Merge pull request #359 from TECH7Fox/up

Update Asterisk to 22.1.0

asterisk/CHANGELOG.md

@@ -2,6 +2,18 @@
 
 # Changelog
 
+## 5.0.0
+
+### Breaking Changes
+
+- `chan_sip` was [removed](https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-chan_sip/) in Asterisk 21, thus it is no longer supported by this add-on.
+  You should [migrate](https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Migrating-from-chan_sip-to-res_pjsip/) to `res_pjsip` if you were using `chan_sip`.
+
+### Upgrades
+
+- Upgrade Asterisk from 20.8.1 to 22.1.0
+- Upgrade debian-base from 7.4.0 to 7.6.2
+
 ## 4.4.0
 
 ### New Features

asterisk/Dockerfile

@@ -1,5 +1,5 @@
 # https://github.com/hassio-addons/addon-debian-base/releases
-FROM ghcr.io/hassio-addons/debian-base:7.6.1 AS base
+FROM ghcr.io/hassio-addons/debian-base:7.6.2 AS base
 
 SHELL [ "/bin/bash", "-euxo", "pipefail", "-c" ]
 
@@ -106,14 +106,14 @@ RUN export DEBIAN_FRONTEND=noninteractive; \
 
 # Taken from https://metadata.ftp-master.debian.org/changelogs/main/a/asterisk/unstable_changelog
 # (replace all ~ with _ and remove final -)
-ARG ASTERISK_OPUS_VERSION="20.8.1_dfsg+_cs6.14.40431414"
+ARG ASTERISK_OPUS_VERSION="22.1.0_dfsg+_cs6.14.60671435"
 WORKDIR /usr/src/asterisk-opus
 RUN curl -fsSL "https://salsa.debian.org/pkg-voip-team/asterisk/-/archive/upstream/${ASTERISK_OPUS_VERSION}/asterisk-upstream-${ASTERISK_OPUS_VERSION}.tar.gz?path=Xopus" | \
     tar --strip-components 2 -xz
 
 # https://github.com/asterisk/asterisk/tags
 # https://www.asterisk.org/downloads/asterisk/all-asterisk-versions/
-ARG ASTERISK_VERSION="20.8.1"
+ARG ASTERISK_VERSION="22.1.0"
 WORKDIR /usr/src/asterisk
 RUN curl -fsSL "http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-${ASTERISK_VERSION}.tar.gz" | \
     tar --strip-components 1 -xz
@@ -182,8 +182,7 @@ RUN \
         --enable chan_bridge_media \
         --enable chan_iax2 \
         --enable chan_pjsip \
-        --enable chan_rtp \
-        --enable chan_sip ; \
+        --enable chan_rtp; \
     # enable good things \
     menuselect/menuselect --enable BETTER_BACKTRACES menuselect.makeopts; \
     # applications \
@@ -212,7 +211,6 @@ RUN \
     menuselect/menuselect --disable pbx_spool menuselect.makeopts; \
     # resource modules \
     menuselect/menuselect --disable res_config_sqlite3 menuselect.makeopts; \
-    menuselect/menuselect --disable res_monitor menuselect.makeopts; \
     menuselect/menuselect --disable res_adsi menuselect.makeopts; \
     # utilities \
     menuselect/menuselect --disable astdb2sqlite3 menuselect.makeopts; \
@@ -241,7 +239,7 @@ RUN \
     # install example configuration \
     DESTDIR="${INSTALL_DIR}" make samples
 
-## Install chan-gongle
+## Install chan-dongle
 WORKDIR /usr/src/asterisk-chan-dongle
 # https://github.com/wdoekes/asterisk-chan-dongle/commits/master
 ARG CHAN_DONGLE_VERSION="13450af5f648ddf4bc279c08e53917f503253bdd"

asterisk/config.yaml

@@ -1,5 +1,7 @@
 name: Asterisk
-version: 4.4.0
+version: 5.0.0
+breaking_versions:
+  - 5.0.0
 slug: asterisk
 description: PBX server for SIP devices like doorbells and phones
 url: https://github.com/TECH7Fox/asterisk-hass-addons

asterisk/rootfs/etc/asterisk/extensions.conf

@@ -11,9 +11,6 @@ exten => 444,1,Park(,s)
 exten => 555,1,ParkedCall(default,701)
 ;
 exten => _X!,1,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})})
-;
-;extension for dialing chan_sip devices.
-;exten => _X!,1,Dial(SIP/${EXTEN})
 
 [parkedcallstimeout]
 exten => s,1,Hangup()

asterisk/rootfs/etc/asterisk/modules.conf

@@ -8,4 +8,3 @@ noload => res_hep_rtcp.so
 noload => app_voicemail_imap.so
 noload => app_voicemail_odbc.so
 noload => chan_dongle.so ; comment out this line if you want to use chan-dongle
-noload => chan_sip.so ; comment out this line if you want to use chan_sip

asterisk/rootfs/etc/cont-init.d/asterisk.sh

@@ -167,16 +167,6 @@ bashio::var.json \
         -template "${tempio_dir}/pjsip_default.conf.gtpl" \
         -out "${etc_asterisk}/pjsip_default.conf"
 
-rm -f "${etc_asterisk}/sip_default.conf"
-bashio::var.json \
-    auto_add "^${auto_add}" \
-    auto_add_secret "${auto_add_secret}" \
-    video_support "^${video_support}" \
-    persons "^${persons}" |
-    tempio \
-        -template "${tempio_dir}/sip_default.conf.gtpl" \
-        -out "${etc_asterisk}/sip_default.conf"
-
 rm -f "${etc_asterisk}/asterisk_mbox.ini"
 bashio::var.json \
     port "$(bashio::config 'mailbox_port')" \