PhreakNet Community Docs

Contributions

This is the most comprehensive documentation about using Asterisk for typical and atypical purposes, and this is an actively maintained and living resource. This documentation wasn't written overnight — quite the contrary, it's grown piece by piece since 2018. Community contributions are always appreciated, and if there's something you think should be documented here that isn't, let us know and we can add it.

This documentation is written by hobbyists, for hobbyists. If you notice a typo or something that's not quite right here, we'd love to fix it! If you find something missing here that you think should be present, we'd love to add it! Contributions are always welcome and are greatly appreciated. To make a contribution, please drop the editor a line at docs @  phreaknet.org (trim the spaces out please, before copying into your mail program). Alternately, post to the mailing list if you think the change warrants community discussion.

In general, you should create a new ticket on InterLinked Issues so we can keep track of any pending issues. Below are some steps for common contributions.

Contributing to the Docs

Please open a ticket on InterLinked Issues, the in-house issue tracker. The category should be "PhreakNet Docs".

You can either describe the update that is needed and include changes in the issue or provide a diff. In the latter case, you must create the diff on InterLinked Paste and link the paste to the issue. Make sure to set the expiration to at least a couple weeks or longer (the default is 24 hours).

Contributing to PhreakNet boilerplate config

1. Please open a ticket on InterLinked Issues, the in-house issue tracker. The category should be "PhreakNet" (not "Docs").
2. Describe what's wrong (e.g. syntax error) or what could be improved with the relevant boilerplate config file(s). If you are contributing a patch, continue, otherwise, stop here.
3. Updates to boilerplate config are twofold: the boilerplate config itself is updated, but patches for boilerplate config are also generated so that, as much as possible, people do not need to download the file again and start over. User-contributed patches will be run against the boilerplate config to generate the newer version.
4. Use PhreakScript to generate a PhreakPatch for the files that need to be changes, e.g. phreaknet genpatch. You will need to run this for each file that needs to be updated. This is an interactive utility, so follow the instructions provided for the process. Make sure you update the version number and date in the relevant file. DO NOT include a verbose description of the fix anywhere in the config file (e.g. 2021-07-15 JS: fixed blah-blah-blah). Instead, include this in the issue tracker and it will be included in the change log. This avoids cluttering the boilerplate configs themselves with unnecessary comments. Updating an individual context or subroutine with a last updated date is okay.
5. Save the patch file in a directory on your server, then upload these to a web directory (link to InterLinked Paste if necessary)
6. Link the patches to the issue
7. Once the patch is merged, a description of the bug fix will be included in the PhreakNet Docs change log.
8. Please submit a pull request to the appropriate GitHub repository.

Contributing to Asterisk or DAHDI

First, you should test your code (a lot) and make sure no regressions or unexpected behavior are encountered. Consider adding test cases using either the test suite or unit tests. Test suite cases are more flexible but unit tests are easier and more self-contained for single modules, especially functions that are expected to produce a certain output. Make sure to build Asterisk with development mode if using PhreakScript. You must compile with developer mode since the compilation requirements for development mode are stricter and will catch errors and warnings that a normal install will not.

Second, you should try to submit your changes upstream to Asterisk or DAHDI. No, seriously, we're not joking. If we think your change is likely to be accepted by Sangoma, we will reject it, unless you have tried and Sangoma has rejected it. As many changes should be submitted upstream to Sangoma as feasibly possible.

For the time being, DAHDI contributions may be done directly since the DAHDI project is not actively being maintained.

1. Please open a ticket on InterLinked Issues, the in-house issue tracker. The category should be "Asterisk" or "DAHDI" (for both DAHDI Linux and DAHDI Tools as appropriate). Describe the change and why it is relevant. Changes do not necessarily need to relate to PhreakNet or the needs of telephone collectors, provided it is likely to be useful to at least a few people.
2. You must provide a diff of changes from the current master branch of Asterisk or DAHDI Tools or the current next branch of DAHDI Linux (not master, since next is newer than master).

Contributing to PhreakScript

1. Please open a ticket on InterLinked Issues, the in-house issue tracker. The category should be "PhreakScript".
2. Describe the change that needs to be made with sufficient detail.
3. If the change is significant, you must post to the PhreakNet mailing list and describe the change, so people are given an opportunity to comment and provide feedback. Major changes will not be accepted without a) posting to the mailing list and b) providing at least 24-48 hours for comments. For bug fixes that do not introduce regressions or any minor changes, it is not necessary to post to the mailing list.
4. If you are contributing a patch, you should open a pull request on GitHub. Please make sure your commit message complies with the Asterisk commit guidelines (yes, we use the same ones!). Please wrap your commits at 72 characters, and use a JIRA-style directive and/or GitHub-directive to reference and close the linked InterLinked and/or GitHub issue(s) respectively, e.g.:

   app_foobar: Add bar to the foobar.
   
   Lorem ipsum dolor sit amet, consectetur adipiscing elit, sed do
   eiusmod tempor incididunt ut labore et dolore magna aliqua. Ut enim ad minim
   veniam, quis nostrud exercitation ullamco laboris nisi ut aliquip ex ea
   commodo consequat. Duis aute irure dolor in reprehenderit in voluptate velit
   esse cillum dolore eu fugiat nulla pariatur. Excepteur sint occaecat
   cupidatat non proident, sunt in culpa qui officia deserunt mollit anim id est
   laborum.
   
   PHREAKSCRIPT-123 #close
   Resolves: #45

   
   If there is a GitHub issue open for the issue (in addition to or instead of the InterLinked Issue), that should be referenced as well as above.

Dedication

This documentation is dedicated to the late Brian Clancy, who sadly passed away of terminal illnesses on 2020/02/13. Without Brian, not only would this documentation not exist, this network would not exist. Brian was a good friend to all and a kind and patient mentor from the beginning. Brian lives on as the voice of the network, reachable on network operator numbers, Coin Zone calls, TIME (231-2301), and as a conversation partner (549-3344). But his most important impact has been his influence on the countless individuals whom he has left behind. He will be forever remembered for his persistence, work ethic, and strong-spirited determination. Thank you, Brian, for all you have done!

Hardware

To operate your own node, you will need a server running Asterisk, a powerful but free VoIP/TDM/analog PBX/CO switch software solution. You don't need a powerful system to run Asterisk — many node operators use servers with 1GB of RAM or less — even 512MB will be sufficient. uLaw (pronounced mu-law or, frequently but erroneously, u-law), is the preferred codec in the United States for VoIP, as it is 64kbps, or PSTN quality. Consequently, you will also need an Internet connection (but you knew that already, didn't you?). Although you can operate a node even with a very low-speed Internet connection, it is undesirable to have a connection speed of under 512kbps. A T1 (1.544Mbps) will be sufficient, and any package of broadband Internet will be more than enough bandwidth for several VoIP calls.

Some Asterisk servers are hosted servers, meaning they are not physically located at the premises of their owners. Hosted servers can be a great option for those just getting started or those without decent Internet connections or the space and resources to operate an on-premises Asterisk server. If you go this route, do understand its limitations. First, we can't stress enough how important it is to choose a reputable host. We once used a HiFormance host on a plan that was (wait for it!) $10/year. Of course, you know what they say: if it sounds too good to be true, it probably is. In mid-December 2018, without warning, HiFormance servers stopped working without warning — owners were unable to SSH into the servers and the services on them went down. With some effort, it was discovered that HiFormance was closing its doors. Ironically, a post on the website informed its customers that HiFormance would be closing its doors and that data should be backed up, yet, HiFormance never emailed its customers and most people discovered this only after their servers stopped working. Two years later, in November 2020, our VPSCheap server stopped working for a few days due to "complications" that arose during a cluster migration. A week later, the server came back up, but it was a brand new server — all the data was gone. Bad VPS hosts are unfortunately quite common, and it may be worth paying a couple bucks more for a more stable and reliable solution.

From our experiences with multiple VPS providers, we've seen some of the things that work and don't work well. We recommend using Digital Ocean's $4 per month Droplet, which is very affordable and reliable. You can sign up using our referral link if you aren't already a member, and we'll both get some free credit (after you verify your account, you'll get $100 to spend on "Droplets", which are VPSes). Thanks for helping to keep our costs low, so we can work on more cool projects for you!

Note: If you do use Digital Ocean, you should add a reliable DNS resolver, as by default, they provide their own. An issue observed with these is they intermittently fail to resolve ddns.us (and possibly other) hostnames. This can be fixed by editing /etc/systemd/resolved.conf.d/DigitalOcean.conf and adding another resolver, like 1.1.1.1, to the end of the resolver list. This will provide an alternate that doesn't have this issue. Then run systemctl restart systemd-resolved, and you should be able to resolve these hostnames and call these numbers.

Operating your Asterisk server locally is also an option. Almost any old computer from this millennium will suffice, but your options are really quite flexible. Asterisk can run seamlessly on a Raspberry Pi, which is a compact, low-energy solution that works well for many node operators. While you may run into roadblocks with connecting channel banks and other equipment to a Pi, for those with just ATAs (analog telephone adapters), a Pi should be more than sufficient. Many Asterisk nodes are operated using Raspberry Pis.

Telephony Cards

If you want the premium analog experience, an ATA is not sufficient, you'll want something that can interoperate with the DAHDI channel driver in Asterisk (DAHDI consists of the drivers for telephony cards, and Asterisk uses these to control them). There are a few off-brand cards, but the majority of cards fall into two categories: Digium or Sangoma cards. Digium cards are those originally manufactured by Digium, and Sangoma cards are those originally manufactured by Sangoma, prior to Sangoma's acquisition of Digium. UNDER NO CIRCUMSTANCES WHATSOEVER SHOULD YOU PURCHASE OR OBTAIN A SANGOMA CARD. These are known to be very buggy and defective, due to underlying timing bugs in the wanpipe software which Sangoma cards require (but Digium cards do not). For example, dial pulsing will not work at all on T1 CAS channels using a Sangoma card. Digium cards do not have any of these issues and should work without any problems.

Both of these cards can be found readily on the used market. However, even if you have to pay a bit more for one, a Digium card is a much better investment. If you have PCI slots in your machine (as opposed to PCIe), you can find some very cheap Digium cards on the used market as well.

Software

Asterisk is a free open-source VoIP telephony toolkit that has "taken the telecom industry by storm". It is widely used in the telephony community, by professionals and hobbyists alike, and is a requirement for operating a node, no matter what hardware is being used.

To start with, you'll need a flavor of Linux (also free and open-source) in order to run Asterisk. We recommend Debian, which works well on a variety of platforms. Any Linux commands featured in this documentation will assume an instance of Debian, but if you use a different flavor of Linux, you can consult your operating system's documentation for the proper commands and syntax. In many cases, minor variations in syntax and keywords will be the key distinguishment.

As for Asterisk itself, we recommend using the latest LTS release, Asterisk 18. Unlike other software packages, Asterisk has a large number of versions circulating, a great number of which are currently in use. Systems running Asterisk 1.0 and 1.2, for instance, are plentiful on C*NET. For our purposes, we've not needed any legacy functionality found in these versions that the newer versions lack; however, if you choose to install the Project MF patches on your system, an older version may be required. However, due to the latest PhreakNet patches, most of this functionality is now available in modern Asterisk.

Mastery

A quick note before we start here. This documentation goes into a fair amount of detail on some things, but it's simply impossible to cover everything comprehensively. There are two good ways to explore further and learn about the various things you can do with Asterisk and how to use them:

• Asterisk Wiki — in particular, the command references for applications, functions, and modules
• PhreakNet Asterisk documentation
• The sample configs — For the configuration files, most (if not all) options are explained fairly comprehensively.

A frequently asked question is how one builds Asterisk expertise. It really is a skill that takes experience to come — that is a lot of doing it. One can spend years trying to master Asterisk and not come close to doing or even learning everything possible, so don't be overwhelmed by not understanding how everything fits together. You will always be finding better ways to do certain things, and often there is more than one way to skin a cat. So, instead, choose to focus on specific objectives and outcomes and then look at the documentation for the relevant resources (e.g. application, function, module, config file, etc.). This kind of curious and progressive learning will go a long way.

If you're trying to do something, chances are somebody else has or might want to as well. Asking questions on the list facilitates the proliferation of knowledge and sharing of techniques, examples, etc. and is encouraged. Don't be shy!

SSH

Most of this documentation assumes you are using Debian (if you're using a different Linux distro, you might need to use yum instead of apt for instance, and some of the package names may also be different). It also assumes you're using a headless/non-GUI installation. If you have a GUI, you'll probably want to get rid of it. You'll also want to make sure SSH is enabled so you can remotely manage your system. If you're using a VPS, this part is kind of done for you, but here is the general process, starting with using passwd root to set a strong password for the 'root' account. Some VPS hosts allow you to reset this if needed, but you shouldn't forget this. 'root' is equivalent to the built-in 'Administrator' account on Windows.

Helpful link: How to Install and Set Up Headless Linux

You may also wish to use SSH keys instead of a password login. Most SFTP clients, like FileZilla, also support this. If you're using Windows, you should use KiTTY instead of PuTTY. KiTTY is basically "PuTTY on steroids", and allows you to, among things that PuTTY cannot do, store passwords for saved sessions so that you don't need to enter them manually each time. You don't have to use KiTTY, but you should use an SSH client that has this feature. This gives you no excuse not to use a long and secure password (and length is paramount, not complexity!).

Now, run nano /etc/ssh/sshd_config.

Hit CTRL+W to search for PermitRootLogin. It probably won't be found, but we just want to make sure. If it is found, change no to yes. If it isn't found, go to the end of the file and add PermitRootLogin yes on a new line. Then use CTRL+X, y, ENTER to save changes and exit.

Finally, many commands that need to be run throughout the Docs need to be performed as root. If something isn't working, make sure that you either a) logged in as root, b) su to root or c) use sudo to run with root privileges. For general system monitoring and accessing the Asterisk CLI, the root account is not needed, depending on how you configure Asterisk.

If you are using FreeBSD, you will need to run the following in the root account to allow su'ing into the root account from a non-root account: pw usermod USERNAME -G wheel.

Installing Asterisk/DAHDI

There are several different ways to use Asterisk. Packages like FreePBX exist that offer a GUI (graphical user interface) for configuration, which runs on Asterisk behind the scenes. You can, of course, use "just Asterisk", which is commonly referred to as "vanilla Asterisk", and this is by far the best option and is highly recommended. While GUI-based options like FreePBX may make it easier to get started initially, they are not worth it in the long-run, as you will find yourself restricted in terms of what you can configure, as much of the customizability and flexibility of Asterisk disappears when you use a GUI like FreePBX. The rest of this documentation will assume you are using vanilla Asterisk as it is the only way to truly realize the full power of Asterisk. However, other systems are used not infrequently in the greater hobbyist community so there may be others who have done something you are trying to do.

timedatectl set-ntp on
timedatectl set-ntp true
timedatectl status
					

apt install snapd
snap install core
snap install hello-world
hello-world
snap install --classic certbot
ln -s /snap/bin/certbot /usr/bin/certbot
certbot certonly --apache
certbot renew --dry-run
					

# Install dependencies
apt -y install linux-headers-`uname -r` \
build-essential \
binutils-dev \
git \
bison \
flex \
default-libmysqlclient-dev \
freetds-dev \
libbluetooth-dev \
libcodec2-dev \
libcurl4-openssl-dev \
libedit-dev \
libfftw3-dev \
libghc-postgresql-simple-dev \
libgmime-3.0-dev \
libical-dev \
libjansson-dev \
libneon27-dev \
libosptk-dev \
libnewt-dev \
libradcli-dev \
librust-backtrace-dev \
libsndfile1-dev \
libspandsp-dev \
libspeexdsp-dev \
libsqlite0-dev \
libsqlite3-dev \
libssl-dev \
libunbound-dev \
libvorbis-dev \
libxml2-dev \
unixodbc-dev \
uuid-dev \
xmlstarlet \
libiksemel-dev \
libtolua-dev \
libsrtp2-dev \
libldap2-dev \
libsnmp-dev \
libpopt-dev 
					

apt-get dist-upgrade
apt-get install linux-headers-`uname -r`
cd /usr/src
wget http://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-current.tar.gz
wget http://downloads.asterisk.org/pub/telephony/dahdi-tools/dahdi-tools-current.tar.gz
tar -zxvf dahdi-linux-current.tar.gz
tar -zxvf dahdi-tools-current.tar.gz
cd /usr/src/dahdi-linux-3.4.0

# if running make results in an error about pci-aspm.h not existing:
nano include/dahdi/kernel.h
#undef CONFIG_PCI


make
					

rm -rf /usr/lib/asterisk/modules # optional - if upgrading Asterisk, wipe out all the modules in advance of the upgrade, to remove old modules that aren't in the new versions
cd /usr/src
wget https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18-current.tar.gz
tar -zxvf asterisk-20-current.tar.gz
cd asterisk-21.4.3
./contrib/scripts/install_prereq install
					

./contrib/scripts/get_mp3_source.sh
./configure --with-jansson-bundled
cp contrib/scripts/voicemailpwcheck.py /usr/local/bin
chmod +x /usr/local/bin/voicemailpwcheck.py
					

sed -i 's/TLSv1.2/TLSv1.0/g' /etc/ssl/openssl.cnf
sed -i 's/DEFAULT@SECLEVEL=2/DEFAULT@SECLEVEL=1/g' /etc/ssl/openssl.cnf
					

make menuselect
					

make # compile Asterisk. This is the longest step, if you are installing for the first time.
make samples # do not run this on upgrades or it will wipe out your config!
make install # actually install modules
make config # install init script
make install-logrotate # auto compress and rotate log files
					

noload => chan_sip.so
					

apt-get install sudo
adduser -c "Asterisk" asterisk
passwd asterisk
					

# Asterisk needs read access to certs for TLS:
# Allow group to open relevant folders
chmod -R 740 /etc/letsencrypt/live/
chmod -R 740 /etc/letsencrypt/archive/
# Make the relevant letsencrypt folders owned by said group.
chgrp -R asterisk /etc/letsencrypt/live
chgrp -R asterisk /etc/letsencrypt/archive

chown -R asterisk /etc/asterisk/ /usr/lib/asterisk /var/spool/asterisk/ /var/lib/asterisk/ /var/run/asterisk/ /var/log/asterisk /usr/sbin/asterisk
					

#
# Don't fork when running "safely"
#
ASTARGS="-U asterisk"
					

AST_USER="asterisk"
AST_GROUP="asterisk"
					

asterisk ALL=(ALL:ALL) NOPASSWD:/usr/sbin/iptables
					

#!/bin/sh
# WARNING: Do not use this to install modern versions of Asterisk. Use PhreakScript instead.
# This is provided only for reference only for installing obsolete versions of Asterisk on obsolete versions of Debian.
# Super Simple Asterisk Installer
# (C) PhreakNet 2022 https://phreaknet.org
# Install Asterisk 12 on Debian 8.11

RELEASE=asterisk-12.0.0

# nano /etc/apt/sources.list
# deb https://archive.debian.org/debian/ jessie contrib main non-free
apt-get update

wget https://downloads.asterisk.org/pub/telephony/asterisk/old-releases/$RELEASE.tar.gz
tar -zxvf $RELEASE.tar.gz
cd $RELEASE
apt-get install aptitude -y --force-yes
apt-get install build-essentials -y --force-yes
apt-get install libncurses-dev -y --force-yes
apt-get install uuid-dev -y --force-yes
./contrib/scripts/install_prereq install
./configure
make menuselect.makeopts

# Disable everything
menuselect/menuselect --disable-category MENUSELECT_ADDONS --disable-category MENUSELECT_APPS --disable-category MENUSELECT_CDR --disable-category MENUSELECT_CEL --disable-category MENUSELECT_CHANNELS --disable-category MENUSELECT_CODECS --disable-category MENUSELECT_FORMATS --disable-category MENUSELECT_FUNCS --disable-category MENUSELECT_PBX --disable-category MENUSELECT_RES --disable-category MENUSELECT_TESTS --disable-category MENUSELECT_CFLAGS --disable-category MENUSELECT_OPTS_app_voicemail --disable-category MENUSELECT_UTILS --disable-category MENUSELECT_AGIS --disable-category MENUSELECT_EMBED --disable-category MENUSELECT_CORE_SOUNDS --disable-category MENUSELECT_MOH --disable-category MENUSELECT_EXTRA_SOUNDS menuselect.makeopts

# Enable the bare essentials
menuselect/menuselect --enable LOADABLE_MODULES menuselect.makeopts
menuselect/menuselect --enable app_chanspy --enable app_confbridge --enable app_dial --enable app_dumpchan --enable app_originate --enable app_mixmonitor --enable app_playback --enable app_playtones --enable app_read --enable app_senddtmf --enable app_stack --enable app_verbose --enable app_adsiprog --enable app_getcpeid menuselect.makeopts
menuselect/menuselect --enable chan_bridge_media --enable chan_iax2 --enable chan_sip menuselect.makeopts
menuselect/menuselect --enable chan_a_mu --enable codec_ulaw menuselect.makeopts
menuselect/menuselect --enable format_wav --enable format_pcm menuselect.makeopts
menuselect/menuselect --enable func_callerid --enable func_curl --enable func_global --enable func_groupcount --enable func_module --enable func_volume menuselect.makeopts
menuselect/menuselect --enable pbx_config --enable pbx_spool menuselect.makeopts
menuselect/menuselect --enable res_adsi --enable res_convert --enable res_crypto --enable res_curl menuselect.makeopts
menuselect/menuselect --enable EXTRA-SOUNDS-EN-ULAW menuselect.makeopts

make -j2
make install
if [ ! -d /etc/asterisk ]; then
	make samples
fi
					

Reserving an Office Code

If you're not already a PhreakNet community member, you'll need to follow the Getting Started guide to get set up.

Self-service reservation of office codes can be done through the PhreakNet portal. If you need assistance or have questions, you can post to the list or call the Business Office at 811.

Below, we'll break down the components of the registration process:

• Thousand Block(s) — The NNX-X blocks you wish to reserve for trunking to and from the network.
• Exchange Name — To get with the spirit of 2L+5N dialing (as opposed to 7N dialing), we encourage you to pick out an exchange name for your assigned numbers. Unless you have a good reason not to, it should be a Bell System-approved exchange name. You may wish to consult a list of telephone exchange names.
• Hostname — The public FQDN (fully qualifiede domain name) or IP address of your Asterisk server. The default IAX2 port is 4569. If yours is not, specify it afterwards with a colon.
• Username — The username of the IAX2 user. If no username is explicitly specified, it defaults to the name of the section.
• Password — The IAX2 secret. This should be a long alphanumeric string. This is required for IAX2 encryption to function.

If you don't have a static IP address, you can use a DDNS (dynamic DNS) service to automatically have an A record updated whenever your IP address changes. PhreakNet doesn't offer any DDNS services, but our friends over at C*NET do. You'll get your own .ckts subdomain that you can then use for your C*NET, PhreakNet, and other routings.

Other important attributes that are defined for an IAX2 user:

• Authentication Mode Either none (default), plaintext (deprecated), md5, or rsa. MD5 and RSA should be used. For compatability reasons, RSA cannot be the primary authentication mode — MD5 should be used as the primary authentication mode.
• Inkeys These are the RSA public keys permitted for RSA authentication on an inbound call. For RSA authentication on PhreakNet, the dialplan manages these automatically.
• Context The dialplan context(s) to use for the IAX2 user. The last one is the default one if none is specified for an inbound call. Specifying multiple contexts allows a single IAX2 user to be used to access multiple dialplan contexts — very handy for users that use RSA authentication, since these have more moving parts and overhead.

In most cases, calls are peer to peer with all routing info contained in the dial string. A username, host, and password may be specified here.

Existing Dialplans

If you have an existing dialplan, it should be fairly easy to get set up, though some caveats should be noted.

If you are getting started with PhreakNet and are already an NPSTN member, the following should be noted:

• The verification.conf provided below is the most up-to-date and comprehensive copy, and can replace any existing suite of verification subroutines. It is not PhreakNet specific
• Any eXtensible dialplan code can be safely removed. SIPxNumToPeer is provided for backwards compatability in the provided extensions.conf
• The clli variable is not PhreakNet specific. If you are using it currently (e.g. for NPSTN), you should rename clli to npstnclli (NPSTN-specific clli) and use clli for your general clli.
• phreaknet.conf contains a duplicate [from-internal]. You'll want to set your [from-internal], if you have one, to the desired behavior.
• Most contexts have been named with a phreaknet- prefix, when PhreakNet specific, to add specifity and clarity and reduce potential naming conflicts.
• phreaknet-aux.conf contains mostly "library code" as the eXtensible dialplan did. Most things you'll need to edit in the boilerplate code are in phreaknet.conf for easy access.

Boilerplate Code

The sections "Initial Configuration" and "Common Contexts" contain individual contexts in a well-documented manner for those thirsty for the details. However, for your convenience, if you'd like to get up and running with little attention to the details, you can simply use our boilerplate code. Below, you'll find an essential copy of each of the vital Asterisk configuration files you'll need to get started. You will need to adjust some things, such as properly setting certain variables and defining numbers, and so forth, but the code is otherwise as plug and play as possible. Helpful comments are included to help beginners get started, learning, and up and running quickly — no need to fear Asterisk!

If you are getting started for the first time, backup the files below on your system. Then, replace them with the ones below and go through each file, customizing to the extent required as directed.

If you already have an Asterisk system, you'll want to review the contents of these files and merge their contents into your system, since everything you need to get started is included there.

• asterisk.conf — general Asterisk config file — the two important settings are verbose and timestamps. verbose should be at least 3, and timestamps should be yes.
• iax.conf — for IAX2 trunking, used for calls to and from the network
• sip.conf — note that SIP is deprecated and will eventually be removed from Asterisk. Migration to PJSIP is recommended.
• pjsip.conf — you should use PJSIP instead of SIP if possible (if you are using SIP endpoints). You only need SIP *OR* PJSIP.
• musiconhold.conf
• extensions.conf
• verification.conf — a separate dialplan file containing just the code for the verification contexts, to minimize the clutter in extensions.conf — put this in /etc/asterisk/dialplan
• verify.conf — configuration file for the app_verify module. See verification for more details.
• phreaknet.conf — a separate dialplan containing PhreakNet main code, to minimize clutter in extensions.conf — put this in /etc/asterisk/dialplan
• phreaknet-aux.conf — a separate dialplan containing PhreakNet helper code, to minimize clutter in extensions.conf — put this in /etc/asterisk/dialplan
• If you have a channel bank, you'll need to set up DAHDI as well (not to mention compile Asterisk with DAHDI). Unfortunately, that is currently beyond the scope of this documentation.

A command like this can be used to download a file and replace an existing one:

Audio Files

First, be sure to grab the Pat Fleet sound library for Asterisk, to replace the default Allison Smith prompts to the extent possible. Trust us — this is a major upgrade.

You will probably want at least a few audio files to get started — the basic call progress tones and intercepts. We encourage all node owners to add variety to the network by giving their nodes their own unique soundscapes. However, if you want to get up and running immediately, we've assembled a set of the basic audio files you will need. Download these files and place them in /var/lib/asterisk/sounds/en/custom/signal/, unless otherwise stated:

• dialtone.ulaw
• busy.ulaw
• reorder.ulaw
• ccad.ulaw
• acbn.ulaw
• ringback.ulaw — save in /var/lib/asterisk/moh/ringback/

These files are provided for convenience, but if everyone uses them, the network's going to sound pretty stagnant, so please consider finding your own unique audio sources. There are a few available at Telephone World, This Is A Recording (for intercepts). You can also scour Evan Doorbell recordings for something more your style if you're looking for something particular. For example, if you decide your switch is going to be a simulated "Number 5 Crossbar" switch, then you should look for #5XB sounds you can use (you can use Audacity to prepare the audio and export it as a WAV file to then convert with SoX).

Note that the above is not your one-stop shop for getting started! Be sure you have completed everything outlined in the "Pre-Requisites" section above.

The boilerplate code above is only a starting point. Many of the contexts individually laid out in this documentation, like the simulated signaling subroutines (e.g. MFer, SFer, dial pulser) are not included in the boilerplate configuration files above. Only the basics that every node really needs to fully participate in the network are included in the boilerplate code. Beyond the contents of the files above, you may wish to further add to your system using other contexts outlined in this documentation.

Encryption

A word about authentication and encryption:

By default, VoIP, even using IAX2, is not encrypted. Encryption is easy, and it's now included in the boilerplate iax.conf. No other changes are necessary and there are not compatability issues. Some will argue that MD5 is not very robust, but it's better than nothing.

A few notes about IAX2 encryption that are not obvious from looking at the official Asterisk IAX2 or encryption documentation:

• Although the Asterisk documentation claims it does, plain text authentication does not support encryption (note the difference!). Hence, authentication is a necessary but insufficient condition alone for IAX2 encryption.
• Encryption requires either MD5 or RSA authentication
• Plain Text and MD5 authentication use identical configurations and dial syntax. The difference is that MD5 uses a challenge/response rather than sending the password in plain text across the wire
• A password may be provided even if the dialed destination does not require one. That is perfectly OK. Obviously, if that password is not defined by the peer (and no password is), the call will proceed as usual and it will not be encrypted or authenticated. If the dialed endpoint happens to require authentication and that is, in fact, the correct password, the call will be encrypted.
• You can tell if IAX2 encryption is working by doing a packet capture in Wireshark. Do a packet capture of an unauthenticated/unencrypted call and then compare with a packet capture of an encrypted call. Import it into Wireshark: the unencrypted pcap will have lots of 7Es, 7Fs, and FFs (if the call was mostly silence); the encrypted call should appear to be mostly random.
• If you analyze a packet capture of an encrypted call in Wireshark, you may see G.723.1 in the first couple hundred packets. This is nonsense. Wireshark doesn't know the protocol of the packets, since the stream is encrypted.
• In the Asterisk console, a call is not encrypted if you see Accepting UNAUTHENTICATED call... when a call comes in. It must say Accepting AUTHENTICATED call...

[myrsaendpoint]
type=peer
host=iaxhost.example.com
username=iaxuser
auth=rsa
forceencryption=yes
outkey=privatekeyname
secret=iaxpassword
				

cd /var/lib/asterisk/keys/
openssl genrsa -des3 -out phreaknetrsa.key -passout pass:SOMESECRETPASSWORDHERE 2048
openssl rsa -in phreaknetrsa.key -pubout -passin pass:SOMESECRETPASSWORDHERE -out phreaknetrsa.pub
				

cd /var/lib/asterisk/keys/
astgenkey -q -n phreaknetrsa
cat phreaknetrsa.pub # upload your public key at https://portal.phreaknet.org/rsa
touch /etc/asterisk/iax-phreaknet-rsa-in.conf
touch /etc/asterisk/iax-phreaknet-rsa-out.conf
				

chown asterisk phreaknetrsa.key
chown asterisk /etc/asterisk/iax.conf
chown asterisk /etc/asterisk/iax-phreaknet*
				

module reload res_crypto
keys init
keys show
				

inkeys=phreaknetrsa ; colon-separated list of public keys to accept

				

phreaknet update
phreaknet trace
				

Initial Configuration

At this point, we will assume you have a working Asterisk server and have reached out to get a thousand block allocated to you. In the meantime, you can work on getting your exchange setup so that everything will be working by the time your exchange appears in the route table (which usually occurs in less than 1 business day).

A note about Asterisk directories is necessary at this point. For those of you using the regular (compiled from source) version of Asterisk on Debian Linux, directories of particular importance are noted below:

/etc/asterisk/ — where the Asterisk configuration files reside
/var/lib/asterisk/agi-bin/ — where Asterisk AGI programs go
/var/lib/asterisk/moh/ — where music-on-hold audio files reside
/var/lib/asterisk/sounds/en/custom/ — where your custom audio files go (most any audio besides MOH)
/var/log/asterisk/ — where the Asterisk logs are located
/var/spool/asterisk/ — where recordings and voicemails are created

These paths are the default ones, assuming you haven't changed them in the Asterisk configuration.

Additionally, the AstDB database (e.g. the underlying database used by DB, DBDelTree, DB_KEYS, DB_DELETE, etc.) is stored in /var/lib/asterisk/astdb.sqlite3. Be sure to back this up periodically if you are using it!

As you spend more time working Asterisk, you'll become more and more familiar with these directories.

To get started, we need to adjust some settings and configure some settings. The following configuration (.conf) files are all located in /etc/asterisk/. You will need to modify them. There are a few ways you can do this. You can opt for the "local modification" approach or the "direct modification" approach.

• Local Modification — this entails keeping a local copy of all your configuration files, essentially mirroring what is on the server. The principle is simple: perform all your edits locally, and do a one-way write to the server to copy your changes over. If you opt for this approach, you can use a text-editor of your choice, like Notepad++ on Windows, to make changes. If you go this route, you don't need to worry separately about backups, either, since what is on the server in production is just a copy of your local files. If you haven't modified a particular file before, you'll need to copy it over from the server, first.
  If you are using Notepad++, you can add a custom language package to Notepad++ to add dialplan color syntax highlighting.
• Direct Modification — this entails directly modifying the files on the server. Since there is no GUI, this involves using command-line based text editors, like nano or vim. If you are comfortable with doing complex text manipulation and editing in the terminal, you may find this to be an easier approach. If you go this route, you won't need to worry about version conflicts, but you will need to ensure your configuration files are being backed up (along with your other files, like audio files, etc.!)

Regardless of which method you are using, but particularly if you opt for the local modification approach, you will need an SFTP client, like FileZilla. If you use FileZilla, you will need to change the "Default Transfer Type" to binary. You can do this by going to Edit → Settings → Transfers → FTP: File Types → Default transfer type. Select "Binary" and uncheck the two "Treat files… as ASCII file(s)" at the bottom as well.

You will need to use an FTP client like FileZilla (but using the SFTP protocol) to transfer over your audio files as well as your .conf files each time you make a revision if you opt for the local modification editing approach. You can use your client's site manager to save your connection settings so you don't need to re-enter the connection information each time.

[options]
verbose = 3
timestamp = yes
				

[modules]
autoload=yes
				

noload => chan_skinny.so ; Don't load skinny (tcp port 2000)  
noload => chan_mgcp.so ; Don't load MGCP (udp port 2727)
noload => chan_unistim.so ; Don't load unistim (udp port 5000)
noload => chan_ooh323.so ; Don't load ooh323 (tcp port 1720)
noload => chan_323
noload => pbx_dundi.so ; Don't load dundi (udp port 4520)
				

[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0:16555
tos = cs3

[transport-tcp]
type = transport
protocol = tcp
bind = 0.0.0.0:16555
tos = cs3

; If you don't have TLS certificates and aren't using TLS, don't uncomment these. PJSIP will error out and not load.
;[transport-tls]
;type = transport
;protocol = tls
;bind = 0.0.0.0:16556
;tos = cs3
;cert_file = /etc/letsencrypt/live/example.com/fullchain.pem
;priv_key_file = /etc/letsencrypt/live/example.com/privkey.pem
;verify_server = no
;method = tlsv1
				

[lines-endpoint](!)
type = endpoint
disallow = all
allow = g722
allow = ulaw
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
direct_media = no
inband_progress = yes
tos_audio = ef
device_state_busy_at = 1
trust_id_outbound = no
trust_id_inbound = no
notify_early_inuse_ringing = yes
context = from-internal ; context in the dialplan in which this user originates a call
allow_subscribe = yes
subscribe_context = phreaknet-hints
				

[lines-aor](!)
type = aor
max_contacts = 1
qualify_frequency = 30
				

[lines-auth](!)
type = auth
				

; every user needs to have an AOR (address of record) section, auth section, and endpoint section. To minimize clutter, we use templates for each.
[DeskPhone1](lines-aor)

[DeskPhone1](lines-auth)
username = DeskPhone1
password = samplepasswordhere

[DeskPhone1](lines-endpoint)
callerid = "John Smith" <5552368> ; change the CNAM and caller ID of your line here
;media_encryption = sdes ; for SRTP (voice path), if your endpoint supports it. Make sure to connect to the TLS port so the signaling is encrypted, too.
auth = DeskPhone1
;outbound_auth = DeskPhone1
aors = DeskPhone1
;mailboxes = 2368@vmcontext ; for voicemail MWI
				

[outgoing-aor](!)
type = aor
maximum_expiration = 60
minimum_expiration = 60
default_expiration = 180

[outgoing-identify](!)
type = identify
endpoint = outgoing
match = 127.0.0.1

[outgoing-endpoint](!)
type = endpoint
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
direct_media = no
tos_audio = ef
disallow = all
allow = g722
allow = ulaw
language = en
aors = outgoing

[outgoing](outgoing-aor)
[outgoing](outgoing-identify)
[outgoing](outgoing-endpoint)
transport = transport-udp
				

# termtype: te
bchan=1-23
dchan=24

# Span 1: WCT13x/0 "Wildcard TE131/TE133 Card 0" (MASTER) ESF/B8ZS RED ClockSource
span=1,1,0,esf,b8zs
fxols=1
fxols=2
fxols=3
fxols=4
fxols=5
fxols=6
fxols=7
fxols=8
fxols=9
fxols=10
fxols=11
fxols=12
fxols=13
fxols=14
fxols=15
fxols=16
fxols=17
fxols=18
fxols=19
fxols=20
fxols=21
fxols=22
fxols=23
fxols=24
echocanceller=mg2,1-23

# Global data

loadzone        = us
defaultzone     = us

# Span 1: WCT13x/0 "Wildcard TE131/TE133 Card 0" (MASTER) ESF/B8ZS RED ClockSource
span=1,1,0,esf,b8zs
fxoks = 1-24
echocanceller=mg2,1-23

# Global data

loadzone        = us
defaultzone     = us

DAHDI Tools Version - 3.1.0

DAHDI Version: 3.1.0
Echo Canceller(s): HWEC
Configuration
======================

SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 01)
Channel 02: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 02)
Channel 03: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 03)
Channel 04: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 04)
Channel 05: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 05)
Channel 06: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 06)
Channel 07: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 07)
Channel 08: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 08)
Channel 09: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 09)
Channel 10: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 10)
Channel 11: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 11)
Channel 12: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 12)
Channel 13: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 13)
Channel 14: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 14)
Channel 15: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 15)
Channel 16: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 16)
Channel 17: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 17)
Channel 18: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 18)
Channel 19: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 19)
Channel 20: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 20)
Channel 21: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 21)
Channel 22: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 22)
Channel 23: FXO Loopstart (Default) (Echo Canceler: mg2) (Slaves: 23)
Channel 24: FXO Loopstart (Default) (Echo Canceler: none) (Slaves: 24)

24 channels to configure.

Setting echocan for channel 1 to mg2
Setting echocan for channel 2 to mg2
Setting echocan for channel 3 to mg2
Setting echocan for channel 4 to mg2
Setting echocan for channel 5 to mg2
Setting echocan for channel 6 to mg2
Setting echocan for channel 7 to mg2
Setting echocan for channel 8 to mg2
Setting echocan for channel 9 to mg2
Setting echocan for channel 10 to mg2
Setting echocan for channel 11 to mg2
Setting echocan for channel 12 to mg2
Setting echocan for channel 13 to mg2
Setting echocan for channel 14 to mg2
Setting echocan for channel 15 to mg2
Setting echocan for channel 16 to mg2
Setting echocan for channel 17 to mg2
Setting echocan for channel 18 to mg2
Setting echocan for channel 19 to mg2
Setting echocan for channel 20 to mg2
Setting echocan for channel 21 to mg2
Setting echocan for channel 22 to mg2
Setting echocan for channel 23 to mg2
Setting echocan for channel 24 to none

[Mar 25 20:17:06] WARNING[585]: chan_dahdi.c:7376 handle_alarms: Detected alarm on channel 1: Red Alarm
[Mar 25 20:17:06] WARNING[585]: chan_dahdi.c:7376 handle_alarms: Detected alarm on channel 2: Red Alarm
[Mar 25 20:17:06] WARNING[585]: chan_dahdi.c:7376 handle_alarms: Detected alarm on channel 3: Red Alarm
[Mar 25 20:17:06] WARNING[585]: chan_dahdi.c:7376 handle_alarms: Detected alarm on channel 4: Red Alarm
[Mar 25 20:17:06] WARNING[585]: chan_dahdi.c:7376 handle_alarms: Detected alarm on channel 5: Red Alarm
[Mar 25 20:17:06] WARNING[585]: chan_dahdi.c:7376 handle_alarms: Detected alarm on channel 6: Red Alarm
[Mar 25 20:17:06] WARNING[585]: chan_dahdi.c:7376 handle_alarms: Detected alarm on channel 7: Red Alarm
[Mar 25 20:17:06] WARNING[585]: chan_dahdi.c:12535 mkintf: Attempt to configure channel 7 with signaling FXO Loopstart ignored because it is already configured to be FXO Loopstart.
[Mar 25 20:17:06] WARNING[585]: chan_dahdi.c:7376 handle_alarms: Detected alarm on channel 7: Red Alarm

# Autogenerated by /usr/sbin/dahdi_span_assignments on Sun Sep 15 08:05:47 PM EDT 2024
# Map devices + local spans to span + base channel number

# Device: [] @Board_ID_Switch_4 /sys/devices/pci0000:00/0000:00:01.0/0000:04:00.0/0000:05:08.0/pci:0000:05:08.0
/sys/devices/pci0000:00/0000:00:01.0/0000:04:00.0/0000:05:08.0/pci:0000:05:08.0 1:4:73
/sys/devices/pci0000:00/0000:00:01.0/0000:04:00.0/0000:05:08.0/pci:0000:05:08.0 2:5:97
/sys/devices/pci0000:00/0000:00:01.0/0000:04:00.0/0000:05:08.0/pci:0000:05:08.0 3:3:49
/sys/devices/pci0000:00/0000:00:01.0/0000:04:00.0/0000:05:08.0/pci:0000:05:08.0 4:6:121

# Device: [] @PCI_Bus_08_Slot_09 /sys/devices/pci0000:00/0000:00:06.0/0000:07:00.0/0000:08:08.0/pci:0000:08:08.0
/sys/devices/pci0000:00/0000:00:06.0/0000:07:00.0/0000:08:08.0/pci:0000:08:08.0 1:2:25

# Device: [] @PCI_Express_Bus_14_Slot_09 /sys/devices/pci0000:00/0000:00:1c.6/0000:0d:00.0/0000:0e:08.0/pci:0000:0e:08.0
/sys/devices/pci0000:00/0000:00:1c.6/0000:0d:00.0/0000:0e:08.0/pci:0000:0e:08.0 1:1:1 
					

* Dial(DAHDI/pseudo[/extension])
* Dial(DAHDI/[c|r|d][/extension])
* Dial(DAHDI/(g|G|r|R)[c|r|d][/extension])
*
* g - channel group allocation search forward
* G - channel group allocation search backward
* r - channel group allocation round robin search forward
* R - channel group allocation round robin search backward
*
* c - Wait for DTMF digit to confirm answer
* r - Set distintive ring cadance number
* d - Force bearer capability for ISDN/SS7 call to digital.

	[ringback]
	mode=files
	directory=moh/ringback/rbs1
				

	[default]
	mode=files
	directory=moh/silence
	random = no

	[asteriskdefault]
	mode=files
	directory=moh
	random=yes
				

Before continuing, you may wish to reload or restart Asterisk. To restart Asterisk entirely, type core restart now at the Asterisk CLI. To just reload a particular module, you can generally type the module name followed by the reload, such as sip reload. To reload music-on-hold, for example, you would type moh reload. To reload voicemail, you would type voicemail reload. To reload the dialplan (which is something you'll end up doing quite frequently, you'll type dialplan reload. To reload all modules without restarting Asterisk, you can simply type reload.

Occasionally, you may need to reboot your server as well. To do this, from the Asterisk CLI, type exit to quit the Asterisk CLI and exit to the Linux command line. Then type reboot. That's it!

Playback() has a root of /var/lib/asterisk/sounds/en/ so you only need to specify directories downstream from that point. Of course, adjust the Playback() path to that of your audio files. You will need to use SoX to convert WAV recordings of your switch tones to μLaw files; refer to the SoX section in Appendix A for more information.

Notice that the file extension (.ulaw) is not specified. Asterisk will search for the best audio file for the call and automatically select that one. If there is only one file with that name (excluding the file extension), then you can be assured it will be the ulaw file.

Hopefully all the audio files you need are on your Asterisk switch by now. Once you've modified your files (and copied them over to the server if you modified them locally), be sure to enter dialplan reload at the Asterisk CLI prompt. Take care that you don't see a WARNING message when you hit Enter; if you do, debug the error before continuing.

If you end up modifying a lot of different config files for different modules, reload will reload everything in Asterisk. Generally, it's better to just reload the specific thing you need (like dialplan) since reloading everything will take a little while (by this, we mean up to 10 seconds, rather than instantly).

Finally, some (few) things may require a restart: core restart now. Once Asterisk exits, enter asterisk -r to enter back into the Asterisk CLI.

(0|[2-9]11|11[2-9]|660S0|95[89]|10[2-9]xx|100xx|101xxxx|[2-9][2-9]xxxxx|1[2-9][2-9]xxxxx)
				

{ 0|[2-9]11|11[2-9]|660|95[89]|10[2-9]xx|100xx|101xxxx|[2-9][2-9]xxxxx|1[2-9][2-9]xxxxx}
				

(<:20>S0)
				

# TLS parameters
smtpd_tls_cert_file=/etc/letsencrypt/live/example.com/fullchain.pem
smtpd_tls_key_file=/etc/letsencrypt/live/example.com/privkey.pem
smtpd_tls_security_level=encrypt
smtp_use_tls=yes
smtpd_tls_session_cache_database = btree:${data_directory}/smtpd_scache
smtp_tls_session_cache_database = btree:${data_directory}/smtp_scache

phreaknet update
phreaknet trace
				

[Oct 23 20:08:52] DEBUG[24501][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 1
[Oct 23 20:08:52] DEBUG[24501][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:52] DEBUG[24499][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:52] DEBUG[24501][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 1
[Oct 23 20:08:52] DEBUG[24499][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 2
[Oct 23 20:08:52] DEBUG[24501][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:52] DEBUG[24499][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead
[Oct 23 20:08:52] DEBUG[24501][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 1
[Oct 23 20:08:52] DEBUG[24499][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 2
[Oct 23 20:08:52] DEBUG[24501][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:52] DEBUG[24501][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 1
[Oct 23 20:08:52] DEBUG[24499][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:52] DEBUG[24499][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 2
[Oct 23 20:08:52] DEBUG[24501][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:52] DEBUG[24501][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 1
[Oct 23 20:08:52] DEBUG[24499][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:52] DEBUG[24499][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 2
[Oct 23 20:08:52] DEBUG[24501][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:52] DEBUG[24501][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 1
[Oct 23 20:08:52] DEBUG[24499][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:52] DEBUG[24499][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 2
[Oct 23 20:08:52] DEBUG[24501][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:52] DEBUG[24501][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 1
[Oct 23 20:08:52] DEBUG[24499][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:52] DEBUG[24499][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 2
[Oct 23 20:08:52] DEBUG[24501][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:53] DEBUG[24501][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 1
[Oct 23 20:08:53] DEBUG[24499][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 2
[Oct 23 20:08:53] DEBUG[24501][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:53] DEBUG[24499][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:53] DEBUG[24501][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 1
[Oct 23 20:08:53] DEBUG[24499][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 2
[Oct 23 20:08:53] DEBUG[24501][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:53] DEBUG[24499][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:53] DEBUG[24501][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 1
[Oct 23 20:08:53] DEBUG[24499][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 2
[Oct 23 20:08:53] DEBUG[24501][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:53] DEBUG[24501][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 1
[Oct 23 20:08:53] DEBUG[24499][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:53] DEBUG[24499][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 2
[Oct 23 20:08:53] DEBUG[24501][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[Oct 23 20:08:53] DEBUG[24501][C-00000028]: chan_dahdi.c:9047 my_dahdi_write: Write returned -1 (Unknown error 500) on channel 1
[Oct 23 20:08:53] DEBUG[24499][C-00000028]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
				

# lspci -vb | grep -B 3 -A 5 "IRQ 11"

00:1a.0 USB controller: Intel Corporation 82801I (ICH9 Family) USB UHCI Controller #4 (rev 02) (prog-if 00 [UHCI])
        Subsystem: Super Micro Computer Inc 82801I (ICH9 Family) USB UHCI Controller
        Flags: bus master, medium devsel, latency 0, IRQ 11
        I/O ports at cc00
        Capabilities: [50] PCI Advanced Features
        Kernel driver in use: uhci_hcd
        Kernel modules: uhci_hcd

--
        Kernel driver in use: pcieport

00:1c.5 PCI bridge: Intel Corporation 82801I (ICH9 Family) PCI Express Port 6 (rev 02) (prog-if 00 [Normal decode])
        Flags: bus master, fast devsel, latency 0, IRQ 11
        Bus: primary=00, secondary=03, subordinate=03, sec-latency=0
        I/O behind bridge: 0000e000-0000efff
        Memory behind bridge: fea00000-feafffff
        Prefetchable memory behind bridge: 0000000400400000-00000004005fffff
        Capabilities: [40] Express Root Port (Slot+), MSI 00
--

01:00.0 Network controller: Digium, Inc. Wildcard TE133 single-span T1/E1/J1 card (PCI Express) (rev 02)
        Physical Slot: 0
        Flags: bus master, fast devsel, latency 0, IRQ 11
        Memory at fe8e0000 (32-bit, non-prefetchable)
        Capabilities: [40] Power Management version 3
        Capabilities: [48] MSI: Enable+ Count=1/1 Maskable- 64bit+
        Capabilities: [58] Express Endpoint, MSI 00
        Kernel driver in use: wcte13xp
--
02:00.0 Ethernet controller: Intel Corporation 82574L Gigabit Network Connection
        Subsystem: Super Micro Computer Inc 82574L Gigabit Network Connection
        Physical Slot: 0-1
        Flags: bus master, fast devsel, latency 0, IRQ 11
        Memory at fe9e0000 (32-bit, non-prefetchable)
        I/O ports at dc00
        Memory at fe9dc000 (32-bit, non-prefetchable)
        Capabilities: [c8] Power Management version 2
        Capabilities: [d0] MSI: Enable- Count=1/1 Maskable- 64bit+

APIs

PhreakNet provides a series of APIs for member usage. All APIs use the GET method. Overall usage is counted but PhreakNet API calls are not metered.

If there's an API that you'd like to see that doesn't exist yet, post to the list to discuss. Maybe others were looking for the same thing.

Standard Variables

PhreakNet uses a number of standardized dialplan and IAX variables. Their names and purposes are described below:

By their very nature, global variables are static.

Variables will be added as they are needed for network features. All channel variables listed above should be rewritten if calls are regenerated in order to maintain proper channel state (if you are not doing complicated things in your dialplan using ConfBridge(), you probably don't need to worry about this).

Answer Supervision

All nodes should aim for 100% correct supervisory status at all times for network compliance purposes. This includes supervising on billable/chargeable calls and not supervising when it is not appropriate (e.g. test numbers, intercept recordings, operator services, etc.).

The most common way to avoid providing false supervision when it is not desired is by using the noanswer flag for Playback() when needed, as follows: same => n,Playback(myaudio,noanswer) — however, note that, to avoid audio passthrough issues, you should use a Progress() statement on calls where there is not immediate answer supervision to allow audio to pass through, e.g:

	exten => s,1,Progress() ; allow audio pass through before supervision
		same => n,Playback(myaudio,noanswer) ; play without supervising
		same => n,Answer() ; NOW answer the call
		same => n,Playback(myaudio2)
		same => n,Hangup()
			

In other words, you should always explicitly begin with a Progress() or an Answer(). We do not recommend relying on implicit supervisory functions of Asterisk applications like Playback().

If your dialplan is already littered with simple Playback commands, you can use the following RegEx in Notepad++'s find and replace feature to locate Playback() statements that don't have the noanswer switch: ^(?!.*noanswer).*Playback.*$

Our friend Read() also provides answer supervision by default. The n option can be used to disable this, e.g. Read(number,custom/file,7,n).

Also note that the revertive pulsing script provides false supervision (which is NEVER desired). You must manually patch this script as described in the Revertive Pulsing section. If you do not patch this script, you should not use it. Inpulsing and outpulsing should never supervise.

Finally, advanced dialplan code may involve the ConfBridge() application where it is desired not to supervise. You should use the answer_channel=no user option to prevent ConfBridge() from automatically answering.

Intercept Codes

PhreakNet has a number of "NP Intercept" codes. These codes are integrated into the route table and are not provided by end offices or Class 4 tandems. (Thus, these codes should not be used by member nodes, but members may be interested in what these codes mean and what codes exist.)

These codes are manually dialable in the 231-90XX range.

Member nodes may take advantage of network centralized intercept services if they do not wish to provide their own. To do so, perform a lookup for a non-assigned network number and note the format (57 + the dialed #). In the dialplan, adjust your pattern matching such that the dialed number is suffixed to the 57 extension prefix. If no number can be provided, 57 alone is sufficient.

ANI II Digits

ANI II (pronounced "A-N-I two or Annie two") digits provide information about the type of originating line, call routing, and class of service. They are used on the PSTN for a variety of purposes. They provide additional information to downstream nodes about the nature of the call being received.

For the sake of consistency, the ANI II codes used on the network mirror their PSTN counterparts where relevant or possible. However, the exact usage of many codes is slightly different.

ANI2 if conveyed using ${CALLERID(ani2)}, which transmits natively over IAX2 trunks. No special variables are required.

Numbering Plan

A few words about the PhreakNet numbering plan:

For the most part, PhreakNet adheres to the traditional NANPA guidelines as they were originally set. This means that although NXX office codes exist today in the PSTN in the U.S., only NNX office codes are allowed on PhreakNet.

PhreakNet does not use country codes. Any node anywhere in the world on the network can be dialed with just 7 digits. This was an intentional decision made in order to make all nodes seem "local" and not "far away". Dialing more than 7 digits can discourage others from dialing a destination. This was a decision reached to better the network's experience for everyone.

Another dialing option is to dial 1+ for long-distance call. Dialing 1 + 7 digits will route your call over 2600 Hz-controlled trunks to the destination. The call goes to the same destination as if you had dialed it using 7 digits, but using a 2600-Hz controlled trunk, so now you can use your blue box on the call if you like.

In addition to that, PhreakNet has Feature Group B (950-WXXX) and Feature Group D dial-around codes (101XXXX). At this time, a public listing of these carrier access codes is not available.

Note that 10[02-9]XX represents former Feature Group D codes. These codes are part of the PhreakNet dial plan and will be routed to the appropriate intercept.

Any regular (NNX) calls dialable as 7 digits may also be dialed as 0+7D or 1+7D. 0+7D will complete the call through TSPS. 1+7D will complete the call over a 2600 Hz trunk which can be blue boxed.

976 exchange numbers are premium-rate numbers charged at a premium rate.

Furthermore, there are special numbers on PhreakNet, such as the following operator and directory services:

• 0 — Operator — rings human operator, goes to automatic operator if unavailable
• 411 — Directory Assistance — rings human operator, goes to automatic operator if unavailable
• KL5-1212 — same as 411, KLondike5-1212 (555-1212) will ring Directory Assistance
• 611 — Repair Service — rings human operator, goes to automatic operator if unavailable
• 711 — TTY Relay
• 811 — Business Office — rings the business office
• 112 — "Long Distance Operator" — rings automatic operator directly
• 113 — Information — rings automatic information operator directly
• 114 — Repair Service — rings automatic repair agent directly
• 117 — Telegrams
• 118 — StepNet Trunk
• 119 — Long Distance Trunk (Stacking)

Other directory services include:

• POPCORN — POPCORN, a.k.a. 767-2676 will provide your local time
• 211 — "Long Distance Operator" — rings automatic operator directly
• 511 — Temperature & Location
• 811 — Business Office
• 911 — Volunteer Emergency Dispatch

Finally, network-wide test numbers include:

• 311 — "Party Line"
• 660 — Ring & Dial Test
• 958 — ANAC
• 959 — Ringback

To avoid confusion, we will use "Directory Assistance" to refer to 411 and 555-1212, which go first to a human operator and then to an automatic one if none is available and "Information" to refer exclusively to 113, which is our automated directory service.

Other special NXX/NXX codes include 101 (Feature Group D), 950 (Feature Group B), 976 (premium rate), 973 (mobile telephone service), and 555 (fictitious numbers).

Pat Fleet Asterisk Sounds

To replace Asterisk's default Allison Smith voice prompts with "vintage" Pat Fleet prompts, run the following:

Note that not all Allison Smith prompts are currently replicated as Pat Fleet prompts. To only overwrite the ones that weren't redone by Pat, don't run the delete commands.

If the code above doesn't work for any reason, just know that the files and folders in /var/lib/asterisk/sounds/en should be replaced with those in the ZIP file, with the exception of the custom subdirectory which contains your custom recordings!

Automatic Intercept System

1. If you don't already have the Jane Barbe spliced AIS recordings, download them all from the C*NET Sounds Download page. You only need the files in the sounds folder itself, not any of its subfolders. Furthermore, you need only download the .wav files (not the .tgz files, for instance).
2. Use SoX to convert all of the wav files to ulaw files (see the SoX section in Appendix A for more on this).
3. Move all of the audio files beginning with a numeral (a number followed by either an n or a d to /var/lib/asterisk/sounds/en/custom/digits/. You may need to create this directory.
4. Move the remaining files to /var/lib/asterisk/sounds/en/custom/ais/. You may need to create this directory.
5. Add the following to your extensions.conf:

   [ais7] ; PhreakNet 20190212 NA
   exten => start,1,Set(LOCAL(num)=${ARG1})
   	same => n,Playback(custom/digits/${num:-7:1}n,noanswer)
   	same => n,Playback(custom/digits/${num:-6:1}n,noanswer)
   	same => n,Playback(custom/digits/${num:-5:1}d,noanswer)
   	same => n,Playback(custom/digits/blank,noanswer)
   	same => n,Playback(custom/digits/${num:-4:1}n,noanswer)
   	same => n,GotoIf($[${num:-3:3}=000]?thousand)
   	same => n,Playback(custom/digits/${num:-3:1}n,noanswer)
   	same => n,Playback(custom/digits/${num:-2:1}n,noanswer)
   	same => n,Playback(custom/digits/${num:-1:1}d,noanswer)
   	same => n,Return()
   	same => n(thousand),Playback(custom/ais/thousand,noanswer)
   	same => n,Return()
   
   [aisnis] ; PhreakNet 20190212 NA
   exten => start,1,Playback(custom/ais/clunk,noanswer)
   	same => n,Playback(custom/ais/numreach,noanswer)
   	same => n,Gosub(ais7,start,1(${ARG1}))
   	same => n,Playback(custom/ais/notinservice,noanswer)
   	same => n,Playback(custom/ais/pleasecheck,noanswer)
   	same => n,Playback(custom/ais/dialagain,noanswer)
   	same => n,Playback(custom/digits/blank,noanswer)
   	same => n,Return
   
   [aischanged] ; PhreakNet 20190212 NA ; This requires an argument with the new number
   exten => start,1,Set(LOCAL(num)=${ARG1})
   	same => n,Set(LOCAL(newnum)=${ARG2})
   	same => n,Verbose(AIS Number Changed -- "${num}")
   	same => n,Playback(custom/ais/clunk,noanswer)
   	same => n,Playback(custom/ais/numreach,noanswer)
   	same => n,Gosub(ais7,start,1(${num}))
   	same => n,Playback(custom/ais/hasbeenchanged,noanswer)
   	same => n,Playback(custom/ais/newnumberis,noanswer)
   	same => n,Gosub(ais7,start,1(${newnum}))
   	same => n,Playback(custom/ais/pleasemakenote,noanswer)
   	same => n,Playback(custom/digits/blank,noanswer)
   	same => n,Return
   
   [aisnpachanged] ; PhreakNet 20190212 NA ; This requires an argument with the new number
   exten => start,1,Set(LOCAL(oldnum)=${ARG1})
   	same => n,Set(LOCAL(newnum)=${ARG2})
   	same => n,Verbose(AIS Number Changed To Different NPA -- "${oldnum}")
   	same => n,Playback(custom/ais/clunk,noanswer)
   	same => n,Playback(custom/ais/numreach,noanswer)
   	same => n,GoToIf($[${oldnum}=nonum]?changed)
   	same => n,Gosub(ais7,start,1(${oldnum}))
   	same => n(changed),Playback(custom/ais/hasbeenchanged,noanswer)
   	same => n,Playback(custom/ais/newnumberis,noanswer)
   	same => n,GoToIf($[${LEN(${newnum})}=7]?seven)
   	same => n,Playback(custom/ais/area,noanswer)
   	same => n,Playback(custom/digits/${newnum:-10:1}n,noanswer)
   	same => n,Playback(custom/digits/${newnum:-9:1}n,noanswer)
   	same => n,Playback(custom/digits/${newnum:-8:1}d,noanswer)
   	same => n,Playback(custom/digits/blank,noanswer)
   	same => n(seven),Gosub(ais7,start,1(${newnum}))
   	same => n,Playback(custom/ais/pleasemakenote,noanswer)
   	same => n,Playback(custom/digits/blank,noanswer)
   	same => n,Playback(custom/ais/numreach,noanswer)
   	same => n,Playback(custom/ais/hasbeenchanged,noanswer)
   	same => n,Playback(custom/ais/newnumberis,noanswer)
   	same => n,GoToIf($[${LEN(${newnum})}=7]?seven2)
   	same => n,Playback(custom/ais/area,noanswer)
   	same => n,Playback(custom/digits/${newnum:-10:1}n,noanswer)
   	same => n,Playback(custom/digits/${newnum:-9:1}n,noanswer)
   	same => n,Playback(custom/digits/${newnum:-8:1}d,noanswer)
   	same => n,Playback(custom/digits/blank,noanswer)
   	same => n(seven2),Gosub(ais7,start,1(${newnum}))
   	same => n,Playback(custom/ais/pleasemakenote,noanswer)
   	same => n,Playback(custom/ais/dialagain,noanswer)
   	same => n,Playback(custom/digits/blank,noanswer)
   	same => n,Return
   
   [aisworking] ; PhreakNet 20190212 NA
   exten => start,1,Playback(custom/ais/clunk,noanswer)
   	same => n,Playback(custom/ais/numreach,noanswer)
   	same => n,Gosub(ais7,start,1(${ARG1}))
   	same => n,Playback(custom/ais/isworkingnum,noanswer)
   	same => n,Playback(custom/ais/willdialagainplease,noanswer)
   	same => n,Playback(custom/digits/blank,noanswer)
   	same => n,Return
   				

6. Now, going back to our starter dialplan code, let's modify it so that AIS is used on all calls to nonexistent extensions. Here is what we had initially:

   [KLondike5]
   … all your 555 extensions here …
   include => invalidincoming
   					

   Now, tweak your code so it is as follows:

   [KLondike5]
   … all your 555 extensions here …
   include => KLondike5-unmatched
   
   [KLondike5-unmatched]
   exten => _XXXX,1,Gosub(aisnis,start,1(${OC1}${EXTEN:-4}))
   	same => n,Hangup()
   					

Now, any time anyone calls a 555-XXXX number that doesn't exist, he will hear AIS instead of CBCAD!

An important technical footnote is necessary here: if you have multiple office codes, you will need a separate AIS "catch" context for each of them. The reason for this is that by the time a call hits [KLondike5], only the last 4 digits have been sent to that context, meaning the [KLondike5] context has no idea what office code was dialed. However, AIS expects the proper office code! Notice that ${OC1} is specified to add the office code back in before sending the call to AIS.

Consequently, if you had both the KLondike5 and KLondike6 exchanges, your code might look like this:

[KLondike5]
… all your 555 extensions here …
include => KLondike5-unmatched

[KLondike5-unmatched]
exten => _XXXX,1,Gosub(aisnis,start,1(${OC1}${EXTEN:-4}))
	same => n,Hangup()

[KLondike6]
… all your 556 extensions here …
include => KLondike6-unmatched

[KLondike6-unmatched]
exten => _XXXX,1,Gosub(aisnis,start,1(${OC2}${EXTEN:-4}))
	same => n,Hangup()
			

ANAC

A prerequisite for creating an ANAC (automatic number announcement circuit) is having the Jane Barbe AIS audio files downloaded. If you already have AIS on your system, you're all set. If you don't, you don't need AIS to have an ANAC. Simply navigate to the C*NET Sounds Download page and download all the files in that directory beginning with a numeral (you will see two for each number, one ending with an "n", and the other ending with a "d"). You don't need the other audio files in this directory unless you'd also like to set AIS up on your node.

Download all these files and use SoX to convert them to ulaw files (see Appendix A for more on this). Then, place the ulaw files inside /var/lib/asterisk/sounds/en/custom/digits/ on your system. You may need to create the digits directory if you don't have AIS on your node (and hence never created that directory).

Then, copy and paste the following code into your dialplan (i.e. extensions.conf):

[anac] ; PhreakNet 20190212 NA
exten => start,1,Set(LOCAL(num)=${FILTER(0-9,${CALLERID(num)})})
	same => n,GotoIf($[${LEN(${num})}=7]?d7)
	same => n,GotoIf($[${LEN(${num})}=6]?d6)
	same => n,GotoIf($[${LEN(${num})}=5]?d5)
	same => n,GotoIf($[${LEN(${num})}=4]?d4)
	same => n,GotoIf($[${LEN(${num})}=3]?d3)
	same => n,GotoIf($[${LEN(${num})}=2]?d2)
	same => n,GotoIf($[${LEN(${num})}=1]?d1)
	same => n,GotoIf($[${LEN(${num})}=0]?d0)
	same => n(d7),Playback(custom/digits/${num:-7:1}n,noanswer)
	same => n(d6),Playback(custom/digits/${num:-6:1}n,noanswer)
	same => n(d5),Playback(custom/digits/${num:-5:1}d,noanswer)
	same => n,Playback(custom/digits/blank,noanswer)
	same => n(d4),Playback(custom/digits/${num:-4:1}n,noanswer)
	same => n(d3),Playback(custom/digits/${num:-3:1}n,noanswer)
	same => n(d2),Playback(custom/digits/${num:-2:1}n,noanswer)
	same => n(d1),Playback(custom/digits/${num:-1:1}d,noanswer)
	same => n(d0),Playback(beep,noanswer)
	same => n(done),Return
			

Now, to make your ANAC work, you'll need to put it on an extension. "958" is a network-wide ANAC, so you may want to put your ANAC on an extension ending in 958, like so:

exten => 9958,1,Gosub(anac,start,1)
	same => n,Hangup()
			

The reason we haven't included an Answer() statement here is because this is a test line, and test lines in the old network hardly ever supervised. Your ANAC should not supervise.

Speaking Clock

Doing time announcements is something relatively complex that Asterisk makes relatively simple. The code featured here is the same code that runs POPCORN network-wide. You can set-up a speaking clock or "time number" on your node that is hardcoded to use your timezone, if you wish. The following code makes this relatively easy to do:

[time] ; PhreakNet 20190212 NA ; ARG1 is number of times to announce the time, ARG2 is the timezone to use
exten => s,1,Set(NumLoops=0)
	same => n(start),Set(FutureTime=$[${EPOCH} + 8])
	same => n,Set(FutureTimeMod=$[${FutureTime} % 10])
	same => n,Set(FutureTime=$[${FutureTime} - ${FutureTimeMod} + 10])
	same => n,Gosub(sub-hr12format,s,1(${ARG2}))
	same => n,WaitUntil(${FutureTime})
	same => n(playbeep),Playback(beep)
	same => n,GotoIf($[${LEN(${ARG4})}=6]?skipwait)
	same => n,Wait(5)
	same => n(skipwait),Set(NumLoops=$[${NumLoops} + 1])
	same => n,GotoIf($[${NumLoops} < ${ARG1}]?start)
	same => n,GotoIf($[${LEN(${ARG3})}=4]?skipgoodbye)
	same => n,Playback(goodbye)
	same => n(skipgoodbye),Return()

[sub-hr12format]
exten => s,1,Answer
exten => s,n,Playback(at-tone-time-exactly)
exten => s,n,SayUnixTime(${FutureTime},${ARG1},IM 'vm-and' S 'seconds' p)
exten => s,n,Return()
exten => s,n,Playback(at-tone-time-exactly)
exten => s,n,SayUnixTime(${FutureTime},${ARG1},IM 'vm-and' S 'seconds' p)
exten => s,n,Return()
			

ARG1 is the number of times to announce the time. POPCORN is set to announce the time 100 times, so it goes on for quite a while. You'll probably want to set yours to a number between 5 and 10.

ARG2 is the timezone to use. The timezones are the "tz" format used in Unix. You can use this list of timezones and corresponding TZ database names to help you.

ARG3 and ARG4 are optional, and you will probably not need to use them. If ARG3 is set to skip, or any 4 characters for that matter, the speaking clock will not say "goodbye" before exiting. You may need to use ARG3 if you use this speaking clock as part of a broader context, rather than just a single, dedicated extension.

If ARG4 is set to nowait, or any 6 characters for that matter, then the speaking clock will not wait 5 seconds after the beep before entering another cycle. This is usually useful when you are calling [time] with ARG1 set to 1 (i.e. you only want the time announced once), and you wish to immediately exit the subroutine after the beep and continue with the dialplan code in the extension that called it. Note that if ARG1 is NOT 1 and you use ARG4 to disable the wait, it will not only disable the wait the last cycle, but every cycle, which is potentially undesirable. ARG4 is only meant to be set if ARG1 is equal to 1, but if you want to make the subroutine a bit more foolproof, you might modify it slightly so that it only calls GotoIf($[${LEN(${ARG4})}=6]?skipwait) if it is in its last cycle (in which case NumLoops will be 1 less than ARG1).

Now, to create a speaking clock for say, Chicago, or any city located in the Central Time Zone, that announces the time 7 times, all you need to do is add the following to your dialplan:

The speaking clock above is the traditional "at the tone, the time will be exactly X:YY and ZZ seconds, A.M. … beep.

If you'd like something a bit more customizable, check out the Asterisk TIM project on GitHub. At this time, the voice provided is Pat Simmons, but you can obtain and use your own custom audio prompts.

Airport Weather

For instructions on how to get airport-code-based weather reports in Asterisk, see this NerdVittles tutorial.

Currently, we don't have any of these types of services. The temperature readings provided at BE1-1200 and related numbers are powered by paid APIs.

Milliwatt Test Tone

In Asterisk, there are actually several ways you could implement a milliwatt (or 1004 Hz test) tone. Asterisk itself has a Milliwatt function:

[mw]
exten => start,1,Wait(0.5)
	same => n,Milliwatt(m)
	same => n,Return
			

We wait 0.5 seconds before starting as milliwatt numbers usually offer a split second of silence before blasting your eardrums. The m option was added to Asterisk by PhreakNet to add the correct standard timing (1 second pause every 10 seconds).

By now, you probably know that to call this, you would need to call Gosub(mw,start,1(600)) from an extension in your dialplan. (You can tell this is a subroutine because it has the Return statement in it.)

However, a great many people choose to create their milliwatt test numbers themselves. Since a milliwatt test tone consists of pure frequencies, this is relatively easy. To make things interesting, we have here chosen to combine 1004 Hz and 1000 Hz for our milliwatt (the latter was the original frequency, according to Steph Kerman "A mW is inherently a pure tone, originally nominally 1000Hz but eventually these tones were offset by 4Hz to 404, 1004 and 2804 for compatibility with digital transmission"):

[mwtone]
exten => start,1,Wait(0.5)
	same => n,PlayTones(1004/1000)
	same => n,Wait(${ARG1})
	same => n,Return
			

In this case, calling Gosub(mwtone,start,1(600)) would play a 1004 Hz tone for 600 seconds (or 10 minutes). Note that isn't quite the same as a Milliwatt tone test. You can tweak the timings to make it work like one, though.

Now, if you want to really kick it up a notch, you can increase the volume of the channel so the milliwatt tone comes out more loudly than it would otherwise:

[mwtone]
exten => start,1,SET(VOLUME(TX)=8)
	same => n,Wait(0.5)
	same => n,PlayTones(1004/1000)
	same => n,Wait(${ARG1})
	same => n,SET(VOLUME(TX)=1)
	same => n,Return
			

A volume specification of about 8 matches PSTN milliwatt test numbers fairly closely. Of course, you can play around with this until you find a volume setting that works for you. Simply replace 8 with ${ARG2}, and then pass in a second argument containing the new volume. This way, you can easily create multiple extensions with different volume milliwatt tones.

Ringback

Ringback is a fairly simple task that is quite involved in terms of dialplan code. Here is a simple way to do it:

[ringback]
exten => s,1,Progress
	same => n,PlayTones(600*120)
	same => n,Wait(60)
exten => h,1,Originate(Local/${CALLERID(num)}@to-phreaknet,app,Wait,1209600,,,ac(5559959)n(Ringback))
			

The PlayTones(600*120) is what will happen when this number is called. In this case, you will here 600 Hz modulated with 120 Hz, or old city dial tone (though it sounds a bit modern, given that it's generated by a computer rather than an electromechanical tone plant). This line is optional, but does provide an indication to the caller that something has happened; if you don't want the caller to hear anything, you could replace this with Wait(60).

The ringback itself doesn't actually happen until the caller hangs up (which is why the h extension is used here as well). When the caller hangs up, Asterisk creates a call file, dumps it in the spool directory, and it gets processed.

You will need to replace 5559959 here with the extension at which you create your ringback number. If you choose to have it be extension 9959 on your exchange, you don't need to change anything; otherwise, adjust the last 4 digits to the proper extension number. You should make sure that the number you use here matches the number used to call your ringback line. That way, if somebody calls your ringback line, the incoming Caller ID on the automatically generated call matches the number he dialed in the first place.

Note that [ringback] is not a subroutine. You will need to call it as exten => 9959,1,Goto(ringback,s,1).

Revertive Pulsing Script

First, as outlined in the "Pre-Requisites" section, you will need to have PHP installed in order to use the revertive pulsing script. Once PHP is installed, exit the Asterisk CLI to the main command line and run the following commands:

wget https://octothorpe.info/downloads/pulsar-agi.tar.gz
mv pulsar-agi.tar.gz /var/lib/asterisk
cd /var/lib/asterisk
tar xvfz pulsar-agi.tar.gz
chmod 777 /var/lib/asterisk/agi-bin/pulsar.agi

A file named pulsar.agi should now be located in /var/lib/asterisk/agi-bin/.

Audio files go in /var/lib/asterisk/sounds/pulsar/.

It's worth mentioning that by default this script DOES supervise, which is incorrect! If you modify the AGI file, you can add the noanswer flag to the Playback command so that revertive pulsing does NOT return premature answer supervision (e.g. turn Playback(file) into Playback(file,noanswer).

Do not use revertive pulsing if your revertive pulsing supervises (i.e. don't use revertive pulsing if you are not going to modify the AGI). Otherwise, it will supervise, which is against the supervision standards.

Now, add the following subroutine to your dialplan:

[revertive] ; PhreakNet 20190212 NA ; Revertive pulsing (requires PHP installation)
exten => start,1,Verbose(Revertive Pulse Generator: ${ARG1})
	same => n,AGI(pulsar.agi,${ARG1},${ARG2},${ARG3}) ; ARG2 can be set to 1 to indicate a "B-side" that adds 5 pulses to the second RP digit.
	same => n,Return() ; ARG1 should be a 4-digit extension and ARG3 must be one of the following: panel,1xb,5xb
			

ARG1 is the 4-digit extension to revertive pulse.

ARG2 can be set to 1 to indicate a "B-side" that adds 5 pulses to the second RP digit. Otherwise, set it to 0.

ARG3 should be the type of revertive pulsing to simulate. Your options are panel, 1xb, or 5xb. If you are simulating one of these switches by chance, then use that one. Each type of revertive pulsing sounds very distinctive, so make sure to get this one right!

As you can see, you might call the revertive pulser like this: Gosub(revertive,start,1(${EXTEN:-4},0,5xb)). This will simulate #5XB revertive pulsing.

Unlike MF digits, which you could use for both inpulsing and outpulsing, revertive pulsing is a signaling method you should only use for incoming calls. Let's revisit our incoming contexts:

[internal-users]
include => local
include => long-distance
include => invalidincoming

[external-users]
include => local
include => invalidincoming

[local]
exten => _${OC1}XXXX,1,Goto(KLondike5,${EXTEN:-4:4},1)
			

You might modify [local] so it is as follows:

[local]
exten => _${OC1}XXXX,1,Gosub(revertive,start,1(${EXTEN:-4},0,5xb))
	same => n,Goto(KLondike5,${EXTEN:-4:4},1)
			

Now, all callers dialing into whichever office code ${OC1} happens to be will hear Number 5 crossbar revertive inpulsing before the call connects!

The only catch here is, if you look at the [internal-users] context, you will see that you will also hear revertive pulsing on a call that is "local" to you, i.e. within the same exchange. At worst, this is undesired, at best, it is not really technically correct.

The solution is having a different context for incoming "long-distance" calls, like so:

[internal-users]
include => local-intraoffice
include => long-distance
include => invalidincoming

[external-users]
include => local-interoffice
include => invalidincoming

[local-intraoffice]
exten => _${OC1}XXXX,1,Goto(KLondike5,${EXTEN:-4:4},1)

[local-interoffice]
exten => _${OC1}XXXX,1,Gosub(revertive,start,1(${EXTEN:-4},0,5xb))
	same => n,Goto(KLondike5,${EXTEN:-4:4},1)
			

Now, callers from other exchanges will hear revertive pulsing on all calls to KLondike5, but you will not. It is recommended that you adopt this approach.

If you'd like, in order to economize on code, you might adjust the last two contexts to the following:

[local-intraoffice]
exten => _${OC1}XXXX,1,Goto(KLondike5,${EXTEN:-4:4},1)

[local-interoffice]
exten => _${OC1}XXXX,1,Gosub(revertive,start,1(${EXTEN:-4},0,5xb))
	same => n,Goto(local-intraoffice,${EXTEN:-4:4},1)
			

What's the point of this, you might ask? We just made external callers hop through another step!

The advantage of structuring your code this way is you can simply define the inpulsing for each exchange in [local-interoffice], then send it to the other context. If you wanted a sound played on all incoming calls, you could include it in the first context only; in other words, you can cut down on future duplicated code.

The downside of using this approach is, as before, all internal and external callers are now sent to the same destination context. If you had a [KLondike5-internal] context, you might want to keep them separate, in order to allow different classes of callers access to different classes of extensions:

[internal-users]
include => local-intraoffice
include => long-distance
include => invalidincoming

[external-users]
include => local-interoffice
include => invalidincoming

[local-intraoffice]
exten => _${OC1}XXXX,1,Goto(KLondike5-internal,${EXTEN:-4:4},1)

[local-interoffice]
exten => _${OC1}XXXX,1,Gosub(revertive,start,1(${EXTEN:-4},0,5xb))
	same => n,Goto(KLondike5,${EXTEN:-4:4},1)

[KLondike5-internal]
exten => 6666,1,Goto(verysecretcontext,s,1)
include => KLondike5

[KLondike5]
; all public extensions here
include => invalidincoming ; if you have AIS, you should include KLondike5-unmatched instead

Now, if you call KL5-6666, you will end up going to [verysecretcontext], but callers from other exchanges will end up hearing an intercept message (or AIS) instead. Technically, this might not make sense, as the extension exists; it's just not accessible to external callers, and from Asterisk's perspective when routing an incoming call from another exchanges, the extension does not exist since it's in a different context to which it does not have access.

Part of Asterisk is structuring your dialplan code properly, and this is a skill that takes time to develop. Fortunately, you have an almost infinite amount of flexibility in terms of how you route your calls, so pick the approach that works best for you, knowing that you have the flexibility to tweak it should you ever need to.

MFer

mf1 = !700+900/55,!0/50
mf2 = !700+1100/55,!0/50
mf3 = !900+1100/55,!0/50
mf4 = !700+1300/55,!0/50
mf5 = !900+1300/55,!0/50
mf6 = !1100+1300/55,!0/50
mf7 = !700+1500/55,!0/50
mf8 = !900+1500/55,!0/50
mf9 = !1100+1500/55,!0/50
mf0 = !1300+1500/55,!0/50
kp = !1100+1700/100,!0/50
kp2 = !1300+1700/100,!0/50
st = !1500+1700/35,!0/50
st2 = !900+1700/35,!0/50
			

[signallookup] ; PhreakNet 20210208 NA 
exten => MF,1,Return("|","!0/${ARG3}","!700+900/${ARG1}|!0/${ARG2}","!700+1100/${ARG1}|!0/${ARG2}","!900+1100/${ARG1}|!0/${ARG2}","!700+1300/${ARG1}|!0/${ARG2}","!900+1300/${ARG1}|!0/${ARG2}","!1100+1300/${ARG1}|!0/${ARG2}","!700+1500/${ARG1}|!0/${ARG2}","!900+1500/${ARG1}|!0/${ARG2}","!1100+1500/${ARG1}|!0/${ARG2}","!1300+1500/${ARG1}|!0/${ARG2}","!1100+1700/${ARG5}|!0/${ARG2}","!1500+1700/${ARG4}|!0/${ARG2}","!900+1700/${ARG4}|!0/${ARG2}","!1300+1700/${ARG4}|!0/${ARG2}","!700+1700/${ARG4}|!0/${ARG2}")
exten => SF,1,Return("|",200,100,"!0/200","!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}") ; ARG1 = frequency (2600), ARG2 = on (60ms), ARG3 = off (40ms) / SFX = interdigit silence

[mfer] ; PhreakNet 20181212 NA/BJC, revised 20210208
exten => start,1,Set(number=${FILTER(0-9,${ARG1})}) ; ARG1 = digits to MF
	same => n,Gosub(signallookup,MF,1(50,50,500,35,100)) ; ARG1 = on, ARG2=off, ARG3=silence, ARG4=ST duration, ARG5=KP duration
	same => n,Set(ARRAY(digitsep,mfx,mf1,mf2,mf3,mf4,mf5,mf6,mf7,mf8,mf9,mf0,mfkp,mfst,mfstp,mfst2p,mfst3p)=${GOSUB_RETVAL}) ; R1, CCIT5: STP ~ code 12, ST2P ~ KP2, ST3P ~ code 11
	same => n,Set(c=0)
	same => n,Set(numdigits=${LEN(${ARG1})})
	same => n,Set(tonestring=${mfx}${digitsep}${mfkp}${digitsep})
	same => n,While($[${c}<${numdigits}])
	same => n,Set(mfnext=${mf${number:${c}:1}})
	same => n,Set(tonestring=${tonestring}${mfnext}${digitsep})
	same => n,Set(c=${INC(c)})
	same => n,EndWhile
	same => n,Set(mfnext=${mfst})
	same => n,Set(tonestring=${tonestring}${mfnext}${digitsep})
	same => n,Set(tonestring=${tonestring}${mfx})
	same => n,Set(toneduration=$[100*${LEN(${ARG1})}+1785])
	same => n,PlayTones(${FILTER(0-9\x21/|+,${tonestring})})
	same => n,Wait($[${toneduration}/1000])
	same => n,StopPlayTones()
	same => n,Return()
			

[mfer2] ; PhreakNet 20181212 NA/BJC, revised 20210208
exten => start,1,NoOp()
	same => n,Set(lon=50) ; 50
	same => n,Set(loff=50) ; 50
	same => n,Set(lsil=500) ; 500
	same => n,Set(lst=50) ; 35/50
	same => n,Set(lkp=100) ; 100
	same => n,Gosub(signallookup,MF,1(${lon},${loff},${lsil},${lst},${lkp})) ; ARG1 = on, ARG2=off, ARG3=silence, ARG4=ST duration, ARG5=KP duration
	same => n,Set(ARRAY(digitsep,mfx,mf1,mf2,mf3,mf4,mf5,mf6,mf7,mf8,mf9,mf0,mfkp,mfst,mfstp,mfst2p,mfst3p)=${GOSUB_RETVAL}) ; R1, CCIT5: STP ~ code 12, ST2P ~ KP2, ST3P ~ code 11
	same => n,Set(number=${FILTER(0-9,${ARG1})}) ; ARG1 = digits to MF
	same => n,Set(mfstart=${FILTER(0-9,${ARG2})}) ; ARG2 = optional ST delimiter
	same => n,Set(number2=${FILTER(0-9,${ARG3})}) ; ARG3 = optional 2nd group of digits to MF
	same => n,Set(cv=0)
	same => n,Set(numdigits=${LEN(${number})})
	same => n,Set(tonestring=${mfx}${digitsep}${mfkp}${digitsep})
	same => n,While($[${cv}<${numdigits}])
	same => n,Set(mfnext=${mf${number:${cv}:1}})
	same => n,Set(tonestring=${tonestring}${mfnext}${digitsep})
	same => n,Set(cv=${INC(cv)})
	same => n,EndWhile()
	same => n,Set(mfnext=${mfst})
	same => n,ExecIf($["${mfstart}"="mfstp"]?Set(mfnext=${mfstp})) ; R1 STP, CCIT5: Code 12
	same => n,ExecIf($["${mfstart}"="mfst2p"]?Set(mfnext=${mfst2p})) ; R1 ST2P, CCIT5: KP2
	same => n,ExecIf($["${mfstart}"="mfst3p"]?Set(mfnext=${mfst3p})) ; R1 ST3P, CCIT5: Code 11
	same => n,Set(tonestring=${tonestring}${mfnext})
	;same => n,Set(tonestring=${digitsep}${tonestring}${mfx})
	same => n,Set(toneduration=$[$[${lon}+${loff}]*$[${LEN(${number})}+${LEN(${number2})}]+${lsil}+${lkp}+${loff}+${lst}+${loff}])
	same => n,GotoIf($[${LEN(${number2})}=0]?play)
	same => n,Set(cv=0)
	same => n,Set(numdigits=${LEN(${number2})})
	same => n,Set(tonestring=${tonestring}${digitsep}${mfkp}${digitsep})
	same => n,While($[${cv}<${numdigits}])
	same => n,Set(mfnext=${mf${number2:${cv}:1}})
	same => n,Set(tonestring=${tonestring}${mfnext}${digitsep})
	same => n,Set(cv=${INC(cv)})
	same => n,EndWhile()
	same => n,Set(tonestring=${tonestring}${mfst})
	same => n,Set(toneduration=$[${toneduration}+${lkp}+${loff}+${lst}+${loff}]) ; add on aditional KP/ST
	same => n(play),Set(td=0)
	same => n,Set(UNSHIFT(tones,|)=${tonestring})
	same => n,While($["${SET(t=${SHIFT(tones,|)})}" != ""])
	same => n,Set(t=${FILTER(0-9,${CUT(t,/,2)})})
	same => n,Set(td=$[${td}+${t}])
	same => n,EndWhile()
	same => n,Set(toneduration=${td}) ; increased buffer needed for longer strings
	same => n,PlayTones(${FILTER(0-9\x21/|+,${tonestring})})
	same => n,Wait($[${toneduration}/1000])
	same => n,StopPlayTones()
	same => n,Return()
			

[local-interoffice]
exten => _${OC1}XXXX,1,Gosub(revertive,start,1(${EXTEN:-4},0,5xb))
	same => n,Goto(KLondike5,${EXTEN:-4:4},1)
exten => _${OC2}XXXX,1,Gosub(mfer,start,1(${EXTEN}))
	same => n,Goto(KLondike6,${EXTEN:-4:4},1)
			

exten => _${OC1}XXXX,1,Gosub(mfer,start,1(${EXTEN}))
	same => n,Gosub(revertive,start,1(${EXTEN:-4},0,5xb))
	same => n,Goto(KLondike5,${EXTEN:-4:4},1)
			

NOTE: All the above is obsolete and has been superseded by the native SendMF() application in Asterisk.

Now, external callers would hear MFing, followed by #5XB revertive pulsing.

You have the flexibility to choose from various inpulsing and outpulsing options for your node, but remember to stay "period-correct". You might hear MFing followed by revertive pulsing, for instance, but you would not be likely to hear revertive pulsing followed by MFing! Pay attention to the order of your inpulsing and outpulsing and make sure it stays realistic.

Note: Do not add any outpulsing directly in the dialphreaknet subroutine! Instead, create a separate outpulsing context that you call from within phreaknet-route.

SFer

If you do not already have the [signallookup] subroutine (the same one used for the MFer), you will need that:

[signallookup] ; PhreakNet 20210208 NA ; Bell System specs say 1st 50 should be 55? / MFX - silence / ST - Don has 35 here set to 50
exten => MF,1,Return("|","!0/${ARG3}","!700+900/${ARG1}|!0/${ARG2}","!700+1100/${ARG1}|!0/${ARG2}","!900+1100/${ARG1}|!0/${ARG2}","!700+1300/${ARG1}|!0/${ARG2}","!900+1300/${ARG1}|!0/${ARG2}","!1100+1300/${ARG1}|!0/${ARG2}","!700+1500/${ARG1}|!0/${ARG2}","!900+1500/${ARG1}|!0/${ARG2}","!1100+1500/${ARG1}|!0/${ARG2}","!1300+1500/${ARG1}|!0/${ARG2}","!1100+1700/100|!0/${ARG2}","!1500+1700/${ARG4}|!0/${ARG2}","!1300+1700/100|!0/${ARG2}","!900+1700/${ARG4}|!0/${ARG2}","!700+1700/${ARG4}|!0/${ARG2}")
exten => SF,1,Return("|",200,100,"!0/200","!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}","!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}|!${ARG1}/${ARG2}|!0/${ARG3}") ; ARG1 = frequency (2600), ARG2 = on (60ms), ARG3 = off (40ms) / SFX = interdigit silence
			

Regardless, you will need the [sfer] subroutine:

[sfer] ; PhreakNet 20190218 NA, revised 20210208 ; ARG1 = digits to SF
exten => start,1,Set(number=${FILTER(0-9,${ARG1})}) ; ARG1 = digits to do something with
	same => n,Gosub(signallookup,SF,1(2600,60,40))
	same => n,Set(ARRAY(digitsep,sfinter,sfpulse,sfx,sf1,sf2,sf3,sf4,sf5,sf6,sf7,sf8,sf9,sf0)=${GOSUB_RETVAL})
	same => n,Set(c=0)
	same => n,Set(p=0)
	same => n,Set(numdigits=${LEN(${ARG1})})
	same => n,Set(tonestring=)
	same => n,While($[${c}<${numdigits}])
	same => n,Set(nextp=${number:${c}:1})
	same => n,ExecIf($["${nextp}"="0"]?Set(nextp=10))
	same => n,Set(p=$[${p}+${nextp}])
	same => n,Set(sfnext=${sf${number:${c}:1}})
	same => n,Set(tonestring=${tonestring}${sfnext}${digitsep})
	same => n,Set(tonestring=${tonestring}${sfx}${digitsep})
	same => n,Set(c=${INC(c)})
	same => n,EndWhile
	same => n,Set(toneduration=$[${sfpulse}*${p}+${sfinter}*${numdigits}])
	same => n,PlayTones(${FILTER(0-9\x21/|+,${tonestring})})
	same => n,Wait($[${toneduration}/1000])
	same => n,StopPlayTones()
	same => n,Return()
			

Dial Pulser

Below is the dialplan coded needed for the dial pulser:

[dialpulser] ; PhreakNet 20190212 NA
exten => start,1,Set(number=${ARG1}) ; ARG1 = digits to dial pulse
	same => n,Set(c=0)
	same => n,Set(numdigits=${LEN(${ARG1})})
	same => n,While($[${c}<${numdigits}])
	same => n,Playback(custom/dialpulses/dp${number:${c}:1},noanswer)
	same => n,Set(c=${INC(c)})
	same => n,EndWhile
	same => n,Return
			

ARG1 is simply what number to dial pulse. The number can be of any length (beside 0, obviously).

You will need audio recordings of each number being dial pulsed. You can easily extract dial pulses from various Evan Doorbell recordings. Prefix each digit with dp when you save the audio file, convert to ulaw, and then move to /var/lib/asterisk/sounds/en/custom/dialpulses/.

If you'd prefer to use our dial pulsing audio files, you may download them here.

Conference Bridges

Creating a conference bridge in Asterisk is relatively easy. Here, we will provide a template for creating a kind of so-called "party line" conference bridge, much like those on which Evan Doorbell spent hours talking and listening in the 1970s.

You will need to add the following to confbridge.conf:

[bridge]
type=bridge
sound_join=/var/lib/asterisk/sounds/en/custom/conf/confjoin  ; The sound played to everyone when someone enters the conference.
sound_leave=/var/lib/asterisk/sounds/en/custom/conf/confdis ; The sound played to everyone when someone leaves the conference.

[caller]
type=user
dsp_drop_silence=no
announce_only_user=no

[volmenu]
type=menu
4=decrease_listening_volume
5=reset_listening_volume
6=increase_listening_volume
7=decrease_talking_volume
8=reset_talking_volume
9=increase_talking_volume
0=toggle_mute

[advmenu]
type=menu
1=participant_count
4=decrease_listening_volume
5=reset_listening_volume
6=increase_listening_volume
7=decrease_talking_volume
8=reset_talking_volume
9=increase_talking_volume
0=toggle_mute
			

You will need the "confjoin" and "confdis" audio files. You can use whatever audio files you like here; we recommend extracting actual "party line" conference joining and disconnect noises from an Evan Doorbell recording.

To have an extension go to a conference bridge, specify the following in your dialplan: ConfBridge(1,bridge,caller).

In this case, we have given the bridge an ID of 1. This number should be unique for each conference. You may choose to set this equal to the extension on which this conference is located.

You can have multiple extensions go to the same bridge but with different "experiences". For example, you could create a separate extension that joins the same bridge but also features a menu: ConfBridge(1,bridge,caller,volmenu). As you've probably figured out by now, ARG1 for ConfBridge is which bridge to use, ARG2 is which bridge profile to user, ARG3 is which user profile to use, and ARG4, which is optional, is which menu to use, if any. If you're trying to recreate a 1970s "party line", though, omit the menu.

It is important to note that conferences are not created or defined in confbridge.conf! Creating a second conference is as simple as making another extension with a ConfBridge call that uses something different for ARG1. For instance, ConfBridge(2111,bridge,caller) and ConfBridge(052,bridge,caller) would be two different conferences entirely, though they both use the same bridge and user profiles, so the experiences (i.e. sounds and functionality) will be the same.

Non-Supervising Conference Bridges

By default, the ConfBridge() application will supervise with no alternative, unlike the Playback() application which accepts a noanswer argument. Because ConfBridge is often the only way to get much functionality in Asterisk to work as desired, this is a significant limitation; however, modifying the source code can remove this behavior for better operation.

As of June 2021, Asterisk's app_confbridge now has the answer_channel user option. Set answer_channel = no, with Asterisk 18.5+/Asterisk 16.19+. No patching is required.

You must either use Progress() or Answer(). Failing to do so may result in no audio being passed at all.

Take care to revisit all uses of ConfBridge in your dialplan code to ensure you're using Progress or Answer before any calls to ConfBridge.

Answering Machines

Answering machines and voicemail are, in many ways, the antithesis of each other. Functionally, an answering machine has no place in Asterisk. However, it may be neat to simulate a physical telephone answering device (or answering machine) in Asterisk, to make it sound like somebody has reached an answering machine. This can be done with fairly trivial dialplan code.

Behind the scenes, the VoiceMail application can still be used. For the most part, the caller would never notice the difference. However, by default, Asterisk plays its own beep tone to the caller. Simply add the t option, specifying a custom beep tone to use, or leaving this argument empty to suppress the beep entirely.

This allows you to use your own outgoing answering machines without a modern Asterisk tone coming on at the end, and you can even use beep tones that are noticably analog and old sounding.

Note that if a user presses the # key while recording, Asterisk will say "thank you" and exit the VoiceMail application (playing auth-thankyou). This patch does not address that behavior. If users are allowed to review messages in voicemail.conf, an additional menu plays afterwards anyways, so it's not merely a question of suppressing an auth-thankyou somewhere, and it's probably best to just leave this behavior be.

Hook Flashing

As of April 2021, Asterisk itself has very minimal support for hook flashing. All Asterisk has historically done is propogate these events across channels from one to the other, if they are compatible. It has not had any ability to do anything with these events or expose them in any way (i.e. AMI).

There are two primary reasons for this:

• In Asterisk, the "modern" and "hip" way to do certain tasks usually accessed through a hook flash is through special feature codes like *2 that are defined in features.conf
• ATAs can handle hook flashes locally and perform a small but generally sufficient set of actions, such as Call Waiting, Three Way Calling, etc.

For building advanced functionality, it is essential to be able to handle hook flashes in Asterisk rather than let them be handled locally. In fact, ATAs have better support for this than they do handling these events locally. Grandstream ATAs, for example, will not connect to the switch at all if you flash to spawn a three-way call, even if you have off-hook auto-dial configured. Generally speaking, you are limited to the idiosyncrasies of different ATAs and many events are simply impossible to handle in the desired way. Sending flash events to Asterisk and performing bridge operations switch-side is vastly superior, though it requires a significant investment in configuring such a system (not for the light-hearted!) You will need to handle flash events as desired and implement all relevant functionality yourself, including writing your own Call Waiting from scratch. However, if you are designing an actual Class 5 switch, this is probably what you are trying to do anyways.

cd /tmp
wget https://code.phreaknet.org/asterisk/sip.patch
wget https://code.phreaknet.org/asterisk/file.patch  
wget https://issues.asterisk.org/jira/secure/attachment/60149/ASTERISK-29373.diff
wget https://code.phreaknet.org/asterisk/stasis_sample.patch
wget https://code.phreaknet.org/asterisk/stasis_channels.h.patch
wget https://code.phreaknet.org/asterisk/stasis_channels.c.patch
wget https://code.phreaknet.org/asterisk/manager_channels.patch
wget https://code.phreaknet.org/asterisk/stasis.patch
wget https://code.phreaknet.org/asterisk/channel.patch
cd /usr/src/asterisk-18.3.0/
patch -u -b channels/chan_sip.c -i /tmp/sip.patch # add application/hook-flash support for SIP INFO flash events
patch -u -b main/file.c -i /tmp/file.patch # ignore flash events and don't throw an error
patch -u -b res/res_rtp_asterisk.c -i /tmp/ASTERISK-29373.diff # don't create duplicate flash events for RTP flashes
patch -u -b configs/samples/stasis.conf.sample /tmp/stasis_sample.patch # AMI flash event
patch -u -b include/asterisk/stasis_channels.h /tmp/stasis_channels.h.patch # AMI flash event
patch -u -b main/stasis_channels.c /tmp/stasis_channels.c.patch # AMI flash event
patch -u -b main/manager_channels.c /tmp/manager_channels.patch # AMI flash event
patch -u -b main/stasis.c /tmp/stasis.patch # AMI flash event
patch -u -b main/channel.c /tmp/channel.patch # stop throwing warning for flash event and handle flash events by triggering AMI event
make
make install
asterisk -rx "core restart now"
			

As of Asterisk 18.4 and 18.5, all of these patches are part of "official" Asterisk from Sangoma.

This will expose hook flashes as AMI events. You will need to handle them using an out of band AMI process, such as PAMI for PHP. You can then spawn dialplan execution to perform flash logic and do other operations as desired based on the state of the call(s).

As of April 2021, the patches to add AMI support have been submitted to Digium but are not yet part of the master Asterisk source code. Until the feature is available natively, you simply need to patch your copy and recompile. It should not take more than a minute or two to do.

Credits & Contributions: J. Colp, Digium for correcting improper RTP stream handling of flash events; N.A. for two issues which add application/hook-flash support to SIP, suppress errors in file.c, and adding full AMI (Asterisk Manager Interface) support for flash events.

Announcement Drums

It is relatively simple to set up shared (and even talkable) busy signals or simulated drum announcement machines with Asterisk. The below will cover a simple example of a drum announcement. There are multiple ways to do this and this is just an example.

You will likely want the non-supervising ConfBridge patch for this kind of stuff.

Typically, the idea with these is:

• If the announcement is idle and it gets a call, "start" the drum up
• All callers who reach the announcement before it gets to the beginning are all simultaneously cut into it at the same time
• When no callers are left, the "machine" turns itself off. In software, this is even more important, because there's no legitimate reason to waste resources when nothing needs them

The example below cuts callers into an announcement for one announcement cycle and then sends them to another destination.

The example below supervises when callers are cut into the bridge. For intercepts, this may not be desirable. Instead of the initial Dial(), you could use MusicOnHold() directly and then StopMusicOnHold() instead of Answer() in the [drum]. With the patch to ConfBridge, callers will then be cut into a bridge, and the call still will not have supervised. All depends on how you want to set things up.

The example below is similar to the code that powers Centralized Intercept on PhreakNet. POPCORN (767-2676) on PhreakNet also uses similar "drum technology". Callers are cut in at the same time and stay on for 4 full cycles of the time from when they entered. Then, they're dropped to a "post-time conference" for about 10 seconds, then disconnected. This is an example of a more elaborate system.

exten => _X!,1,Dial(Local/${EXTEN}@drum,,m(ringback)g) ; drum announcement
	same => n,Goto(somewhere-else,s,1)

[drum]
exten => _X!,1,Set(GROUP(drumintercept)=1)
	same => n,Set(CDR_PROP(disable)=1)
	same => n,Set(CONFBRIDGE(user,wait_marked)=yes)
	same => n,Set(CONFBRIDGE(user,end_marked)=yes)
	same => n,Set(CONFBRIDGE(user,dtmf_passthrough)=no)
	same => n,Set(CONFBRIDGE(user,startmuted)=yes)
	same => n,Set(CONFBRIDGE(user,dsp_drop_silence)=yes)
	same => n,GotoIf($[${GROUP_COUNT(1@drumintercept)}=1]?startannouncement:waitloop)
	same => n(startannouncement),Originate(Local/1@drum-announcement,exten,announcement-access,1,,a)
	same => n,Wait(0.1)
	same => n(waitloop),GotoIf($["${CONFBRIDGE_INFO(parties,drumintercept1)}"="0"]?startannouncement) ; if nobody is in the conference, start it up
	same => n,WaitForCondition(#,#[#{GROUP_COUNT(1@interceptcutin)}>0])
	same => n(bridge),Answer()
	same => n,ConfBridge(drumintercept1,silentbridge,incogtalkeroptimized)
	same => n,Set(GROUP(drumintercept)=)
	same => n,Hangup()

[announcement-access]
exten => 1,1,Set(CONFBRIDGE(user,marked)=yes)
	same => n,Set(CDR_PROP(disable)=1)
	same => n,ConfBridge(drumintercept1,silentbridge,incogtalkeroptimized)
	same => n,Hangup()

[announcement-monitor]
exten => 1,1,Answer()
	same => n,Set(CDR_PROP(disable)=1)
	same => n,ConfBridge(drumintercept1,silentbridge,incoglistener)
	same => n,Hangup()

[drum-announcement]
exten => 1,1,Answer()
	same => n,Set(CDR_PROP(disable)=1)
	same => n,Wait(${RAND(2,3)})
	same => n,Set(GROUP(interceptcutin)=1)
	same => n,Wait(0.85)
	same => n,Set(GROUP(interceptcutin)=)
	same => n,Set(VOLUME(TX)=2)
	same => n,Playback(custom/intercept-announcement,noanswer)
	same => n,Wait(3)
	same => n,System(asterisk -rx "confbridge kick drumintercept1 all") ; you could kicked all the unmarked users here, instead of all the users
	same => n,Hangup() ; hang up, and kick out everyone presently in the bridge
			

Here is an example of a more featured intercept with configurable options, which does not supervise (provided that the answer_channel = no user option is present, either added at runtime or in the user template used):

	same => n,Gosub(intercept-drum,s,1(sample,ringback,1,0,0)) ; announcement
	same => n,Hangup()

[intercept-drum-announcements]
exten => sample,lookup,custom/myintercept

; a dial is not used, which means this works without supervising! Actually ringing the special op. will mean that we need FRAME_DROP.
[intercept-drum] ; ARG1 = intercept, ARG2 = ringback MOH class, ARG3 = # of iterations (0 = forever)
				; ARG4 = 0 = wait for next iteration to cut in, 1 = cut in immediately, ARG5 = talkable intercept: 0 = no, 1 = yes
exten => s,1,Progress()
	same => n,StartMusicOnHold(${ARG2})
	same => n,Set(CONFBRIDGE(user,template)=incogtalkeroptimized)
	same => n,Set(CONFBRIDGE(user,wait_marked)=yes)
	same => n,Set(CONFBRIDGE(user,end_marked)=yes)
	same => n,Set(CONFBRIDGE(user,dtmf_passthrough)=no)
	same => n,ExecIf($["${ARG5}"!="1"]?Set(CONFBRIDGE(user,startmuted)=yes))
	same => n,Set(CONFBRIDGE(user,dsp_drop_silence)=yes)
	; if this seems backwards, it is. But we want variables set on the control channel, not the bridge channel.
	same => n,ExecIf($[${GROUP_COUNT(${ARG1}@interceptdrumon)}=0]?Originate(Local/${ARG1}@intercept-drum-announce,exten,intercept-drum-announcement,${ARG1},1,,av(anycut=${ARG4})))
	same => n,Set(GROUP(interceptdrumlisten)=${ARG1})
	same => n,Wait(0.05) ; wait for the machine to "start up"...
	same => n,Set(c=0)
	same => n,WaitForCondition(#,#[#{GROUP_COUNT(${ARG1}@interceptcutin)}>0],,0.2)
	same => n,StopMusicOnHold() ; ConfBridge marked user wait is useless. It still adds you to the bridge in a janky way.
	same => n,While($["${ARG3}"="0"|${INC(c)}<=${ARG3}])
	same => n,ConfBridge(interceptdrum${ARG1},silentbridge)
	same => n,EndWhile()
	same => n,Set(GROUP(interceptdrumlisten)=)
	same => n,Return()

[intercept-drum-announcement]
exten => _[0-9A-Za-z].,1,Set(CDR_PROP(disable)=1)
	same => n,ExecIf($[${GROUP_COUNT(${EXTEN}@interceptdrumon)}>0]?Hangup)
	same => n,Set(GROUP(interceptdrumon)=${EXTEN})
	same => n,Answer()
	same => n,Wait(${RAND(1,3)}) ; takes a moment to "start up"
	same => n(cutin),Set(GROUP(interceptcutin)=${EXTEN})
	same => n,Wait(0.5) ; cut in waiting callers. In theory, we could put everyone in a "waiting" ConfBridge, but that's probably even more CPU.
	same => n,ExecIf($["${anycut}"!="1"]?Set(GROUP(interceptcutin)=)) ; now, nobody else can join until the next loop, unless that's permitted.
	same => n,Playback(${LOOKUP(${EXTEN}@intercept-drum-announcements)})
	same => n,Wait(0.4)
	same => n,ConfKick(interceptdrum${EXTEN},participants) ; kick all current listeners to the intercept if it's 1x only.
	same => n,Wait(0.4)
	same => n,GotoIf($[${GROUP_COUNT(${EXTEN}@interceptdrumlisten)}>0]?cutin) ; callers are waiting for this intercept, cut them in
	same => n,Hangup()

[intercept-drum-announce]
exten => _[0-9A-Za-z].,1,Set(CDR_PROP(disable)=1)
	same => n,Set(CONFBRIDGE(user,template)=incogtalkeroptimized)
	same => n,Set(CONFBRIDGE(user,marked)=yes)
	same => n,Answer()
	same => n,ConfBridge(interceptdrum${EXTEN},silentbridge)
	same => n,Hangup()

[intercept-drum-monitor]
exten => _[0-9A-Za-z].,1,Set(CDR_PROP(disable)=1)
	same => n,Answer()
	same => n,ConfBridge(interceptdrum${EXTEN},silentbridge,incoglistener)
	same => n,Hangup()
			

Crosstalk

One would think that ChanSpy() could be used for this kind of thing, but ChanSpy is, for these purposes, almost completely useless, due to a long persistent limitation with audiohooks in Asterisk.

You will need to use ConfBridge. The non-supervising ConfBridge patches are a must-have pre-req. Once you have that, the mechanism is as follows:

• "Regenerate" the call. We use "regenerate" as pseudo-jargon to precisely mean: generating, via a call file (not via Originate(), due to the significant limitations of this application), a near-identical copy of the current call. This means all channel variables, and of course, the Caller ID, name, and presentation, as well as anything else you want to carry forward henceforth.
• Drop the original call into a ConfBridge(). Not supervised, so there won't be any answer supervision sent to the caller. The regenerated call is what's actually going to go to the destination, and this will enter the bridge as well.
• Use the G option for Dial() to fork the call on answer supervision. Once the called party answers, split the caller and callee paths. We don't need the caller path anymore, so get rid of it (Hangup), though be careful with the h extension! Next, set a flag to indicate the call has been answered (e.g. by using a global variable or a DB entry), then drop the original call from the ConfBridge. In the dialplan, it will check whether the call has been answered using that aforeset flag, and if it has, it will Answer() the call and then re-enter the bridge. (Basically, we kick the caller out of the bridge to provide him with answer supervision, then drop him back in.)
• All in the meanwhile, since we now have a ConfBridge, you can mix whatever audio you want into the bridge. For crosstalk, there's a fairly logical way to do this. You could have unique crosstalk per call or shared crosstalk across all calls, and source this crosstalk from audio files, other conference bridges, etc.

There is more involved, of course. Each call effectively has a unique conference bridge. Hangup handling using the h extension is an art that must be carefully mastered. Because the calling party is no longer directly connected to the called party, you will need to manually check for different cases and kick the other party if one has left. And if you forget to do this in all the required places, then you may find that you have "ghost" bridges with nobody in them piling up as calls take place. When a call is completed, the conference bridge should get destroyed.

Because there are some people interested in this concept, a subset of the crosstalk/carrier code used on the main tandem is provided below for reference.

;;; Carrier Code:

[carriersetup]
exten => in,1,Set(CDR_PROP(disable)=1)
	same => n,Gosub(carrierlookup,${ARG1},1)
	same => n,ExecIf($[${GOSUB_RETVAL}=0]?Return)
	same => n,Set(LOCAL(channel)=${FILTER(0-9,${UNIQUEID})})
	same => n,Set(__callednumber=${ARG1}) ; do we really need this?
	same => n,ExecIf($[${GROUP_COUNT(1@carriercalls)}=0]?DBdeltree(carrierchannels))
	same => n,Set(GROUP(carriercalls)=1)
	same => n,Set(GROUP(carriercrosstalk)=1)
	same => n,Set(DB(carrierchannels/${channel})=${CHANNEL})
	same => n,ExecIf($[${GROUP_COUNT(0@carriercrosstalk)}=0]?Originate(Local/s@carriercrosstalk,app,Wait,9999999,,,av(CDR_PROP(disable)=1^CALLERID(num)=${CALLERID(num)}^CALLERID(name)=${CALLERID(name)})))
	same => n,Set(randl=BCDE) ; Subtle L-Carrier track options (we'll randomly pick one)
	same => n,Set(LOCAL(carriergroup)=${randl:$[${RAND(1,${LEN(${randl})})}-1]:1})
	same => n,Set(SHARED(carriergroup)=${carriergroup})
	same => n,Originate(Local/${carriergroup}@carriernoise,exten,carrierinject,${channel},1,,av(CALLERID(num)=${CALLERID(num)}^CALLERID(name)=${CALLERID(name)})) ; Main L-Carrier
	same => n,Originate(Local/monitor@carriercrosstalk,exten,carrierinject,${channel},1,,av(CALLERID(num)=${CALLERID(num)}^CALLERID(name)=${CALLERID(name)})) ; Crosstalk
	same => n,Return(${channel})
exten => out,1,Set(CDR_PROP(disable)=1)
	same => n,Gosub(carrierlookup,${ARG1},1)
	same => n,ExecIf($[${GOSUB_RETVAL}=0]?Return)
	same => n,Set(LOCAL(channel)=${FILTER(0-9,${UNIQUEID})})
	same => n,Set(__callednumber=${ARG1}) ; do we really need this?
	same => n,ExecIf($[${GROUP_COUNT(1@carriercalls)}=0]?DBdeltree(carrierchannels))
	same => n,Set(GROUP(carriercalls)=1)
	same => n,Set(GROUP(carriercrosstalk)=1)
	same => n,Set(DB(carrierchannels/${channel})=${CHANNEL})
	same => n,ExecIf($[${GROUP_COUNT(0@carriercrosstalk)}=0]?Originate(Local/s@carriercrosstalk,app,Wait,9999999,,,av(CDR_PROP(disable)=1^CALLERID(num)=${CALLERID(num)}^CALLERID(name)=${CALLERID(name)})))
	same => n,Set(randl=BCDE) ; Subtle L-Carrier track options (we'll randomly pick one)
	same => n,Set(LOCAL(carriergroup)=${randl:$[${RAND(1,${LEN(${randl})})}-1]:1})
	same => n,Set(SHARED(carriergroup)=${carriergroup})
	same => n,Originate(Local/${channel}@carrierinject,exten,carriernoise,${carriergroup},1,,av(CALLERID(num)=${CALLERID(num)}^CALLERID(name)=${CALLERID(name)})) ; Main L-Carrier
	same => n,Originate(Local/monitor@carriercrosstalk,exten,carrierinject,${channel},1,,av(CALLERID(num)=${CALLERID(num)}^CALLERID(name)=${CALLERID(name)})) ; Crosstalk
	same => n,Return(${channel})

[regeneratecall] ; ARG1 = local channel ID, ARG2 = local channel context, ARG3 = context, ARG4 = in/out, ARG5 = called #
exten => s,1,Originate(Local/${ARG1}@${ARG2},exten,${ARG3},${ARG5},1,,av(CALLERID(num)=${CALLERID(num)}^CALLERID(name)=${CALLERID(name)}^CALLERID(pres)=${CALLERID(pres)}^calldirection=${ARG4}^__channel=${ARG1}^__autovonchan=${autovonchan}^__clidverif=${clidverif}^__mlppdigit=${mlppdigit}^__forcesecurechannel=${forcesecurechannel}^__dtlocation=${dtlocation}))
	same => n,Return(${ARG1})

[carrierbridge]
exten => _X!,1,Set(channel=${EXTEN}) ; if caller hangs up, end call
	same => n,Set(CDR_PROP(disable)=1)
	same => n,Progress()
	same => n,Set(carriergroup=${SHARED(carriergroup,${DB(carrierchannels/${channel})})})
	same => n,ExecIf($["${carriergroup}"!="B"]?Set(VOLUME(TX)=-${RAND(7,9)}))
	same => n,ExecIf($["${carriergroup}"!="B"]?Playback(custom/switch/connect/carrierconnect1,noanswer))
	same => n,ExecIf($["${carriergroup}"!="B"]?Set(VOLUME(TX)=1))
	same => n,ConfBridge(channel${channel},silentbridge,incogtalker) ; this MUST NOT ANSWER the call
	same => n,ExecIf($["${SHARED(carrieranswered,${DB(carrierchannels/${channel})})}"="1"]?Answer():Hangup) ; prevent answering the call if it wasn't... actually answered!
	same => n,Set(SHARED(carrieranswered,${DB(carrierchannels/${channel})})=2) ; this is the ${channel} channel so we don't need to explicitly specify it with SHARED
	same => n,ConfBridge(channel${channel},silentbridge,incogtalker)
	same => n,Hangup() ; used was kicked (other party hung up)
exten => h,1,ExecIf($[${CONFBRIDGE_INFO(parties,channel${channel})}>0]?ConfKick(channel${channel},all))

[carrieronward]
exten => _X!,1,Set(CDR_PROP(disable)=1)
	same => n,Answer() ; bridge audio to caller immediately
	same => n,Wait(${RAND(1,2)})
	same => n,Set(carriergroup=${SHARED(carriergroup,${DB(carrierchannels/${channel})})})
	same => n,Wait(0.5)
	same => n,ExecIf($["${calldirection}"="out"]?Wait(${RAND(1,${RAND(2,5)})})) ; give time for the L-carrier "B-loop" sound to play
	same => n,ExecIf($["${calldirection}"="in"]?Dial(Local/${EXTEN}@intraoffice-connect,,G(split))) ; channel variables will carry
	same => n,ExecIf($["${calldirection}"="out"]?Dial(Local/${EXTEN}@to-phreaknet,,G(split))) ; channel variables will carry
	same => n,ConfKick(channel${channel},all) ; if called party hangs up without answering, end call
	same => n(caller),Goto(carrieronward-hangup,s,1) ; so h extension doesn't execute
	same => n(split),Goto(caller) ; disconnect caller to get rid of an unnecessary channel
	same => n(callee),Set(SHARED(carrieranswered,${DB(carrierchannels/${channel})})=1)
	same => n,ConfKick(channel${channel},${DB(carrierchannels/${channel})}) ; kick the caller from the bridge, which will Answer() the call and then drop him back in
	same => n,Set(CDR_PROP(disable)=1)
	same => n,Goto(carrierbridgecallee,${channel},1)
exten => h,1,ExecIf($[${CONFBRIDGE_INFO(parties,channel${channel})}>0]?ConfKick(channel${channel},all)) ; if calling party hangs up, end call

[carrieronward-hangup]
exten => s,1,Hangup()

[carrierbridgecallee]
exten => _X!,1,Set(channel=${EXTEN})
	same => n,Set(CDR_PROP(disable)=1)
	same => n,Originate(Local/answer@carriernoise,exten,carrierinject,${channel},1,,a)
	same => n(wait),WaitForCondition(#,#["#{SHARED(carrieranswered,#{DB(carrierchannels/#{channel})})}"="2"],1) ; prevents race conditions, where if the ConfKick before carrieronward,caller has NOT YET taken effect, we don't kill the call (that call must finish before this executes)
	same => n(bridge),ExecIf($[${CONFBRIDGE_INFO(parties,channel${channel})}>0]?ConfBridge(channel${channel},silentbridge,incogtalker))
	same => n,Hangup()
exten => h,1,ExecIf($[${CONFBRIDGE_INFO(parties,channel${channel})}>0]?ConfKick(channel${channel},all)) ; if called party hangs up, end call

[carrierbridgeaccess]
exten => _X!,1,Set(CDR_PROP(disable)=1)
	same => n,Answer()
	same => n,ConfBridge(channel${EXTEN},silentbridge,incogtalker)
	same => n,Hangup()
			

Echo Test

The standard way to create an echo test in Asterisk would be as follows:

exten => 9931,1,Goto(echo,s,1)

[echo]
exten => s,1,Answer() ; Echo test with instructions
	same => n,Playback(demo-echotest)
	same => n,Echo()
	same => n,Playback(demo-echodone)
	same => n,Playback(vm-goodbye)
	same => n,Hangup()
			

We have created a separate context just for the echo test here, as this is generally good practice. Instead of duplicating code should you wish to add another echo test on a different extension, you can simply copy the GoTo statement and send it to the same context.

Note that the use of the s extension is completely arbitrary and could be anything. In macros, you are required to use the s extension, but subroutines often use the start extension instead (but could use anything, including s, as you may see elsewhere in this documentation).

However, this may not be what you want on your node. If you call extension 9931, by dialing, say, 555-9931, you won't immediately be dumped into the echo test. Most Asterisk systems with echo tests use the code above, which plays an introductory message about the echo test first and a concluding message afterward (if you press # as opposed to simply hanging up). If you simply want your extension to go to an echo test (perhaps more realistic if you are trying to stick with the "vintage" theme, you can replace the above [echo] context with the following:

[echo]
exten => s,1,Progress() ; Echo test with no instructions
	same => n,Echo()
	same => n,Hangup()
			

Now, you'll immediately enter the echo test when you call extension 9931, and you won't hear anything if you press "#" before it terminates the call. Here, we use Progress instead of Answer, since an echo test is a test number, so it should not provide answer supervision.

Silent Termination

Creating a silent termination line is incredibly simple:

[silentterm]
exten => start,1,Wait(86400)
	same => n,Return
			

What this subroutine actually does, when called, is wait for 86,400 seconds (or 24 hours), before returning. This way, a caller can't call your silent termination line and stay there forever.

To create a silent termination extension, simply add the following subroutine call to an extension:

exten => 9932,1,Gosub(silentterm,start,1)
	same => n,Hangup
			

Note that this silent termination line will not "supe" (it won't provide answer supervision). You can add an Answer statement first to change that:

exten => 9932,1,Answer
	same => n,Gosub(silentterm,start,1)
	same => n,Hangup
			

Live Feeds

Asterisk allows you to use the mpg123 utility to play MP3 streams from the Internet. Although it is not so popular anymore, in the mid to late-20th century, it was quite common to be able to listen to live music feeds and radio stations using certain access numbers, allowing people to listen to their favorite stations and news from wherever they were (provided they were willing to foot the long-distance charges). In Asterisk, there are two approaches you can take to play an MP3 stream.

The first approach is to define the stream in musiconhold.conf. For our example here, we will create a stream for the KZSU radio station which, as of this writing, features, among other things, a 1950s 701B SxS electromechanical switch in its office. We will also create a stream for KPFA, the flagship station of the Pacifica Radio Network:

[KZSU]
mode=custom
application = /usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://171.66.118.110:8080/kzsu-1-128.mp3

[KPFA]
mode=custom
application = /usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://streams.kpfa.org:8000/kpfa
			

Now, you will need to add the appropriate dialplan code to access the Music on Hold streams you defined:

exten => 5078,1,Answer()
	same => n,Set(TIMEOUT(absolute)=3600)
	same => n,MusicOnHold(KZSU)
	same => n,Hangup()
exten => 5732,1,Answer()
	same => n,Set(TIMEOUT(absolute)=3600)
	same => n,MusicOnHold(KPFA)
	same => n,Hangup()
			

The timeouts you see simply prevent the extension from being tied up forever (even though multiple people can access the stream). After 3600 seconds, or 1 hour, the call will automatically be disconnected. This can help ensure callers don't dial into a stream and let it run for perpetuity.

Now by dialing extension 5078 (or KZSU if dialing by letters) or dialing extension 5732 (or KPFA), you can listen to those respective radio stations.

*Technically, Z is not on the telephone dial, so we have substituted 0 for Z, since the two were equated for ZEnith dialing in many areas.

The downside of this approach is that it doesn't scale well. Every MP3 stream you define in musiconhold.conf is running all the time, 24/7/365. While we don't think this makes much sense (and perhaps constitutes a bug, unintentional or otherwise) in Asterisk, that's how it is. We learned this the hard way when we defined around 100 MP3 streams in musiconhold.conf. As soon we entered moh reload at the Asterisk CLI, the server immediately ground almost to a halt. Needless to say, things were not pleasant. We recommend using this approach only if you have a few streams; if you are looking to have more than 5 or 10 streams, a better approach is to not use musiconhold.conf at all but instead to create the stream on demand directly in the dialplan, like so:

exten => 5078,1,Answer()
	same => n,Set(TIMEOUT(absolute)=3600)
	same => n,MP3Player(http://171.66.118.110:8080/kzsu-1-128.mp3)
	same => n,Hangup()
exten => 5732,1,Answer()
	same => n,Set(TIMEOUT(absolute)=3600)
	same => n,MP3Player(http://streams.kpfa.org:8000/kpfa)
	same => n,Hangup()
			

As you can see, musiconhold.conf is not used at all. The advantage of this is that streams are not running when they are not in use. Resources will only be used to play the stream when there is an active call connected to these extensions.

So, which approach should you use? The downside of the latter approach is that a separate stream instance is created for each caller, even if they are all accessing the same stream. Thus, the approach you implement will depend largely on how and by whom your streams will be used. If you expect a lot of callers will be calling a small number of streams, go with the first method. If you want to create a large number of streams and don't expect much traffic to any single stream in particular, you should consider the second method, since you will not dedicating unnecessary resources to playing the streams when they are not in use.

Finally, worth mentioning is that since the latter approach creates the stream on-demand, rather than accessing an "already-on" stream, there is an additional delay when the stream is called of about 1 second. The delay is not really noticable unless you are listening for it, but you may want to keep this in mind if you need a stream to available the split instant you call it. If this is the case for you, then you will want to define the stream in musiconhold.conf so that it is always playing and immediately accessible. That being said, the discrepancy is only about 1 second, so it is not really worth doing this unless you have a good reason. The former approach is recommended if you only have a few streams and expect high traffic to them, but we personally prefer the latter approach as it allows you to have an unlimited number of streams that, provided they are generally low-traffic, will keep resource utilization to a minimum (i.e. none) when idle.

In other words, if streaming is a big part of what you do, but you don't have too many, define your streams in musiconhold.conf. Otherwise, don't; simply create the streams on demand.

Modems

Contrary to what you may have read elsewhere on the web, you can setup virtual modems in Asterisk! Here, we will set up a virtual modem and use that to create a BBS.

1. Enter the following in the terminal to install Telnet client and server, as well as other utilities you may need:
   sudo apt-get install telnetd -y
sudo apt-get install telnet -y
sudo apt-get install xinetd -y
sudo apt-get install tcpd -y
service xinetd restart
sudo apt-get install lynx -y — text/line mode text browser
   sudo apt-get install less -y — navigation of large outputs
   sudo apt-get install libxml2-utils -y — HTML parser
   sudo apt-get install bc -y — Basic Calculator
   sudo apt-get install sshpass -y — SSH Auto-Password Passer
   sudo apt-get install lynx -y — SSH Auto-Password Passer

If telnet does not work, or later stops working, follow the steps in these two StackOverflow answers:

1. https://stackoverflow.com/a/34390046/6110631
2. https://stackoverflow.com/a/61937840/6110631

Here are the critical steps from the two answers linked above:

1. Add telnet stream tcp nowait telnetd /usr/sbin/tcpd /usr/sbin/in.telnetd to /etc/inetd.conf.
2. Make this your default in /etc/xinetd.conf:

   Simple configuration file for xinetd
   #
   # Some defaults, and include /etc/xinetd.d/
   defaults
   {
   # Please note that you need a log_type line to be able to use log_on_success
   # and log_on_failure. The default is the following :
   # log_type = SYSLOG daemon info
   instances = 60
   log_type = SYSLOG authpriv
   log_on_success = HOST PID
   log_on_failure = HOST
   cps = 25 30
   }
   				

3. sudo /etc/init.d/xinetd restart
4. sudo apt-get install tcpd
2. For this sample, we will be using username "com" and password "com". Wherever you see "com", replace with the info for your system. Enter the following in your terminal:
   nano /etc/issue.net
   Delete the text Debian GNU/Linux 9 and replace it with text like:

   Welcome to the BBS! Login with username "com" and password "com"
   				

   Press CTRL+X to save, then Y to confirm, then press ENTER.
3. Now, create the account "com" with password "com":
   sudo adduser -p $(openssl passwd -l com) com (the first "com" is the password, the second is the password)

To change the password last type sudo passwd com and it will ask for the new password.

To get the user ID of the user, type id -u com.

To make this a passwordless account, first modify /etc/ssh/sshd_config and change #PermitEmptyPasswords no to PermitEmptyPasswords yes
Then, pw usermod com -w none

4. Now, create a blank shell script that will be used as the entry point for loopback Telnet sessions initiated from Asterisk:
   touch /home/com/main.sh
usermod -s /home/com/main.sh com
chmod a+x /home/com/main.sh
   The second line changes the home directory of "com" to the shell script we created, locking the user's connection into that script. The last line sets the proper permissions. At this point, this shell script doesn't do anything; we'll take care of that once the other mechanics are set up properly.

Alternate Instructions: adduser com; sudo chsh -s /home/com/main.sh com

5. The following lines remove the normal login information spew provided to users upon initial connection to the server. This does affect all users, but this provides a necessary improvement in realism:
   touch /home/com/.hushlogin
rm -rf /etc/motd
touch /etc/motd

6. Allow certain system access to the controlled shell environment:

   chmod 777 -R /var/spool/asterisk chmod 777 -R /home/com/log
7. Now the mechanism to initiate a Telnet session from Asterisk has been created. Now, we will install a softmodem in Asterisk that can connect to a shell script via the Telnet mechanisms we just set up:
   wget https://raw.githubusercontent.com/proquar/asterisk-Softmodem/app_softmodem/app_softmodem.c
wget https://code.phreaknet.org/asterisk/app_softmodem.c
mv app_softmodem.c /usr/src/asterisk-13.24.0/apps/
cd /user/src/asterisk-13.24.0/apps/
make apps
make install
service asterisk restart
reboot
8. Now, modify your dialplan to offer callers a way to connect to your softmodem. Here is the dialplan syntax:

   exten => btx,1,Answer()
   	same => n,Softmodem(host, port, options)
   	same => n,Hangup()
   
   Without any arguments the application acts as a V.23 modem and connects to a Telnet server (port 23) on localhost.
   
   Options are:
   	r(...): rx cutoff (dBi, float, default: -35)
   	t(...): tx power (dBi, float, default: -28)
   	v(...): modem version (default: V23):
   			V21        - 300/300 baud
   			V23        - 1200/75 baud
   			Bell103    - 300/300 baud
   			V22        - 1200/1200 baud
   			V22bis     - 2400/2400 baud
   	l or m: least or most significant bit first (default: m)
   	d(...): amount of data bits (5-8, default: 8)
   	s(...): amount of stop bits (1-2, default: 1)
   	u:      Send Ulm Relay Protocol header to Telnet server
   	n:      Send NULL-Byte to modem after carrier detection (Btx specific)
   
   The modem seems to work fine with VOIP as long as you use a codec like G.711 (alaw/ulaw). Please deactivate any echo cancellation you might use.
   					

   In your dialplan, to use a "Bell 103" 300 baud modem, you might use the following:

   exten => 4221,1,Answer()
   	same => n,Softmodem(127.0.0.1,23,v(Bell103)ld(8)s(1)un)
   	same => n,Hangup()
   					

9. Reload your dialplan by typing asterisk -r and then dialplan reload
10. At this point, you will be able to make a call to your Asterisk softmodem, which will establish a loopback Telnet connection that will launch the shell script we created earlier. However, at this point it is still blank, so nothing will happen. From here on, you will need to setup everything else you need and launch it from this main shell script. You can use other kinds of scripts, like PHP scripts to read values from databases and perform external authentication using databases, as well. You could, for example, store usernames and passwords to access your system in a database to which you connect using PHP as opposed to keeping track of authentication information locally. Or, the database could be on the same server. Or, you could not use databases at all. From here, everything is called using shell scripts, so you have the freedom to set things up how you like.
    The tutorial for creating a softmodem ends here. You will need shell scripting experience to code up a BBS. You can call the one created in this example at 564-4221 on PhreakNet.

Fax

Asterisk actually has very good fax support built in (not including the fax stuff that Digium sells), so if you aren't using it, you're missing out.

Receiving faxes in Asterisk is easier than sending them, purely for logistical reasons. It's easy to have an incoming fax emailed to you. It's a bit more involved to receive documents in a compatible format and get them onto the system in a way that they can be sent.

Here are good starting resources for receiving and sending faxes:

• Incoming Fax Handling
• Outgoing Fax Handling

To make incoming faxes behave like voicemails, you can call a System command in the hangup handler to have incoming faxes emailed to you as TIF files.

The link above discussing outgoing faxes discusses an approach for doing this. However, it may be more feasible and conveient to design a secure file upload page than to process incoming emails for fax delivery.

MF/ACTS Detectors

ACTS when used here is an abuse of notation. More properly, they are known as single-slot coin denomination tones and were used in ACTS, among other things.

In December 2019, Asterisk source files, centered around dsp.c, were successfully forked to add MF and ACTS detection support to Asterisk.

Credits: Dylan Cruz, Howard Harte, Joshua Stein, Naveen Albert

Detection support is not integrated, but rather a separate program that can be called at will when a detector is needed. In other words, if you need an ACTS detector, you can bring one into the call and then get rid of it, much like how ACTS detectors are used for PSTN (non-COCOT) payphones.

Thanks to MF and ACTS support, it is now possible to have Project MF in modern Asterisk!

ZIP Download: detect.zip

The ZIP folder contains the following directories:

• arm
• x86

The _arm folder contains code for the ARM architecture (e.g. Raspberry Pi). The x86 folders are for x86/i386.

The contents of the appropriate subfolder of the detect folder should be extracted to /etc/asterisk/scripts/detect. You will need to choose the subfolders specific to your architecture. So, if you are running x86, you would extract detect/x86/mf to /etc/asterisk/scripts/detect/mf — don't preserve the x86 or arm in your file hierarchy — choose the architecture you need and place its subfolders right into /etc/asterisk/scripts/detect/

Note that the detector application is named mf for both the MF and ACTS detectors. Only the folder name distinguishes the two; the executable name should not be changed, and there is no extension.

Usage

ACTS tones are used for payphone coin signalling on modern non-COCOTs, succeeding the gong-style signalling used in 3-slot payphones. MF and 2600 are part of the multifrequency and singlefrequency signaling systems. DP is audible dial pulsing for real-time dial pulsing, which is generally not possible with VoIP connections.

There are three actual detectors that work separately:

• /etc/asterisk/scripts/detect/mf/mf — MF and 2600 Hz detector
• /etc/asterisk/scripts/detect/acts/acts — ACTS detector
• /etc/asterisk/scripts/detect/dp/dp — Dial Pulse detector

The ACTS detector is the simplest to use. For each detected ACTS beep (each of which is worth 5 cents), it returns one $ (dollar-sign) symbol.

Likewise, the dial pulse detector will return one P for each dial pulse.

The MF/2600 detector is one detector that supports multifrequency, single frequency, and 2600 Hz (general) detection. For multifrequency 0 through 9, it returns 0 through 9. For KP, it returns * and for ST, it returns #. For every single instance of 2600, it returns one "S". Thus, one loud constant 2600 burst (such as a trunk reset) returns a single "S"; dial-pulsing 2600 Hz (i.e. single-frequency signaling) at 10pps or 20pps will produce one S per pulse, making it possible to detect single-frequency dialing in addition to single 2600 Hz tones.

Here is a simple subroutine that uses the ACTS detector (on x86/i386 architecture, but that can be changed), to listen for ACTS tones. Notice that it automatically calculates how many "$"s appeared (\x24 is the hex escape code for $).

[acts-read] ; NA 20200104 ; ARG1 = max silence, ARG2 = max duration, RETURNs # of ACTS beeps
exten => s,1,Set(ss=${STRFTIME(${EPOCH},,%s)})
	same => n,Set(file=/tmp/${UNIQUEID}-${ss}.wav)
	same => n,Record(${file},${ARG1},${ARG2},qx) ; q = quiet, x = ignore terminator keys
	same => n,Set(beeps=${SHELL(/etc/asterisk/scripts/detect/acts/acts ${file})})
	same => n,Set(beeps=${FILTER(\x24,${beeps})}) ; \x24 = hex for "$"
	same => n,Set(num=${LEN(${beeps})})
	same => n,NoOp(ACTS: ${beeps} -> ${num})
	same => n,Return(${num})
			

This detector is used on the payphone trunks. So if you're thinking of using this detector to program a payphone controller… it's already been done! Applications of the detectors have already been fairly rigorously exhausted for everyone's benefit, and we've made the applications that make use of these detectors available as well. If you would like to help improve these end-applications or have ideas, please contact the Business Office.

Cadence Plan

The Cadence Plan allows for the use of standardized custom ring cadences across the network. These are used most commonly for party ling (coded) ringing, busy line callback, and priority ring on calls with greater than routine precedence. The cadences and their numbering were carefully chosen to minimize conflicts with the Bellcore specifications while allowing for maximal functionality and accessibility on the network.

ATAs provisioned by the PhreakNet provisioning server are automatically programmed with the cadences below.

Grandstream ATAs typically allow for 10 cadences, while Linksys ATAs typically allow for 8 (sometimes 9) cadences.

If you are using DAHDI, here are the cadence specifications for the [channels] section in chan_dahdi.conf:

[channels]
cadence=2000,-4000
cadence=300,200,1000,200,300,-4000
cadence=800,400,800,-4000
cadence=400,200,300,200,800,-4000
cadence=400,200,800,200,800,-3600
cadence=200,400,200,400,200,-4000
cadence=800,200,400,200,800,-3600
cadence=1650,-350
cadence=800,200,300,200,400,-4000
cadence=400,200,400,-2000

If your ATA uses different formats for specifying cadences, please contact the Business Office and we can work with you to get your ATA set up correctly.

Subroutines like the following may be used to send the right cadence to your ATA:

[SIP-RingHeader] ; ARG1 = cadence / ARG2 = peers
exten => s,1,Set(peerstocheck=${ARG2})
	same => n,Set(i=-1)
	same => n(checkfree),Set(i=${LEN(${CUT(peerstocheck,&,1)})})
	same => n,Set(nextpeer=${peerstocheck:0:${i}})
	;same => n,Set(nextpeer=${nextpeer:4})
	same => n,Set(peerstocheck=${peerstocheck:$[${i}+1]})
	same => n,GotoIf($["${nextpeer:0:4}"="SIP/"]?sip)
	;;;; UNSUPPORTED TECHNOLOGY ;;;;
	same => n,Goto(next)
	
	same => n(sip),Set(nextpeer=${CUT(nextpeer,/,2)})
	same => n,Set(agent=${FILTER(A-Za-z0-9\x20\x2E\x2F\x2D,${SHELL(asterisk -rx 'sip show peer ${nextpeer}' | grep "Useragent" | grep -o ':.*')})})
	same => n,Gosub(SIP-RingHeader-indiv,s,1(${ARG1}, ${agent:1})) ; Trim space that was after the : from agent
	same => n(next),GotoIf($[${LEN(${peerstocheck})}=0]?allchecked:checkfree)
	same => n(allchecked),Return()

[SIP-RingHeader-indiv] ; ARG1 = cadence / ARG2 = SIP ATA user agent
exten => s,1,NoOp(${ARG2}) ; if we know what user agent this is, we'll only send that header
	same => n,GotoIf($[${REGEX("Grandstream" ${ARG2})}=1]?grandstream,1)
	same => n,GotoIf($[${REGEX("Linksys" ${ARG2})}=1]?linksys,1)
	same => n,GotoIf($[${REGEX("OBIHAI" ${ARG2})}=1]?obi,1)
	;same => n,SIPAddHeader(Alert-Info: info=) ; this is a bad fallback
	same => n,Goto(grandstream,1) ; fallback to Grandstream
exten => grandstream,1,SIPAddHeader("Alert-Info:\;info=ring${ARG1}") ; Supports 1-10
	same => n,Return()
exten => linksys,1,SIPAddHeader("Alert-Info:\;info=Bellcore-r${ARG1}") ; Linksys (e.g. PAP2T), etc.
	same => n,Return()
exten => obi,1,SIPAddHeader(Alert-Info: http://127.0.0.1/Bellcore-dr${ARG1:-1}) ; send only a single digit
	same => n,Return()
			

You could then set the cadence by calling it like: same => n,Gosub(SIP-RingHeader,s,1(${EXTEN},${ARG3})) (assuming EXTEN is the ring cadence and ARG3 is the SIP peer).

The SIP header should only get sent once, so if the cadence could be changed before ringing the phone, keep track of that in a separate variable. When determining the cadence to use, consider the priority of the call (MLPP) as well.

[autovoninit] ; Returns number of current trunk calls
exten => _[A-D0],1,NoOp(AUTOVON Precedence: ${EXTEN}) ; AUTOVON checks
	same => n,Gosub(currentldcalls,s,1)
	same => n,GotoIf($[${GOSUB_RETVAL}=0]?:database)
	same => n,DBdeltree(autovonA) ; Reset the key families, to wipe out old, unneeded data (none of which is currently needed)
	same => n,DBdeltree(autovonB)
	same => n,DBdeltree(autovonC)
	same => n,DBdeltree(autovonD)
	same => n,DBdeltree(autovon0)
	same => n(database),Set(GROUP(autovon)=${EXTEN})
	same => n,Set(DB(autovon${EXTEN}/${FILTER(0-9,${UNIQUEID})})=${CHANNEL})
	same => n,Return()

[autovonpreempt]
exten => s,1,GotoIf($["${autovonprioritydigit}"="0"]?done) ; can't preempt any calls
	same => n,Set(callids=${DB_KEYS(autovon0)})
	same => n,GotoIf($[${LEN(${callids})}>0]?0,1)
	same => n(ad),GotoIf($["${autovonprioritydigit}"="D"]?done) ; can't preempt other priority calls
	same => n,Set(callids=${DB_KEYS(autovonD)})
	same => n,GotoIf($[${LEN(${callids})}>0]?D,1)
	same => n(ac),GotoIf($["${autovonprioritydigit}"="C"]?done) ; can't preempt other immediate calls
	same => n,Set(callids=${DB_KEYS(autovonC)})
	same => n,GotoIf($[${LEN(${callids})}>0]?C,1)
	same => n(ab),GotoIf($["${autovonprioritydigit}"="B"]?done) ; can't preempt other flash calls
	same => n,Set(callids=${DB_KEYS(autovonB)}) ; so, yes, only a flash override can get to this point
	same => n,GotoIf($[${LEN(${callids})}>0]?B,1)
	same => n(done),Return() ; Sorry, can't preempt anything
exten => _[A-D0],1,Set(callid=${CUT(callids,\,,1)})
	same => n,Set(preemptchan=${DB(autovon${EXTEN}/${callid})})
	same => n(preemptcall),ChannelRedirect(${preemptchan},autovonpreempted,s,1)
	same => n,GotoIf($["${CHANNELREDIRECT_STATUS}"="SUCCESS"]?success) ; SUCCESS | NOCHANNEL
	same => n,Set(deleted=${DB_DELETE(autovon${EXTEN}/${callid})}) ; call isn't up anymore, so delete from DB
	same => n,Goto(a0) ; try again - hunt for another call to preempt
	same => n(success),Return()

[autovonpreemptfail]
exten => _[ABCD].,1,Progress()
	same => n,Playback(custom/switch/autovon/autovon603&custom/switch/autovon/autovon603,noanswer) ; audio file with 603 intercept
	same => n,Hangup()

[autovonpreempted]
exten => s,1,Set(GLOBAL(autovonpreempted${FILTER(0-9,${UNIQUEID})})=1)
	same => n,Set(GROUP(autovon)=)
	same => n(preempt),PlayTones(440+620) ; can also be defined as preemption in indications.conf
	same => n,Wait(15) ; Play preemption tone to caller for 15s
	same => n,StopPlayTones()
	same => n,Hangup(8) ; AST_CAUSE_PRE_EMPTED

[autovonpreempted-callee]
exten => s,1,Set(CDR_PROP(disable)=1)
	same => n,Hangup() ; this extension is used if none is specified b/c there was no channel (so, no chance of the call being pre-empted, so don't provide the 8 cause code)
exten => _X!,1,Wait(0.3) ; wait ever so slightly, to give the variable time to get set
	same => n,ExecIf($["${$[autovonpreempted${EXTEN}]}"="1"]?PlayTones(440+620):Hangup()) ; can also be defined as preemption in indications.conf
	same => n,Wait(3) ; Play preemption tone to caller for 3s
	same => n,Hangup(8) ; AST_CAUSE_PRE_EMPTED

[currentldcalls]
exten => s,1,Return($[${GROUP_COUNT(A@autovon)}+${GROUP_COUNT(B@autovon)}+${GROUP_COUNT(C@autovon)}+${GROUP_COUNT(D@autovon)}+${GROUP_COUNT(0@autovon)}])
			

#!/usr/bin/perl
use Socket;

open STDOUT, '>/dev/null';
fork and exit;

my $timedata = localtime(time);
my $cidnum = $ARGV[0];
my $cidname = $ARGV[1];
my $extension = $ARGV[2];

my $MSG1 = "STAT CNET Phone Call to $extension";
my $MSG2 = "RING";
my $MSG3 = "NAME $cidname";
my $MSG4 = "TTSN Call from $cidname";
my $MSG5 = "NMBR $cidnum to $extension";
my $MSG6 = "TYPE U";
my $MSG7 = "IDLE $timedata";

my $ipaddr=192.168.1.255;
my $portnum=42685;

socket(SOCKET, PF_INET, SOCK_DGRAM, getprotobyname("udp")) or die "socket: $!";
setsockopt(SOCKET, SOL_SOCKET, SO_BROADCAST, 1) or die "setsockopt: $!\n";
send(SOCKET, $MSG1, 0, sockaddr_in($portnum,$ipaddr)) or die "cannot send to $HOSTNAME($PORTNO): $!";
send(SOCKET, $MSG2, 0, sockaddr_in($portnum,$ipaddr)) or die "cannot send to $HOSTNAME($PORTNO): $!";
send(SOCKET, $MSG3, 0, sockaddr_in($portnum,$ipaddr)) or die "cannot send to $HOSTNAME($PORTNO): $!";
send(SOCKET, $MSG4, 0, sockaddr_in($portnum,$ipaddr)) or die "cannot send to $HOSTNAME($PORTNO): $!";
send(SOCKET, $MSG5, 0, sockaddr_in($portnum,$ipaddr)) or die "cannot send to $HOSTNAME($PORTNO): $!";
send(SOCKET, $MSG6, 0, sockaddr_in($portnum,$ipaddr)) or die "cannot send to $HOSTNAME($PORTNO): $!";
sleep(5);
send(SOCKET, $MSG2, 0, sockaddr_in($portnum,$ipaddr)) or die "cannot send to $HOSTNAME($PORTNO): $!";
sleep(5);
send(SOCKET, $MSG2, 0, sockaddr_in($portnum,$ipaddr)) or die "cannot send to $HOSTNAME($PORTNO): $!";
sleep(5);
send(SOCKET, $MSG7, 0, sockaddr_in($portnum,$ipaddr)) or die "cannot send to $HOSTNAME($PORTNO): $!";
close(SOCKET);
exit;
			

Sounds

In order to make StepNet routing more interesting, some kind of vintage signaling audio should be audible when a StepNet call is being tandemed.

Here are a few extensions that would do the trick:

• MFer
• SFer
• Dial Pulser
• Revertive Pulser

These can all be found in the "Vintage Add-Ons" section of this documentation.

Other audio files needed for StepNet tandems are available as a standardized ZIP file downloadable below. The ZIP file contains all the ULAW files you need; no need for each node owner to convert them using SoX when we can do it once and also reduce the download size!

ZIP file containing standardized StepNet tandem audio files: click here to download.

Extract the audio files inside to the following location: /var/lib/asterisk/sounds/en/custom/stepnet/ (you will need to create the stepnet folder). The files are named sn1, sn2, sn3.

The ${RAND(min,max)} function in Asterisk will allow you to randomly play some of these audio files in the dialplan.

Video SIP Clients

Not all SIP clients support video, but some do. The one that we recommend is MicroSIP. It's one of the most popular softphone applications out there, and it runs on any version of Windows NT.

However, using video with MicroSIP isn't entirely intuitive. The key thing to note is that you need to disable single call mode. Otherwise, there won't be any video support, and there's no hint as to why that is.

If you are using single-call mode, there isn't a button labeled "Video Call" on the pop-out panel, but you can click the video button to the left of the "Call" button after keying in a number. For shortcut buttons, you can change the action from "Call" to "Video Call" to automatically make the INVITE with video support every time (easiest option).

Note that upon doing this, it seems that video support needs to be indicated at the time the SIP INVITE is made to Asterisk. If you're off-hook auto-dialing with MicroSIP, that means as soon as you get dial tone from Asterisk, you need to indicate that you have video support. The remote end doesn't need to send video immediately, but if you don't tell Asterisk you support video from the get go, video support can't be enabled later.

Currently, you need to click "Video Call" in the pop-out panel that seems to only appear after you've made one call when using the application. So make any call using MicroSIP, and then the ancillary window should pop up and you'll see a "Video Call" option. That's what you'll need to allow video.

Finally, you also need to enable video support in the channel driver you are using - whether SIP or PJSIP.

For SIP, add:

videosupport=yes
allow=h264

For PJSIP, add simply:

allow=h264

This needs to be done for any and all peers for which video support is desired.

This allows the H.264 video codec to be used for calls.

Now that you have a SIP client that supports H.264, and Asterisk is configured to allow H.264 for calls, you can take advantage of video in the dialplan. For video to work on a call between two parties, the caller needs to make the call with video support, and the caller needs to receive the SIP INVITE with video support available.

Note that if you use Originate, this means you must explicitly add the h264 codec to the allowed list of codecs (default is simply slin).

If using ConfBridge, the video_mode bridge setting may be useful for controlling video behavior in the bridge.

If video is not working, the first thing to do is get a SIP debug of a call to or from an endpoint. Incoming INVITEs for call originate must have H264 (or the codec being used) in the INVITE. Otherwise, there is no video support. Likewise, calls to endpoints must offer H264. Otherwise, no video support. MicroSIP will have an "Answer with Video" option available if video support is contained in the INVITE on an incoming call.

Incoming Calls

register => 2125551212:password@freedid.ipcomms.net/ipcomms
					

[ipcomms]
secret=password
defaultuser=2125551212
host=freedid.ipcomms.net
allow=ulaw
context=from-ipcomms
dtmfmode=rfc2833
insecure=port,invite
type=peer
directmedia=no
trustrpid=yes
					

[from-ipcomms] ; This is the incoming context for IPComms SIP calls
exten => 2125551212,1,Progress()
	same => n,Set(Var_TO=${CUT(CUT(SIP_HEADER(To),@,1),:,2)}) ; this works with SIP but not PJSIP. PJSIP should go to extension of DNIS.
	same => n,Gosub(pstn-us-verify,s,1)
	;same => n,Answer()
	;same => n,SendDTMF(1) ; for Google Voice
	same => n,Gosub(mfer,start,1(2125551212))
	same => n,Goto(from-pstn,s,1)

[from-pstn]
exten => s,1,Goto(phreaknet-dialtone,${mainphreaknetdisa},1)
					

register => 17775551212:password@callcentric.com/callcentric
					

[callcentric]
type=peer
context=from-callcentric
host=callcentric.com
fromdomain=callcentric.com
defaultuser=17775551212
fromuser=17775551212
secret=password
insecure=port,invite
disallowed_methods=UPDATE
directmedia=no
videosupport=no
disallow=all
allow=ulaw

[callcentric1](callcentric)
host=alpha1.callcentric.com

[callcentric2](callcentric)
host=alpha2.callcentric.com

[callcentric3](callcentric)
host=alpha3.callcentric.com

[callcentric4](callcentric)
host=alpha4.callcentric.com

[callcentric5](callcentric)
host=alpha5.callcentric.com

[callcentric6](callcentric)
host=alpha6.callcentric.com

[callcentric7](callcentric)
host=alpha7.callcentric.com

[callcentric8](callcentric)
host=alpha8.callcentric.com

[callcentric9](callcentric)
host=alpha9.callcentric.com

[callcentric10](callcentric)
host=alpha10.callcentric.com

[callcentric11](callcentric)
host=alpha11.callcentric.com

[callcentric12](callcentric)
host=alpha12.callcentric.com

[callcentric13](callcentric)
host=alpha13.callcentric.com

[callcentric14](callcentric)
host=alpha14.callcentric.com

[callcentric15](callcentric)
host=alpha15.callcentric.com

[callcentric16](callcentric)
host=alpha16.callcentric.com

[callcentric17](callcentric)
host=alpha17.callcentric.com

[callcentric18](callcentric)
host=alpha18.callcentric.com

[callcentric19](callcentric)
host=alpha19.callcentric.com

[callcentric20](callcentric)
host=alpha20.callcentric.com

[callcentricA](callcentric)
host=doll3.callcentric.com

[callcentricB](callcentric)
host=doll4.callcentric.com

[callcentricC](callcentric)
host=doll5.callcentric.com
					

[from-callcentric] ; This is the incoming context for CallCentric SIP calls
exten => callcentric,1,Progress()
	same => n,Set(Var_TO=${CUT(CUT(SIP_HEADER(To),@,1),:,2)}) ; this works with SIP but not PJSIP. PJSIP should go to extension of DNIS.
	same => n,Gosub(pstn-us-verify,s,1)
	;same => n,Wait(4) ; uncomment these if you need support for Google Voice
	;same => n,Answer()
	;same => n,SendDTMF(1)
	same => n,Gosub(mfer,start,1(2125551212))
	same => n,Goto(from-pstn,s,1)
					

Outgoing Calls

[SkyetelCA]
disallow=all
type=peer
insecure=port,invite
host=52.8.201.128
dtmfmode=rfc2833
context=from-skyetel
allow=ulaw
qualify=yes
directmedia=no
directrtpsetup=no
sendrpid=no
canreinvite=no
 
[SkyetelTERM]
disallow=all
type=peer
qualify=yes
insecure=port,invite
host=term.skyetel.com
dtmfmode=rfc2833
context=from-skyetel
allow=ulaw
directmedia=no
directrtpsetup=no
sendrpid=no
canreinvite=no
 
[SkyetelOR]
disallow=all
type=peer
insecure=port,invite
host=52.41.52.34
dtmfmode=rfc2833
context=from-skyetel
allow=ulaw
qualify=yes
directmedia=no
directrtpsetup=no
sendrpid=no
canreinvite=no
 
[SkyetelVA]
disallow=all
host=50.17.48.216
type=peer
allow=ulaw
dtmfmode=rfc2833
insecure=port,invite
context=from-skyetel
qualify=yes
directmedia=no
directrtpsetup=no
sendrpid=no
canreinvite=no
 
[SkyetelHC]
disallow=all
host=hc.skyetel.com
type=peer
allow=ulaw
dtmfmode=rfc2833
insecure=port,invite
context=from-skyetel
directmedia=no
directrtpsetup=no
sendrpid=no
canreinvite=no
 
[SkyetelHC1]
disallow=all
host=52.32.223.28
type=peer
allow=ulaw
dtmfmode=rfc2833
insecure=port,invite
context=from-skyetel
directmedia=no
directrtpsetup=no
sendrpid=no
canreinvite=no
 
[SkyetelHC2]
disallow=all
host=52.4.178.107
type=peer
allow=ulaw
dtmfmode=rfc2833
insecure=port,invite
context=from-skyetel
directmedia=no
directrtpsetup=no
sendrpid=no
canreinvite=no
					

exten => _1NXXXXXXXXX,1,Dial(SIP/SkyetelTERM/${EXTEN},45)
	same => n,Dial(SIP/SkyetelOR/${EXTEN},45)
	same => n,Dial(SIP/SkyetelCA/${EXTEN},45)
	same => n,Dial(SIP/SkyetelVA/${EXTEN},45)
	same => n,Hangup()
					

cd /root
wget http://9142978376.com/newfiles/GV/gvsip-naf.tar
tar xvf gvsip-naf.tar
rm -f gvsip-naf.tar
cd gvsip-naf
./install-gvsip
					

sed -i 's|obihai.telephony.goog|voice.telephony.goog|' /etc/asterisk/pjsip_custom.conf
sed -i 's|obihai.telephony.goog|voice.telephony.goog|' /root/gvsip-naf/install-gvsip
asterisk -rx "core restart when convenient"
					

Automatically blocking spammers with fail2ban/apiban

fail2ban can be used to block SIP spam in Asterisk, which, combined with changing the SIP bindport away from 5060, can very effectively eliminate spam. fail2ban builds on top of iptables by automatically adding iptables DROP entries to your system.

SSLEngine on
SSLCertificateFile	/etc/ssl/private/cisco.crt
SSLCertificateKeyFile /etc/ssl/private/cisco.key
SSLProtocol all
SSLCipherSuite ALL:@SECLEVEL=1 # TLSv1 or better
SSLInsecureRenegotiation on # required for some older ATAs
SSLOptions -StrictRequire +StdEnvVars
SSLVerifyClient require # production
# SSLVerifyClient optional_no_ca # testing
SSLVerifyDepth 2  # no more than 2
SSLCACertificatePath /etc/ssl/private/phone_root_certificates
#SSLCACertificateFile /etc/ssl/private/phone_root_certificates/Sipura_Technology_Client_Root_Authority_1.crt
			

openssl genrsa -out  2048
openssl req -new -key  -out 
			

The certificate signing request process nowadays is automated. Once you've run those commands, navigate to the Cisco portal and upload your CSR. You'll probably want a SHA1 cert, which they still offer as of now, but SHA256 is also available and better for newer ATAs (you could just grab both). The certificate will be valid for 1 year maximum.

Finally, you'll also need the Cisco root certificates to do MTLS and verify clients. You'll need to contact Cisco to get these.

There aren't really any defaults you should rely on for Cisco rules. Instead, you should explicitly provide full URLs, including the protocol and hostname and full request path. There are variables you can use in your rules that will aide in routing, such as $MA for the device's MAC address. See pg. 74 of this provisioning guide for a complete list of macro expansion variables.

That said, newer ATAs such as the SPA-112 may go in this order (depending on Profile Rules):

1. /spa.cfg
2. /spa000a000a00aa.cfg

Cisco provisioning rules work the opposite of Grandstream rules. Grandstream rules will start and go down the list until it receives a valid config, then stop. Cisco will start and go down the list, stopping only when it fails to get a valid config. Thus, rules should go from least-specific to most-specific, which is the opposite order of Grandstream rules. In your bootstrap config, keep this in mind. For permanent, MAC-specific, configs, it really only makes sense to provide one profile rule, since Cisco ATAs will attempt to request all of them, assuming they are all valid.

For MTLS, SSL_CLIENT_S_DN_CN will contain, among other things, the MAC address. SSL_CLIENT_S_DN_ST will contain only the MAC address, all lowercase. This is the case for all Cisco ATAs — they all uniquely identify themselves in their client certificates.

Cisco 79xx phones

The 79xx Cisco series is an older line from Cisco. In comparison to newer SIP hardphones, 79xx phones only supported (unencrypted) TFTP for provisioning, and they can't be provisioned or configured really any other way.

These phones support 3 protocols: SCCP (Skinny), SIP, and MGCP. So there are a few ways you could use them, in increasing order of functionality:

• Use the deprecated, vanilla chan_sip
• Use the core supported, vanilla chan_pjsip
• Use the deprecated, vanilla chan_mgcp
• Use the deprecated, vanilla chan_skinny
• Use the deprecated, vanilla chan_sip with the usecallmanager patches (--cisco option to PhreakScript)
• Use the community chan_sccp (--sccp option to PhreakScript)

Further Resources

• Cisco XML API for Directory

Polycom SoundPoint 331

The Polycom SoundPoint 331 is a SIP hardphone. This guide focuses specifically on this model, but it generalizes to a large number of Polycom phones from this era.

This guide will allow you to bootstrap the phone, e.g. use the phone to allow configuration via the web interface, and then use the web interface to allow configuration using an automated dynamic provisioning method.

1. When the phone is booting up (you cannot do when the phone has booted up), press Setup when it allows you to choose from Start, Setup, etc. Password is 456.
2. Go to DHCP options and use the right-arrow to set from "Option 66 + Static" to "Static".
3. Enter your provisioning server URL under Server Menu.

A Polycom phone will make requests for the following files:

• GET /0004f2ffffff.cfg (if found) OR GET /000000000000.cfg
• GET /000000000000-license.cfg (if found) OR GET /0004f2ffffff-license.cfg
• GET /0004f2ffffff-directory.xml
• PUT /0004f2ffffff-app.log
• GET /SoundPointIPLocalization/English_United_States/SoundPointIP-dictionary.xml

Here, 0004f2ffffff represents the MAC address of the phone, which is all lowercase.

More details here: https://community.polycom.com/t5/VoIP-SIP-Phones/FAQ-What-files-does-my-phone-download-or-upload-and-why/td-p/30582

The Polycom 331 only supports SHA1 certs, not SHA256 - otherwise, you may see something like: ParamParser|Parameter identifier is out of range: "14791"(14791).

4.0.15 is the latest UC firmware supported by the 331. Do NOT use the 4.1.x firmware as this only supports Lync, not SIP! (e.g. the firmware in the GitHub repo below is wrong.)

Helpful Resources

• Polycom UC Firmware Releases (and config samples)
• Admin Guide
• Config file management
• How to manually configure Polycom phones via web interface.
• [FAQ] How can I setup my Phone / Provisioning / Download / Upgrade / Downgrade Software?
• [FAQ] What is the difference between SIP and UCS / UC Software configuration parameters and values?
• Automatically Provisioning Polycom Phones
• How To Provision
• Firmware Binaries
• Could not contact boot server