notes:asterisk - pup.casa Docs

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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.

===== Files removed =====

I deleted the config files installed by ''make samples'' that were installed for all modules I explicitly removed:

  * app_skel.conf
  * cdr_adaptive_odbc.conf
  * cdr_beanstalkd.conf
  * cdr_odbc.conf
  * cdr_pgsql.conf
  * cdr_syslog.conf
  * cel_beanstalkd.conf
  * cel_odbc.conf
  * cel_pgsql.conf
  * func_odbc.conf
  * misdn.conf
  * osp.conf
  * oss.conf
  * res_odbc.conf
  * res_pgsql.conf
  * res_snmp.conf
  * sip.conf
  * sip_notify.conf
  * vpb.conf

===== Files edited =====

<code ini cli.conf>
[startup_commands]
;sip set debug on       = yes
;core set verbose 3     = yes
;core set debug 1       = yes
</code>

<code ini codecs.conf>
; custom CELT codec defs. one custom definition per sample rate.
;[celt44]
;type=celt
;samprate=44100
;framesize=480

;[opus]
;type=opus
;max_playback_rate=8000
;fec=no
;packet_loss=10
;complexity=10
;max_bandwidth=48000
;bitrate=auto
;application=voip
;cbr=no
;dtx=no

[plc]
genericplc				=> true
genericplc_on_equal_codecs		=> false

[silk8]
type					= silk
samprate				= 8000
maxbitrate				= 10000
fec					= true
packetloss_percentage			= 10
dtx					= false

[silk12]
type					= silk
samprate				= 12000
maxbitrate				= 12000
fec					= true
packetloss_percentage			= 10
dtx					= false

[silk16]
type					= silk
samprate				= 16000
maxbitrate				= 20000
fec					= true
packetloss_percentage			= 10
dtx					= false

[silk24]
type					= silk
samprate				= 24000
maxbitrate				= 30000
fec					= true
packetloss_percentage			= 10
dtx					= false

[speex]
vbr					=> true
quality					=> 3
complexity				=> 2
enhancement				=> true
vad					=> true
abr					=> 0
vbr_quality				=> 4
dtx					=> false
experimental_rtcp_feedback		=> false
preprocess				=> false
pp_vad					=> false
pp_agc					=> false
pp_agc_level				=> 8000
pp_denoise				=> false
pp_dereverb				=> false
pp_dereverb_decay			=> 0.4
pp_dereverb_level			=> 0.3
</code>

<code ini console.conf>
[general]

[default]
active = no
</code>

<code ini extconfig.conf>
[settings]
</code>

<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>

<code ini hep.conf>
[general]
enabled = no
;capture_address = 1.2.3.4:1234
;capture_password = password
;capture_id = 1234
;uuid_type = call-id | channel
</code>

<code ini http.conf>
[general]
servername		= Puppybarks
enabled			= yes
bindaddr		= 0.0.0.0
bindport		= 8088
tlsenable		= no
;prefix=asterisk
;sessionlimit=100
;session_inactivity=30000
;session_keep_alive=15000
;enable_static=yes
;enable_status=no
;redirect = / /static/config/index.html
;tlsenable=yes
;tlsbindaddr=0.0.0.0:8089
;
;tlscertfile=
;tlsprivatekey=
; tlscipher=
; ECDHE-RSA-AES128-GCM-SHA256:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES256-GCM-SHA384:
; ECDHE-ECDSA-AES256-GCM-SHA384:DHE-RSA-AES128-GCM-SHA256:DHE-DSS-AES128-GCM-SHA256:
; kEDH+AESGCM:ECDHE-RSA-AES128-SHA256:ECDHE-ECDSA-AES128-SHA256:ECDHE-RSA-AES128-SHA:
; ECDHE-ECDSA-AES128-SHA:ECDHE-RSA-AES256-SHA384:ECDHE-ECDSA-AES256-SHA384:
; ECDHE-RSA-AES256-SHA:ECDHE-ECDSA-AES256-SHA:DHE-RSA-AES128-SHA256:DHE-RSA-AES128-SHA:
; DHE-DSS-AES128-SHA256:DHE-RSA-AES256-SHA256:DHE-DSS-AES256-SHA:DHE-RSA-AES256-SHA:
; AES128-GCM-SHA256:AES256-GCM-SHA384:AES128-SHA256:AES256-SHA256:AES128-SHA:AES256-SHA:
; AES:CAMELLIA:DES-CBC3-SHA:!aNULL:!eNULL:!EXPORT:!DES:!RC4:!MD5:!PSK:!aECDH:
; !EDH-DSS-DES-CBC3-SHA:!EDH-RSA-DES-CBC3-SHA:!KRB5-DES-CBC3-SHA
; tlsdisablev1=yes
; tlsdisablev11=yes
; tlsdisablev12=yes
; tlsservercipherorder=yes
;[post_mappings]
;uploads = /var/lib/asterisk/uploads/
</code>

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

<code ini prometheus.conf>
[general]
enabled                 = yes
core_metrics_enabled    = yes
uri                     = metrics
</code>
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