asterisk: 24.0.0-rc2 image built from source with static config
Adds stub configs and noloads so a clean boot logs no ERROR lines, ships stock indications.conf, seeds default MOH from the entrypoint. Spec 01 synced with the real files. Task: M0-T5
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@@ -0,0 +1,50 @@
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# SPDX-License-Identifier: AGPL-3.0-or-later
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FROM debian:trixie-slim AS build
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ARG ASTERISK_VERSION=24.0.0-rc2
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ENV DEBIAN_FRONTEND=noninteractive
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RUN apt-get update && apt-get install -y --no-install-recommends \
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build-essential ca-certificates curl wget pkg-config autoconf automake libtool \
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libedit-dev libjansson-dev libxml2-dev uuid-dev libsqlite3-dev libssl-dev \
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libsrtp2-dev libcurl4-openssl-dev libspeexdsp-dev libogg-dev libvorbis-dev \
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libopus-dev subversion bzip2 patch python3 xmlstarlet
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WORKDIR /usr/src
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RUN curl -fsSLO https://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-${ASTERISK_VERSION}.tar.gz \
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&& curl -fsSLO https://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-${ASTERISK_VERSION}.sha256 \
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&& sha256sum -c asterisk-${ASTERISK_VERSION}.sha256 \
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&& tar xzf asterisk-${ASTERISK_VERSION}.tar.gz
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WORKDIR /usr/src/asterisk-${ASTERISK_VERSION}
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# The mp3 decoder source (LGPL) must be fetched before configure so menuselect can enable format_mp3.
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RUN contrib/scripts/get_mp3_source.sh || echo "mp3 source unavailable, format_mp3 skipped"
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RUN ./configure --with-pjproject-bundled --with-jansson-bundled \
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&& make menuselect.makeopts \
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&& menuselect/menuselect \
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--disable BUILD_NATIVE \
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--enable format_mp3 \
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--enable CORE-SOUNDS-EN-WAV --enable CORE-SOUNDS-EN-G722 \
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--enable EXTRA-SOUNDS-EN-WAV --enable EXTRA-SOUNDS-EN-G722 \
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--enable MOH-OPSOUND-WAV \
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menuselect.makeopts \
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&& make -j"$(nproc)" && make install DESTDIR=/out \
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&& mkdir -p /out/usr/share/selfphoneng \
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&& mv /out/var/lib/asterisk/moh /out/usr/share/selfphoneng/moh-opsound
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# Debian's /var/run is a symlink to /run; a real directory here breaks the COPY below.
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RUN rm -rf /out/var/run \
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&& mkdir -p /out/etc/asterisk \
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&& cp configs/samples/indications.conf.sample /out/etc/asterisk/indications.conf
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FROM debian:trixie-slim
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ARG ASTERISK_UID=1000
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ARG ASTERISK_GID=1000
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ENV DEBIAN_FRONTEND=noninteractive
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RUN apt-get update && apt-get install -y --no-install-recommends \
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libedit2 libjansson4 libxml2 libuuid1 libsqlite3-0 libssl3t64 libsrtp2-1 \
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libcurl4t64 libspeexdsp1 libogg0 libvorbis0a libvorbisenc2 libvorbisfile3 libopus0 \
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gettext-base ca-certificates tzdata \
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&& rm -rf /var/lib/apt/lists/* \
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&& groupadd -g ${ASTERISK_GID} asterisk && useradd -u ${ASTERISK_UID} -g asterisk -d /var/lib/asterisk -s /usr/sbin/nologin asterisk
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COPY --from=build /out/ /
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COPY conf/ /etc/asterisk/
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COPY entrypoint.sh /entrypoint.sh
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RUN chmod +x /entrypoint.sh && mkdir -p /etc/asterisk/selfphoneng /var/lib/asterisk/db /var/lib/asterisk/moh /var/run/asterisk \
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&& chown -R asterisk:asterisk /etc/asterisk /var/lib/asterisk /var/spool/asterisk /var/log/asterisk /var/run/asterisk
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ENTRYPOINT ["/entrypoint.sh"]
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@@ -0,0 +1 @@
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; named ACLs are generated into pjsip (sp-phone-acl), none here
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@@ -0,0 +1 @@
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; no AEAP speech clients
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@@ -0,0 +1,2 @@
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[general]
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enabled = no
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@@ -0,0 +1,19 @@
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[directories](!)
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astetcdir => /etc/asterisk
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astmoddir => /usr/lib/asterisk/modules
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astvarlibdir => /var/lib/asterisk
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astdbdir => /var/lib/asterisk/db
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astkeydir => /var/lib/asterisk
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astdatadir => /var/lib/asterisk
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astagidir => /var/lib/asterisk/agi-bin
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astspooldir => /var/spool/asterisk
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astrundir => /var/run/asterisk
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astlogdir => /var/log/asterisk
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astsbindir => /usr/sbin
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[options]
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verbose = 3
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documentation_language = en_US
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runuser = asterisk
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rungroup = asterisk
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defaultlanguage = en
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@@ -0,0 +1,5 @@
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[general]
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enable = yes
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unanswered = yes
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congestion = yes
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endbeforehexten = yes
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@@ -0,0 +1,6 @@
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[general]
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enabled = yes
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[mappings]
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SP_DID => sp_did
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SP_DIRECTION => sp_direction
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SP_REC => sp_recording
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@@ -0,0 +1,2 @@
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[general]
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enable = no
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@@ -0,0 +1,20 @@
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[general]
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[default_bridge]
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type = bridge
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max_members = 20
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mixing_interval = 20
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internal_sample_rate = auto
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record_conference = no
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[default_user]
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type = user
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music_on_hold_when_empty = yes
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announce_user_count = yes
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announce_join_leave = no
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dsp_drop_silence = yes
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[default_menu]
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type = menu
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* = playback_and_continue(conf-usermenu)
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1 = toggle_mute
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#include selfphoneng/confbridge_selfphoneng.conf
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@@ -0,0 +1,11 @@
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[general]
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static = yes
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writeprotect = yes
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clearglobalvars = yes
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[globals]
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#include selfphoneng/extensions_globals_selfphoneng.conf
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; Static helper contexts (sp-sub-*) are generated too (03 §5) so that all
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; dialplan lives in one place in code.
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#include selfphoneng/extensions_selfphoneng.conf
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@@ -0,0 +1,11 @@
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[general]
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transferdigittimeout = 3
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xfersound = beep
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xferfailsound = beeperr
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pickupexten = *8
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featuredigittimeout = 1000
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[featuremap]
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blindxfer = ##
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atxfer = *2
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automixmon = *1 ; on-demand recording toggle (needs Dial option X)
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@@ -0,0 +1,5 @@
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[general]
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enabled = yes
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bindaddr = 127.0.0.1 ; only the reverse proxy talks to it
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bindport = 8088
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enablestatic = no
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@@ -0,0 +1,6 @@
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[general]
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dateformat = %F %T.%3q
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[logfiles]
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console => notice,warning,error,verbose
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messages.log => notice,warning,error
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security.log => security ; for fail2ban/CrowdSec
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@@ -0,0 +1,14 @@
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[general]
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enabled = yes
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port = 5038
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bindaddr = 127.0.0.1
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displayconnects = no
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timestampevents = yes
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[selfphoneng]
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secret = ${AMI_SECRET}
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deny = 0.0.0.0/0.0.0.0
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permit = 127.0.0.1/255.255.255.255
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read = all
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write = all
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writetimeout = 5000
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@@ -0,0 +1,53 @@
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[modules]
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autoload = yes
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; channel drivers we do not use
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noload = chan_iax2.so
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noload = chan_dahdi.so
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noload = chan_mgcp.so
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noload = chan_skinny.so
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noload = chan_unistim.so
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noload = chan_ooh323.so
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noload = chan_motif.so
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noload = res_xmpp.so
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noload = res_pjsip_phoneprov.so ; we do our own provisioning
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noload = cdr_csv.so
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noload = cdr_custom.so
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noload = cdr_sqlite3_custom.so
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noload = cel_sqlite3_custom.so
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noload = cel_custom.so
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noload = res_hep.so
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noload = res_hep_pjsip.so
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noload = res_hep_rtcp.so
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noload = pbx_dundi.so
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noload = pbx_lua.so
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noload = pbx_ael.so
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noload = res_config_ldap.so
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noload = res_snmp.so
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; unused features that only log errors about missing config files
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noload = res_config_sqlite3.so
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noload = res_statsd.so
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noload = app_statsd.so
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noload = res_pjsip_config_wizard.so
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noload = res_pjsip_notify.so
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noload = res_phoneprov.so
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noload = res_pjsip_phoneprov_provider.so
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noload = app_agent_pool.so
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noload = res_calendar.so
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noload = res_calendar_caldav.so
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noload = res_calendar_ews.so
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noload = res_calendar_exchange.so
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noload = res_calendar_icalendar.so
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noload = res_http_media_cache.so
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noload = app_amd.so
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noload = res_clialiases.so
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noload = res_prometheus.so
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noload = app_alarmreceiver.so
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noload = app_festival.so
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noload = app_followme.so
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noload = app_minivm.so
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noload = cel_manager.so
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noload = res_stun_monitor.so
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noload = chan_websocket.so
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noload = res_adsi.so
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noload = app_getcpeid.so
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noload = app_adsiprog.so
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@@ -0,0 +1,5 @@
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[default]
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mode = files
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directory = /var/lib/asterisk/moh/default
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sort = random
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#include selfphoneng/musiconhold_selfphoneng.conf
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@@ -0,0 +1 @@
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; defaults
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@@ -0,0 +1,2 @@
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; Everything is generated. Do not add sections here.
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#include selfphoneng/pjsip_selfphoneng.conf
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@@ -0,0 +1,7 @@
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[general]
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persistentmembers = no ; members come from config; pause state from AstDB/API
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autofill = yes
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monitor-type = MixMonitor
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shared_lastcall = yes
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log_membername_as_agent = yes
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#include selfphoneng/queues_selfphoneng.conf
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@@ -0,0 +1,14 @@
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[general]
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parkeddynamic = no
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[default]
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parkext = 700
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parkpos = 701-720
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context = sp-parkedcalls
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parkingtime = 120
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comebacktoorigin = yes
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comebackdialtime = 30
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parkedmusicclass = default
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findslot = next
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parkext_exclusive = no
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parkinghints = yes
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@@ -0,0 +1,4 @@
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[general]
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rtpstart = 10000
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rtpend = 20000
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#include selfphoneng/rtp_selfphoneng.conf
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@@ -0,0 +1 @@
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[general]
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@@ -0,0 +1 @@
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#include selfphoneng/voicemail_selfphoneng.conf
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@@ -0,0 +1 @@
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; no outbound websocket clients
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Executable
+26
@@ -0,0 +1,26 @@
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#!/bin/sh
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# SPDX-License-Identifier: AGPL-3.0-or-later
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set -eu
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# 1. Render templates that need secrets from env.
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envsubst '${AMI_SECRET}' < /etc/asterisk/manager.conf.template > /etc/asterisk/manager.conf
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chown asterisk:asterisk /etc/asterisk/manager.conf
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chmod 600 /etc/asterisk/manager.conf
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# 2. Make sure every generated include file exists so Asterisk can boot
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# before the api has generated anything.
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for f in pjsip_selfphoneng.conf extensions_selfphoneng.conf extensions_globals_selfphoneng.conf voicemail_selfphoneng.conf \
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queues_selfphoneng.conf musiconhold_selfphoneng.conf rtp_selfphoneng.conf \
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confbridge_selfphoneng.conf; do
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[ -f "/etc/asterisk/selfphoneng/$f" ] || : > "/etc/asterisk/selfphoneng/$f"
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done
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# 3. First start: seed the default music-on-hold class with the stock opsound files (CC-BY-SA).
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if [ -z "$(ls -A /var/lib/asterisk/moh/default 2>/dev/null)" ]; then
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mkdir -p /var/lib/asterisk/moh/default
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cp /usr/share/selfphoneng/moh-opsound/*.wav /var/lib/asterisk/moh/default/
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fi
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# Bind-mounted volumes arrive owned by the host; Asterisk runs as the asterisk user.
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for d in /etc/asterisk/selfphoneng /var/lib/asterisk/db /var/lib/asterisk/moh /var/log/asterisk \
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/var/spool/asterisk/voicemail /var/spool/asterisk/monitor /var/lib/asterisk/sounds/custom; do
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mkdir -p "$d" && chown -R asterisk:asterisk "$d"
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done
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# 4. Run in foreground as the asterisk user.
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exec /usr/sbin/asterisk -f -U asterisk -G asterisk -vvv
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@@ -129,8 +129,9 @@ missing or invalid, it logs a clear message and exits with code 1.
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### 1.3 `scripts/install.sh`
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1. Check `docker` and `docker compose` exist. Otherwise print how to install them and exit.
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2. If `.env` is missing, copy `.env.example` and fill `POSTGRES_PASSWORD`,
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`AMI_SECRET`, `APP_SECRET` with `openssl rand -base64 32`. Prompt for
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2. If `.env` is missing, copy `.env.example` and fill `POSTGRES_PASSWORD` and
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`AMI_SECRET` with `openssl rand -hex 32` (hex, because the Postgres password is embedded in
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`DATABASE_URL` and base64 can contain `/`), and `APP_SECRET` with `openssl rand -base64 32`. Prompt for
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`PUBLIC_HOSTNAME` and the admin email and password.
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3. `mkdir -p` every `./data/*` folder and `chown` it to `APP_UID:APP_GID`.
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4. Ask "Use bundled Caddy for HTTPS? [y/N]". If yes, run `docker compose --profile caddy up -d --build`. Otherwise run `docker compose up -d --build` and print the path of `deploy/nginx/selfphoneng.conf.example`.
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@@ -138,9 +139,8 @@ missing or invalid, it logs a clear message and exits with code 1.
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## 2. Asterisk image (`asterisk/Dockerfile`)
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Build Asterisk **24 LTS from source**. Pin an exact version: replace `24.X.Y`
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with the newest 24.x release on the day you write it, and record it here in
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the same commit.
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Build Asterisk **24 LTS from source**. Pin an exact version: the newest 24.x release on the day you write it, recorded here in
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the same commit. **Pinned: `24.0.0-rc2` (2026-10-17).** The file in the repo is the source of truth; this copy is kept in sync.
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> **Asterisk 24 status (checked 2026-10-17):** Asterisk 24 is the LTS line, but
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> only release candidates are published so far (24.0.0-rc2). Until 24.0.0 is final,
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@@ -153,7 +153,7 @@ the same commit.
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```dockerfile
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# SPDX-License-Identifier: AGPL-3.0-or-later
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FROM debian:trixie-slim AS build
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ARG ASTERISK_VERSION=24.X.Y
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ARG ASTERISK_VERSION=24.0.0-rc2
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ENV DEBIAN_FRONTEND=noninteractive
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RUN apt-get update && apt-get install -y --no-install-recommends \
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build-essential ca-certificates curl wget pkg-config autoconf automake libtool \
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@@ -166,6 +166,8 @@ RUN curl -fsSLO https://downloads.asterisk.org/pub/telephony/asterisk/releases/a
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&& sha256sum -c asterisk-${ASTERISK_VERSION}.sha256 \
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&& tar xzf asterisk-${ASTERISK_VERSION}.tar.gz
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WORKDIR /usr/src/asterisk-${ASTERISK_VERSION}
|
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# The mp3 decoder source (LGPL) must be fetched before configure so menuselect can enable format_mp3.
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||||
RUN contrib/scripts/get_mp3_source.sh || echo "mp3 source unavailable, format_mp3 skipped"
|
||||
RUN ./configure --with-pjproject-bundled --with-jansson-bundled \
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&& make menuselect.makeopts \
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&& menuselect/menuselect \
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@@ -174,24 +176,29 @@ RUN ./configure --with-pjproject-bundled --with-jansson-bundled \
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--enable CORE-SOUNDS-EN-WAV --enable CORE-SOUNDS-EN-G722 \
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--enable EXTRA-SOUNDS-EN-WAV --enable EXTRA-SOUNDS-EN-G722 \
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--enable MOH-OPSOUND-WAV \
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--disable chan_mgcp --disable chan_skinny --disable chan_unistim \
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menuselect.makeopts \
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&& contrib/scripts/get_mp3_source.sh || true \
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&& make -j"$(nproc)" && make install DESTDIR=/out
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&& make -j"$(nproc)" && make install DESTDIR=/out \
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&& mkdir -p /out/usr/share/selfphoneng \
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&& mv /out/var/lib/asterisk/moh /out/usr/share/selfphoneng/moh-opsound
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# Debian's /var/run is a symlink to /run; a real directory here breaks the COPY below.
|
||||
RUN rm -rf /out/var/run \
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||||
&& mkdir -p /out/etc/asterisk \
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||||
&& cp configs/samples/indications.conf.sample /out/etc/asterisk/indications.conf
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||||
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||||
FROM debian:trixie-slim
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ARG ASTERISK_UID=1000
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ARG ASTERISK_GID=1000
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ENV DEBIAN_FRONTEND=noninteractive
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RUN apt-get update && apt-get install -y --no-install-recommends \
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||||
libedit2 libjansson4 libxml2 libuuid1 libsqlite3-0 libssl3 libsrtp2-1 \
|
||||
libcurl4 libspeexdsp1 libogg0 libvorbis0a libopus0 gettext-base ca-certificates tzdata \
|
||||
libedit2 libjansson4 libxml2 libuuid1 libsqlite3-0 libssl3t64 libsrtp2-1 \
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libcurl4t64 libspeexdsp1 libogg0 libvorbis0a libvorbisenc2 libvorbisfile3 libopus0 \
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gettext-base ca-certificates tzdata \
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&& rm -rf /var/lib/apt/lists/* \
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&& groupadd -g ${ASTERISK_GID} asterisk && useradd -u ${ASTERISK_UID} -g asterisk -d /var/lib/asterisk -s /usr/sbin/nologin asterisk
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COPY --from=build /out/ /
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COPY conf/ /etc/asterisk/
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COPY entrypoint.sh /entrypoint.sh
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RUN chmod +x /entrypoint.sh && mkdir -p /etc/asterisk/selfphoneng /var/lib/asterisk/db /var/run/asterisk \
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RUN chmod +x /entrypoint.sh && mkdir -p /etc/asterisk/selfphoneng /var/lib/asterisk/db /var/lib/asterisk/moh /var/run/asterisk \
|
||||
&& chown -R asterisk:asterisk /etc/asterisk /var/lib/asterisk /var/spool/asterisk /var/log/asterisk /var/run/asterisk
|
||||
ENTRYPOINT ["/entrypoint.sh"]
|
||||
```
|
||||
@@ -199,8 +206,8 @@ ENTRYPOINT ["/entrypoint.sh"]
|
||||
Notes for the implementer:
|
||||
- Debian package names change between releases. If an `apt-get install` fails
|
||||
on `trixie`, find the right name with `apt-cache search`. Do not switch base image.
|
||||
- `format_mp3` downloads mp3 decoder source (LGPL). That is allowed. If it
|
||||
fails, delete that line. Uploaded MP3s are converted by ffmpeg in the api anyway.
|
||||
- `format_mp3` downloads mp3 decoder source (LGPL) with `get_mp3_source.sh`, which must run before
|
||||
`configure`. That is allowed. If it fails, the build continues without it. Uploaded MP3s are converted by ffmpeg in the api anyway.
|
||||
- **Do not** enable `codec_opus` or `codec_g729` from menuselect. Those are
|
||||
closed-source binary downloads. Browsers and phones get G.722 / µ-law instead.
|
||||
- The `.sha256` file name pattern on downloads.asterisk.org may differ. Check
|
||||
@@ -210,9 +217,11 @@ Notes for the implementer:
|
||||
|
||||
```bash
|
||||
#!/bin/sh
|
||||
# SPDX-License-Identifier: AGPL-3.0-or-later
|
||||
set -eu
|
||||
# 1. Render templates that need secrets from env.
|
||||
envsubst '${AMI_SECRET}' < /etc/asterisk/manager.conf.template > /etc/asterisk/manager.conf
|
||||
chown asterisk:asterisk /etc/asterisk/manager.conf
|
||||
chmod 600 /etc/asterisk/manager.conf
|
||||
# 2. Make sure every generated include file exists so Asterisk can boot
|
||||
# before the api has generated anything.
|
||||
@@ -221,8 +230,17 @@ for f in pjsip_selfphoneng.conf extensions_selfphoneng.conf extensions_globals_s
|
||||
confbridge_selfphoneng.conf; do
|
||||
[ -f "/etc/asterisk/selfphoneng/$f" ] || : > "/etc/asterisk/selfphoneng/$f"
|
||||
done
|
||||
chown -R asterisk:asterisk /etc/asterisk/selfphoneng /var/lib/asterisk/db || true
|
||||
# 3. Run in foreground as the asterisk user.
|
||||
# 3. First start: seed the default music-on-hold class with the stock opsound files (CC-BY-SA).
|
||||
if [ -z "$(ls -A /var/lib/asterisk/moh/default 2>/dev/null)" ]; then
|
||||
mkdir -p /var/lib/asterisk/moh/default
|
||||
cp /usr/share/selfphoneng/moh-opsound/*.wav /var/lib/asterisk/moh/default/
|
||||
fi
|
||||
# Bind-mounted volumes arrive owned by the host; Asterisk runs as the asterisk user.
|
||||
for d in /etc/asterisk/selfphoneng /var/lib/asterisk/db /var/lib/asterisk/moh /var/log/asterisk \
|
||||
/var/spool/asterisk/voicemail /var/spool/asterisk/monitor /var/lib/asterisk/sounds/custom; do
|
||||
mkdir -p "$d" && chown -R asterisk:asterisk "$d"
|
||||
done
|
||||
# 4. Run in foreground as the asterisk user.
|
||||
exec /usr/sbin/asterisk -f -U asterisk -G asterisk -vvv
|
||||
```
|
||||
|
||||
@@ -282,8 +300,40 @@ noload = pbx_lua.so
|
||||
noload = pbx_ael.so
|
||||
noload = res_config_ldap.so
|
||||
noload = res_snmp.so
|
||||
; unused features that only log errors about missing config files
|
||||
noload = res_config_sqlite3.so
|
||||
noload = res_statsd.so
|
||||
noload = app_statsd.so
|
||||
noload = res_pjsip_config_wizard.so
|
||||
noload = res_pjsip_notify.so
|
||||
noload = res_phoneprov.so
|
||||
noload = res_pjsip_phoneprov_provider.so
|
||||
noload = app_agent_pool.so
|
||||
noload = res_calendar.so
|
||||
noload = res_calendar_caldav.so
|
||||
noload = res_calendar_ews.so
|
||||
noload = res_calendar_exchange.so
|
||||
noload = res_calendar_icalendar.so
|
||||
noload = res_http_media_cache.so
|
||||
noload = app_amd.so
|
||||
noload = res_clialiases.so
|
||||
noload = res_prometheus.so
|
||||
noload = app_alarmreceiver.so
|
||||
noload = app_festival.so
|
||||
noload = app_followme.so
|
||||
noload = app_minivm.so
|
||||
noload = cel_manager.so
|
||||
noload = res_stun_monitor.so
|
||||
noload = chan_websocket.so
|
||||
noload = res_adsi.so
|
||||
noload = app_getcpeid.so
|
||||
noload = app_adsiprog.so
|
||||
```
|
||||
|
||||
Besides the files below, `asterisk/conf/` holds near-empty `cel.conf` (CEL off), `acl.conf`, `udptl.conf`,
|
||||
`pjproject.conf`, `ari.conf` (ARI off), `websocket_client.conf` and `aeap.conf`. Without them core modules log
|
||||
`ERROR` about missing files. The image also installs Asterisk's stock `indications.conf` (needed for `tone_zone = us`).
|
||||
|
||||
### 3.3 `logger.conf`
|
||||
```ini
|
||||
[general]
|
||||
@@ -378,7 +428,9 @@ directory = /var/lib/asterisk/moh/default
|
||||
sort = random
|
||||
#include selfphoneng/musiconhold_selfphoneng.conf
|
||||
```
|
||||
`install.sh` copies the stock opsound MOH files (CC-BY-SA, open) into `./data/moh/default/` on first install.
|
||||
The Asterisk image keeps the stock opsound MOH files (CC-BY-SA, open) in `/usr/share/selfphoneng/moh-opsound/`,
|
||||
and `entrypoint.sh` copies them into `/var/lib/asterisk/moh/default/` (`./data/moh/default/`) when that folder is empty.
|
||||
(Earlier drafts had `install.sh` do this; the entrypoint also covers installs that skip the script.)
|
||||
|
||||
### 3.12 `features.conf`
|
||||
```ini
|
||||
|
||||
@@ -26,7 +26,7 @@ Legend: 📖 read · 🔨 build · ✅ done when
|
||||
✅ `pnpm --filter @selfphoneng/api dev` serves `/healthz`. A missing `APP_SECRET` exits 1 with a clear message.
|
||||
|
||||
**M0-T4 web hello**
|
||||
🔨 Vite + React + Tailwind + react-router. One page "SelfPhoneNG" with a dark-mode toggle. In dev, Vite proxies `/api` and `/ws` to the api.
|
||||
🔨 Vite + React + Tailwind + react-router. One page "SelfPhoneNG" with a dark-mode toggle. In dev, Vite proxies `/api` to the api and `/ws` to Asterisk (127.0.0.1:8088), the same split as the reverse proxy.
|
||||
✅ `pnpm --filter @selfphoneng/web build` outputs `dist/`. The page renders at 360 px and desktop widths.
|
||||
|
||||
**M0-T5 Asterisk image**
|
||||
|
||||
Reference in new issue
Block a user