)
ABC considers carefully each of the six basic Geopriv roles introduced in
RFC 3693 and later refined in RFC 6280 Section 3.2, to wit:
1. Target: ABC can be the Target, or Targets can be User-based or Device-based.
2. Device: ABC can be the Device, but more commonly it will be your SIP phone.
3. Rule Maker: ABC provides RM functions. (See Conference Bridge DTMF options.)
4. Location Generator: ABC can be the LG. (See DIY GPS IVR for users/admins.)
5. Location Server: ABC can perform LS. (See encrypted internal AstDB storage.)
6. Location Recipient: ABC considers almost all participants as potential LRs.
ABC tries to get the DL to the LRs by multiple automatic and manual methods.
With ABC as your LS, your personal location data is not *pre-registered* with
any 3rd-party remote/hosted/cloud/government/etc. services in large centralized
databases, with all of the associated privacy and data safety issues that can
entail. But new regulations, such as those covering finer detail refinements in
Dispatchable Location (DL) info, threaten to dramatically increase the size and
scope of these centralized LS databases, opening up previously unknown employee
where-abouts to 'manual ALI query' and other privacy attacks in the future.
In the author's opinion, too many SIP providers are complying with changing
regulatory requirements by (un)intentionally sacrificing liberty in exchange
for safety, acting as the LS when that is wholly un-necessary. While not
exactly *security theater*, the choice between exposing personal information
outside of your PBX/MLTS only at the time of an emergency vs. storing it all
of the time with the telephone company is not to be taken lightly if you value
the privacy of your users. (Do you even SSH or VPN bro ?)
ABC takes a more dynamic approach (Dynamic Location Routing) instead of the
old Registered Address system. ABC only releases your location details
from the local PBX system when it is needed. Until then, ABC can store your data
at rest in encrypted format on the local ASTERISK system's internal SQLite
database (AstDB) using a separate key file. This prevents bulk dumping of clear
text data via ASTERISK CLI commands like "database show".
By default, the key is located on the filesystem in the /var/lib/asterisk/keys/
directory. When you automate backup processes, then you may want to consider
changing these slightly, for example, to not include the AstDB in directory
/var/lib/asterisk/ with /var/lib/asterisk/keys/ directory in the same tarball.
(That would be like hiding the key under the door mat.)
Or, you might consider changing the location of the default key file used for
GPS/PLUSCODE data encryption (TODO: make this easier in a future ABC revision.)
However, should you lose the key file, then users will need to manually
re-update their GPS XYZ co-ordinates, using the included IVR as the LG. Until
they do, the PBX will not know where they are, and it will fail when trying to
do things like send those co-ordinates to the Remote End LR.
Steps to (re)make the key file:
sudo mkdir /var/lib/asterisk/keys
sudo touch /var/lib/asterisk/keys/pngnpbx-abc-key.txt
sudo chown asterisk.asterisk /var/lib/asterisk/keys/pngnpbx-abc-key.txt
Feel free to store 16 random alpha-numeric characters in that file,
otherwise, a somewhat random string will be used. You may also want
to check out the demo key generation by dialing 123*222*0.
Of course, you are free to forego the novel location features, and stick with
the Registered Address system offered by your telephone company. ABC is still
totally compatible with those types of legacy solutions.
LOGGING
=======
Call Detail Records will be saved in files based on whatever ABC account name
you specify ie. the account code will be set to this name on (almost) all
channels. In the examples (below) it would be "abctest.csv" in the same
directory as your "Master.csv" (assuming you are using CSV for CDRs.)
Optionally you can record the call (see example #2 below).
Look for the recordings in "/var/spool/asterisk/monitor/" directory.
The ABC account name will be used to prefix the files (one file per leg.)
CAREFUL when recording eg. legal issues, drive space, privacy, etc.
EXAMPLES
========
YOU MUST CHANGE #YOUR-TRUNK-s#, the "ABCTO" number, the extension numbers, and
choose valid Call-Back numbers, whenever copy-pasta'ing these examples !!!
But please feel free to copy the next ~540 lines directly into your dial plan,
appended somehow to your extensions.conf or extensions_custom.conf files.
;
; Example #1: Simple Mode
; -----------------------
; * PJSIP channel driver.
; * One Remote End.
; * One friend to bridge in.
; * One static call back number.
; * Remapping of 922 -> 933 for testing SIP provider address verification.
[from-internal-custom-abc-example-1]
exten => 922,1,Set(ex1stamp=${STRFTIME(,UTC,%F %H:%M:%S.%3q)} UTC) ; Ready ?
exten => 922,n,NoOp(placeholder) ; TBD
exten => 922,n,Log(VERBOSE,Always Be Conferencing - PenguinPBX.com) ; Please copy and change as you wish!
exten => 922,n,Set(ABCTO=933) ; CHANGE ME! Remapping of the dial to another number (if any.)
exten => 922,n,Set(path=PJSIP/${ABCTO}@#YOUR-TRUNK#) ; CHANGE ME! Endpoint section defined in your PJSIP configuration.
exten => 922,n,Set(callback=3035557729) ; CHANGE ME! Your call-back telephone number (probably 10 digits in the USA.)
exten => 922,n,Set(friend=PJSIP/1113) ; CHANGE ME! The local PJSIP endpoint to bridge into the call.
exten => 922,n,Set(abcshrtname=abctest) ; CHANGE ME! Use abctest as the conf account name.
exten => 922,n,Set(ringtime=120) ; MAYBE CHANGE ME! Rings Remote End / path for 2 minutes.
exten => 922,n,AELSub(pngnpbx-abc-simple,${abcshrtname}) ; Simple setup.
exten => 922,n,AELSub(pngnpbx-abc-path,1,${ringtime},${path}) ; Set the first (and only) call path to use on call to Remote End.
exten => 922,n,AELSub(pngnpbx-abc-remote,${callback},static) ; Originate out the path/trunk using your call-back number as the Caller ID number.
exten => 922,n,AELSub(pngnpbx-abc-friend,1,${friend}) ; Bridge in the other attendee automatically.
exten => 922,n,AELSub(pngnpbx-abc-howdy-yall) ; Enter the conference. No more dial plan processing occurs for caller in your CONF file.
;
; Example #2: Using members of Super Awesome Company
; --------------------------------------------------
; * Adds multiple remote paths.
; * Adds call recording.
; * Adds geolocation.
; * Adds additional friend phones.
; * Adds all 15 extension numbers 1101-1115 from basic-pbx sample
; pjsip.conf file as found in ASTERISK certified version 16.3-cert1
; from Dec 2019.
; * Assumes users are all inside of magical imaginary phone booths.
; * Slightly different formatting of exten lines, with 'same' but without '>'.
; (Does the same thing :)
[from-internal-custom-abc-example-2]
exten = 922,1,Set(ex2stamp=${STRFTIME(,UTC,%F %H:%M:%S.%3q)} UTC) ; Ready ?
same = n,NoOp(placeholder) ; TBD
same = n,Log(VERBOSE,Always Be Conferencing - PenguinPBX.com) ; Please copy and change as you wish!
same = n,Set(ABCTO=933) ; CHANGE ME! Remapping of the dial to another number (if any.)
same = n,Set(gpsurl=https://abc.penguinpbx.com/loc/) ; CHANGE ME! URL that will be included in Geolocation SIP header.
same = n,Set(path1=DAHDI/#YOUR-TRUNK-1#/${ABCTO}) ; CHANGE ME! Endpoint section defined in your DAHDI(R) configuration.
same = n,Set(path2=PJSIP/${ABCTO}@#YOUR-TRUNK-2#) ; CHANGE ME! A PJSIP endpoint section - used if first path is down/unreachable.
same = n,Set(path3=PJSIP/${ABCTO}@#YOUR-TRUNK-3#) ; CHANGE ME! Yet another endpoint section defined in your PJSIP configuration.
same = n,AELSub(pngnpbx-abc-init,abctest,record,3600,,announce,,,) ; Use abctest as the conf account name, save WAVs, limit to 1 hour, and announce GPS/PLUSCODE.
same = n,AELSub(pngnpbx-abc-path,1,10,${path1}) ; Set the first call path / trunk name to use on call to Remote End. Ring 10 seconds.
same = n,AELSub(pngnpbx-abc-path,2,60,${path2}) ; Set the second call path - used only on failover of first call path. Ring 1 minute.
same = n,AELSub(pngnpbx-abc-path,3,300,${path3}) ; Set the third call path - used only on failover of second call path. Ring 5 minutes.
same = n,NoOp(About to change flow based on caller ID number.)
same = n,Set(cleancid=${FILTER(0-9a-zA-Z,${CALLERID(num)})})
same = n,Set(gotdp=${DIALPLAN_EXISTS(${CONTEXT},${EXTEN},x${cleancid})})
same = n,GotoIf($[${gotdp}]?x${cleancid})
; C-Level - corner of Park
; These two phones are in the same 1m square open area.
same = n(x1107),NoOp()
same = n(x1111),NoOp()
same = n,AELSub(pngnpbx-abc-gps,,39.73981,-104.99006,5280,feet,${gpsurl}) ; Set Geolocation SIP header to URL and GPS co-ordinates #1.
same = n,AELSub(pngnpbx-abc-sa,144 W Colfax Ave,Denver,CO,80202) ; Street Address, city name, state abbreviation, zip code
same = n,AELSub(pngnpbx-abc-ai,NW corner of Park) ; Additional Information.
same = n,AELSub(pngnpbx-abc-remote,3035557729,static) ; Originate out the trunk(s).
same = n,Goto(muhfriends)
; Customer Experience - in Park but near Memorial
; These four other phones are in another nearby open area.
; With Different Additional Information.
same = n(x1101),NoOp()
same = n(x1102),NoOp()
same = n(x1115),NoOp()
same = n,AELSub(pngnpbx-abc-gps,,39.74007,-104.98912,5280,feet,${gpsurl})
same = n,AELSub(pngnpbx-abc-sa,144 W Colfax Ave,Denver,CO,80202)
same = n,AELSub(pngnpbx-abc-ai,SW of Voorhies Memorial)
same = n,AELSub(pngnpbx-abc-remote,3035557729,static)
same = n,Goto(muhfriends)
; Park Benches for Software Engineers
; Note this uses Plus Codes not "standard" latitude and longitude
same = n(x1110),NoOp()
same = n(x1104),NoOp()
same = n(x1113),NoOp()
same = n(x1114),NoOp()
same = n,AELSub(pngnpbx-abc-gps,85FQP2Q6+H7,,,5280,feet,${gpsurl})
same = n,AELSub(pngnpbx-abc-sa,Unnamed Road,Denver,CO,80202)
same = n,AELSub(pngnpbx-abc-ai,SW corner of Park)
same = n,AELSub(pngnpbx-abc-remote,3035557729,static)
same = n,Goto(muhfriends)
; Directors and all others sit together in the Ampitheater
; Again using Plus Codes - see https://plus.codes/ for more info
same = n(x1103),NoOp()
same = n(x1106),NoOp()
same = n(x1108),NoOp()
same = n,AELSub(pngnpbx-abc-gps,85FQP2Q6+7F,,,5280,feet,${gpsurl})
same = n,AELSub(pngnpbx-abc-sa,101 W 14th Ave,Denver,CO,80202)
same = n,AELSub(pngnpbx-abc-ai,Greek Ampitheater)
same = n,AELSub(pngnpbx-abc-remote,3035557729,static)
same = n,Goto(muhfriends)
; Remote office location in the Mountains for Sales
same = n(x1105),NoOp()
same = n(x1109),NoOp()
same = n(x1112),NoOp()
same = n,AELSub(pngnpbx-abc-gps,,39.588,-105.644,4350,meters,${gpsurl}) ; Set Geolocation SIP header to URL and GPS co-ordinates #2.
same = n,AELSub(pngnpbx-abc-sa,Mount Evans Ridge,Idaho Springs,CO,80452) ; Street Address, city name, state abbreviation, zip Code
same = n,AELSub(pngnpbx-abc-ai,Floor 42) ; Additional Information.
same = n,AELSub(pngnpbx-abc-remote,7205557729,static) ; Originate out the trunk(s) using a different call-back number as the Caller ID number.
same = n,Goto(muhfriends)
same = n(muhfriends),NoOp()
same = n,Log(VERBOSE,Remote dial started. Now adding friends to call.) ; Most critical is the remote call.
same = n,AELSub(pngnpbx-abc-friend,1,PJSIP/1113) ; Bridge in friend.
same = n,AELSub(pngnpbx-abc-friend,2,PJSIP/1111) ; Bridge in another friend.
same = n,AELSub(pngnpbx-abc-friend,3,PJSIP/1109) ; Bridge in yet another friend. (Up to 42 total participants by default.)
same = n,AELSub(pngnpbx-abc-howdy-yall) ; Enter the conference. No more dial plan processing occurs for caller.
;
; Example #3: Dynamic call back numbers for phones without DIDs
; -------------------------------------------------------------
; * Needs two contexts.
; * Adds nine Dynamic Call Back numbers.
; * Adds local call back option with 922*EXTENSION
; * Requires call backs be registered before first use
; by dialing 123*2220*NUMBER
[from-external-custom-abc-example-3]
exten = 3035551111,1,AELSub(pngnpbx-abc-reentry,1,${EXTEN},${abclo}) ; Place the dynamic call back DIDs here, one per line (copy/paste/edit.)
exten = 3035552222,1,AELSub(pngnpbx-abc-reentry,2,${EXTEN},${abclo}) ; Do not change anything except for the phone numbers and position number.
exten = 3035553333,1,AELSub(pngnpbx-abc-reentry,3,${EXTEN},${abclo}) ; You should not provision these DIDs anywhere else in your PBX.
exten = 3035554444,1,AELSub(pngnpbx-abc-reentry,4,${EXTEN},${abclo}) ; This is particularly useful for lobby/waiting area phones without call back numbers.
exten = 3035555555,1,AELSub(pngnpbx-abc-reentry,5,${EXTEN},${abclo}) ; The abclo variable is for loading new call backs into internal AstDB.
exten = 3035556666,1,AELSub(pngnpbx-abc-reentry,6,${EXTEN},${abclo})
exten = 3035557777,1,AELSub(pngnpbx-abc-reentry,7,${EXTEN},${abclo})
exten = 3035558888,1,AELSub(pngnpbx-abc-reentry,8,${EXTEN},${abclo})
exten = 3035559999,1,AELSub(pngnpbx-abc-reentry,9,${EXTEN},${abclo})
[from-internal-custom-abc-example-3]
exten = 922,1,Set(ex3stamp=${STRFTIME(,UTC,%F %H:%M:%S.%3q)} UTC)
same = n,NoOp(placeholder)
same = n,Log(VERBOSE,Always Be Conferencing - PenguinPBX.com) ; Please copy and change as you wish!
same = n,Set(ABCTO=933) ; CHANGE ME! Remapping of the dial to another number (if any.)
same = n,Set(path=PJSIP/${ABCTO}@#YOUR-TRUNK#) ; CHANGE ME! Endpoint section defined in your PJSIP configuration.
same = n,Set(defaultcallback=3035550000) ; CHANGE ME! Default call back number used if dynamic allocation fails.
same = n,Set(friend=PJSIP/1113) ; CHANGE ME! The local PJSIP endpoint to bridge into the call.
same = n,Set(abcshrtname=abctest) ; CHANGE ME! Use abctest as the conf account name.
same = n,Set(ringtime=120) ; MAYBE CHANGE ME! Rings Remote End / path for 2 minutes.
same = n,AELSub(pngnpbx-abc-simple,${abcshrtname}) ; Simple setup.
same = n,AELSub(pngnpbx-abc-path,1,${ringtime},${path}) ; Set the first (and only) call path to use on call to Remote End.
same = n,AELSub(pngnpbx-abc-remote,${defaultcallback},dynamic) ; Originate out the path/trunk using a dynamic call-back number as the Caller ID number.
same = n,AELSub(pngnpbx-abc-friend,1,${friend}) ; Bridge in the other attendee automatically.
same = n,AELSub(pngnpbx-abc-howdy-yall) ; Enter the conference. No more dial plan processing occurs for caller.
exten = _922*XXXX,1,NoOp(get back to conf locally) ; Useful if a friend falls off the wagon. They can get back on the bridge themselves.
exten = _922*XXXX/1113,n,NoOp() ; Check if they are your friend. Based on caller ID number.
exten = _922*XXXX/1107,s,NoOp() ; Didn't try to automatically bridge this friend but let them join later manually.
exten = _922*XXXX/1111,s,NoOp() ; Didn't try to automatically bridge this friend but let them join later manually.
exten = _922*XXXX/1103,s,NoOp() ; Didn't try to automatically bridge this friend but let them join later manually.
exten = _922*XXXX/1106,s,NoOp() ; Didn't try to automatically bridge this friend but let them join later manually.
exten = _922*XXXX,s,Hangup() ; No non-friends allowed.
exten = _922*XXXX,n,NoOp()
same = n,Set(abcconfid=${FILTER(0-9a-zA-Z,${EXTEN:4})})
same = n,Goto(pngnpbx-abc-bridge,${abcconfid},1)
exten = _123*2220*NXXNXXXXXX,1,NoOp(number loading) ; Dial this each time you change dynamic call back DIDs.
same = n,Set(abclo=yes) ; Or you could manually insert entries into AstDB.
same = n,Set(did=${CUT(EXTEN,*,2)}) ; But doing it this way on the phone is probably easier.
same = n,Goto(from-external-custom-abc-example-3,${did},1) ; Also it re-uses the other context so no double-typing of DIDs!
;
; Example #4: Tin Can Mode
; ------------------------
; * Using "Tin Can" mode for Dial to remote instead of Originate.
; * Please see the differences in Modes near the top of this file.
[from-internal-custom-abc-example-4]
exten = 922,1,Set(ex4stamp=${STRFTIME(,UTC,%F %H:%M:%S.%3q)} UTC)
same = n,NoOp(placeholder)
same = n,Log(VERBOSE,Always Be Conferencing - PenguinPBX.com) ; Please copy and change as you wish!
same = n,Set(ABCTO=933) ; CHANGE ME! Remapping of the dial to another number (if any.)
same = n,Set(ABCTRUNK=#YOUR-TRUNK#) ; CHANGE ME! Remapping of the dial to another number (if any.)
same = n,Set(ABCPATH=PJSIP/${ABCTO}@${ABCTRUNK}) ; CHANGE ME! Endpoint section defined in your PJSIP configuration.
same = n,Set(ABCCALLBACK=3035550000) ; CHANGE ME! Default call back number used if dynamic allocation fails.
same = n,Set(ABCFRIEND=PJSIP/1113) ; CHANGE ME! The local PJSIP endpoint to bridge into the call.
same = n,Set(ABCTYPENAME=abcex4) ; CHANGE ME! Use abcex4 as the conf account name.
same = n,Set(ABCRINGTIME=120) ; MAYBE CHANGE ME! Rings Remote End / path for 2 minutes.
same = n,AELSub(pngnpbx-abc-tincan,${ABCTYPENAME}) ; Tin Can mode. Uses Dial instead of Originate to the remote.
same = n,AELSub(pngnpbx-abc-path,1,${ABCRINGTIME},${ABCPATH}) ; Set the first (and only) call path to use on call to Remote End.
same = n,AELSub(pngnpbx-abc-remote,${ABCCALLBACK},static) ; In Tin Can mode, this spawns Originate to a ChanSpy dummy not the Remote End.
same = n,AELSub(pngnpbx-abc-friend,1,${ABCFRIEND}) ; Bridge in the other attendee automatically.
same = n,AELSub(pngnpbx-abc-howdy-yall) ; In Tin Can mode, this dials the remote and then enters the conference after they answer.
; IT DOES NOT IMMEDIATELY PUT THE CALLER INTO THE CONFERENCE BRIDGE.
; IT USES A STANDARD DIAL() TO DIRECTLY ROUTE TO THE REMOTE END.
; IF THE CALLER HANGS UP BEFORE THE REMOTE END ANSWERS IT WILL DROP ON BOTH SIDES.
;
; --------------------------------------------------
; Example #5: More advanced integration with FreePBX
; --------------------------------------------------
; * In Tin Can Mode (Again)
; * Integrates with FreePBX per-device emergency_cid numbers.
; * Copy this context into your /etc/asterisk/extensions_custom.conf file.
; * Just fill in the #XYZ# variables once for your entire system.
; * Then, make a new Custom Destination in FreePBX to point to:
;
; from-internal-custom-abc-example-5,s,1
;
; * Finally, make a new Misc Application in FreePBX to point an
; extension number to the new Custom Destination eg. 933
; * This example uses legacy chan_SIP instead of newer chan_PJSIP
;
[from-internal-custom-abc-example-5]
exten => s,1,Set(ex5stamp=${STRFTIME(,UTC,%F %H:%M:%S.%3q)} UTC)
exten => s,n,NoOp(placeholder)
exten => s,n,Log(VERBOSE,Always Be Conferencing - PenguinPBX.com)
exten => s,n,Set(cidnum=${FILTER(0-9A-Za-z,${CALLERID(num)})})
exten => s,n,Set(callback=${DB(DEVICE/${cidnum}/emergency_cid)})
exten => s,n,Set(ABCTO=${CALLERID(DNID)})
exten => s,n,Set(path1=SIP/${ABCTO}@#SIPTRUNK-1#) ; chan_sip support
exten => s,n,Set(path2=SIP/${ABCTO}@#SIPTRUNK-2#) ; multiple trunks