notes:asterisk - pup.casa Docs

Trace: • pup2k • niko • lamer • xfiles • alpine • maffs-ibook • treefiddy • labrador • asterisk
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Alpine Linux latest-stable (written when latest-stable == 3.16.0)

====== Note about missing packages ======

Some Asterisk modules are unavailable due to the lack of available packages on Alpine Linux. These currently are:

  * ''beanstalk'' - Available as a package, but I'm not installing it :)
  * ''codec2'' - Unavailable as a package, but the [[https://github.com/drowe67/codec2|most recent commit]] does compile and install
  * ''iksemel'' (Jabber-specialised XML parser) - Unavailable as a package, most recent source simply doesn't compile, but [[https://github.com/timothytylee/iksemel-1.4|version 1.4]] does compile and install.
  * ''nbs'' (Network Broadcast Sound) - Unavailable as a package, compiling from source doesn't work without patching for ''musl''.
  * ''osptk'' (Open Settlement Protocol) - Unavailable as a package (unsure why), couldn't be bothered finding source and seeing if it compiles.
  * ''radius'' - Installing ''freeradius-dev'' is not sufficient, and I'm not interested in figuring out what package is required for this one.
  * ''resample'' - Installing ''libresample-dev'' is apparently not sufficient, even though the install-prereqs script checks for the header file installed by that package before attempting to build from source.. I do not care enough.
  * ''vpb'' (Voicetronix) - Unavailable as a package, website is confusing, don't care enough.

===== Applications =====

  * ''app_jack'' is unavailable due to resample
  * ''app_osplookup'' is unavailable due to osptk

===== Call Detail Recording =====

  * ''cdr_beanstalkd'' is unavailable due to beanstalk
  * ''cdr_radius'' is unavailable due to radius

===== Channel Event Logging =====

  * ''cel_beanstalkd'' is unavailable due to beanstalk
  * ''cel_radius'' is unavailable due to radius

===== Channel Drivers =====

  * ''chan_misdn'' (ISDN BRI/PRI) is unavailable due to isdnnet, misdn and suppserv
  * ''chan_nbs'' (network broadcast sound) is unavailable due to nbs
  * ''chan_phone'' (linux telephony api) is unavailable due to ixjuser
  * ''chan_vpb'' (Voicetronix) is unavailable due to vpb

===== Resource Modules =====

  * ''res_corosync'' is unavailable due to corosync
  * ''res_timing_kqueue'' is unavailable due to kqueue

====== Install commands ======

Not exactly a ready-to-run script, more a log of all the commands I ran in the pursuit of sippies. During the menuconfig step, I enabled everything possible except for any deprecated modules that aren't enabled by default anyway.

<code bash install.sh>
apk add -t pbx-build-deps alpine-sdk alsa-lib-dev asterisk-openrc \
                          binutils-dev bison bluez-dev bsd-compat-headers \
                          cmake curl-dev \
                          dahdi-linux dahdi-linux-dev dahdi-tools dahdi-tools-dev doxygen \
                          fftw-dev findutils flex freeradius-utils freetds-dev \
                          gmime-dev graphviz-dev gsm-dev \
                          ilbc-dev imap-dev \
                          jansson-dev \
                          libedit-dev libical-dev libpri-dev libresample libsamplerate-dev libsndfile-dev libsrtp-dev libtool libxml2-dev libxslt-dev libzip-dev \
                          lua-dev lua5.2-dev lua5.3-dev lua5.4-dev \
                          mariadb-connector-c-dev mariadb-dev mysql-client \
                          neon-dev net-snmp net-snmp-dev newt-dev \
                          openldap-clients openldap-dev openssl-dev opus-dev opusfile-dev \
                          pjproject-dev popt-dev portaudio-dev postgresql14-client postgresql14-dev \
                          spandsp spandsp-dev speex-dev speexdsp-dev sqlite sqlite-dev \
                          tar tiff-dev \
                          unbound-dev unixodbc-dev uriparser uriparser-dev util-linux-dev \
                          xmlstarlet
mkdir -p /usr/local/src/patches/asterisk /usr/local/src/dist/asterisk /usr/local/src/build/codec2
cd /usr/local/src/dist
git clone https://github.com/drowe67/codec2
git clone https://github.com/timothytylee/iksemel-1.4 iksemel
cd /usr/local/src/build/codec2
cmake /usr/local/src/dist/codec2
make -j5
make install
cd /usr/local/src/dist/iksemel
# not sure what the order of operations is supposed to be here because I ran autoupdate
#  but then got messages from autogen.sh about obsolete things I should fix with autoupdate..
autoupdate
./autogen.sh
./configure
# edit makefile here to remove all variables referencing the doc folder,
#  followed by removing the doc folder from SUBDIRS
#  otherwise make will error out because it wants to build the documentation
# TODO: make a patch for that
vi Makefile
make -j5
make install
cd /usr/local/src/patches/asterisk
wget https://git.alpinelinux.org/aports/plain/main/asterisk/10-musl-mutex-init.patch
wget https://git.alpinelinux.org/aports/plain/main/asterisk/20-musl-astmm-fix.patch
wget https://git.alpinelinux.org/aports/plain/main/asterisk/30-asterisk-mariadb.patch
wget https://git.alpinelinux.org/aports/plain/main/asterisk/40-asterisk-cdefs.patch
wget -O 50-usecallmanager-18.11.3.patch https://raw.githubusercontent.com/usecallmanagernz/patches/master/asterisk/cisco-usecallmanager-18.11.3.patch
cd /usr/local/src/dist/asterisk
wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-18.11.3.tar.gz
cd /usr/local/src/build
tar xzf /usr/local/src/asterisk/dist/asterisk-18.11.3.tar.gz
mv asterisk-18.11.3 asterisk
cd /usr/local/src/build/asterisk
patch --strip=1 < /usr/local/src/patches/asterisk/10-musl-mutex-init.patch
patch --strip=1 < /usr/local/src/patches/asterisk/20-musl-astmm-fix.patch
patch --strip=1 < /usr/local/src/patches/asterisk/30-asterisk-mariadb.patch
patch --strip=1 < /usr/local/src/patches/asterisk/40-asterisk-cdefs.patch
patch --strip=1 < /usr/local/src/patches/asterisk/50-usecallmanager-18.11.3.patch
# Asterisk official documentation makes a big stink about how you should only use their bundled pjproject
#  otherwise you'll have instabilities and whatever, but their bundled pjproject doesn't build cleanly on Alpine
#  so they can stink up somewhere else.
CFLAGS="-DENABLE_SRTP_AES_GCM -DENABLE_SRTP_AES_256" ./configure --without-pjproject-bundled
make menuconfig
./contrib/scripts/get_mp3_source.sh
make -j9
make install samples
addgroup -S asterisk
adduser -S -D -h /var/lib/asterisk -s /sbin/nologin -G asterisk -g asterisk asterisk
addgroup -S dialout
addgroup asterisk dialout
chown -R asterisk:asterisk /run/asterisk
chown -R asterisk:asterisk /var/lib/asterisk
chown -R asterisk:asterisk /var/log/asterisk
chown -R asterisk:asterisk /var/spool/asterisk
</code>

====== Configuration ======

This part was initially extremely daunting. Not helping matters is that the fact that there are no pages online (that I could find) that discuss a viable minimal configuration for a recent Asterisk version. Apparently either nobody is starting fresh with Asterisk in 2022 or nobody is interested in writing about it. Further frustrating matters is that Asterisk's official wiki, a Confluence site, went down shortly after I began the process of setting Asterisk up.

===== modules.conf =====

I started here, and essentially trimmed the config file down to ''autoload=yes'', because I disabled all modules in ''make menuconfig'' that I didn't want to actually use.

<code ini modules.conf>
[modules]
autoload=yes
</code>

===== features.conf =====

I then went here, and I trimmed it down to just the things that seemed relevant.

<code ini features.conf>
[general]
transferdigittimeout	=> 3
xfersound		= beep
xferfailsound		= beeperr
;pickupsound		= beep
;pickupfailsound	= beeperr
featuredigittimeout	= 1000
;recordingfailsound	= beeperr
atxfernoanswertimeout	= 15
atxferdropcall		= no
atxferloopdelay		= 10
atxfercallbackretries	= 2
transferdialattempts	= 3
transferretrysound	= beep
transferinvalidsound	= beeperr

atxferabort		= *1
atxfercomplete		= *2
atxferthreeway		= *3
atxferswap		= *4
pickupexten		= *8

[featuremap]
; requires two channels to be both answered and bridged, chan_local is needed w/ Answer in order to use them while RP is ringing or in progress
disconnect		=> *0 ;requires H or h
automon			=> *1 ;requires W or w
atxfer			=> *2 ;requires T or t
automixmon		=> *3 ;requires X or x
blindxfer		=> #1 ;requires T or t
parkcall		=> #72;requires X or x

[applicationmap]
</code>
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