####################################### # Asterisk & Browser Phone Dockerfile # ####################################### # The easiest way to kick the tires on WebRTC. # Note: Make sure you have Docker installed and running # 1️⃣ Step 1) Build this docker # =========================== # Show the terminal (if you are in Visual Studio Code) and issue the following command: # docker build --tag browser-phone:latest . # (Don't forget to include that little fullstop at the end) # Note: Last time I ran this it took about 15minutes... go take the dog for a walk! # 2️⃣ Step 2) Run this docker # ========================= # In the same terminal, run the following command: # docker run -t -i -p 5060:5060/udp -p 8089:8089/tcp browser-phone:latest # It can take a moment, but this will not be long # 3️⃣ Step 3) Navigate to https://localhost:8089/ # ============================================= # Note: You will probably have Chrome or Edge security warning page up in front of you right now. # This is because the script makes a self-signed localhost certificate and installs it. # It's fine for testing, so just click the "Advanced" button on that page, and click # "Proceed to localhost (unsafe)". (Trust me, this is perfectly safe.) # Your browser phone is up and running... see easy as 1... 2... 3... # 👩🏼‍🦰 Add a User # ============= # I have pre-loaded some users in the config. You will have User1, User2 and User3 all ready for you. # The Browser Phone may be promoting you for account details, They will be as follows: # Secure WebSocket Server (TLS): localhost # WebSocket Port: 8089 # WebSocket Path: /ws # Full Name: User One (or anything you like) # Domain: localhost # SIP Username: User1 (or User2 or User3) # SIP Password: 1234 # Subscribe to VoiceMail (MWI): off # Chat Engine: SIP # That should be fine for you to create an account, so click Save, and let the page reload. # You should see the status go to "Registered" at the top. # Once registered, make a test call to music on hold with the number *65 # Hmmm... good ol elevator music # 🎉 Congratulations!! # ======================================== # Don't change this: FROM debian:bookworm # This is the release version from Asterisk, to change the version, refer to Asterisk GitHub site, and set accordingly. ENV ASTERISK_VERSION=releases/20 # ======================================== ENV DEBIAN_FRONTEND=noninteractive RUN apt-get update RUN apt-get install -y git iputils-ping traceroute WORKDIR /usr/local/src # Download src RUN git clone --branch ${ASTERISK_VERSION} --single-branch --depth 1 https://github.com/asterisk/asterisk.git # Install asterisk WORKDIR /usr/local/src/asterisk RUN contrib/scripts/install_prereq install RUN ./configure RUN make menuselect.makeopts RUN menuselect/menuselect \ --disable BUILD_NATIVE \ --disable-all \ --enable chan_bridge_media \ --enable chan_rtp \ --enable chan_pjsip \ --enable bridge_native_rtp \ --enable bridge_simple \ --enable codec_gsm \ --enable codec_a_mu \ --enable codec_alaw \ --enable codec_ulaw \ --enable codec_opus \ --enable codec_resample \ --enable format_gsm \ --enable format_wav \ --enable format_wav_gsm \ --enable format_pcm \ --enable format_ogg_vorbis \ --enable format_h264 \ --enable format_h263 \ --enable func_base64 \ --enable func_callerid \ --enable func_channel \ --enable func_curl \ --enable func_cut \ --enable func_db \ --enable func_logic \ --enable func_math \ --enable func_sprintf \ --enable func_strings \ --enable app_confbridge \ --enable app_db \ --enable app_dial \ --enable app_echo \ --enable app_exec \ --enable app_mixmonitor \ --enable app_originate \ --enable app_playback \ --enable app_playtones \ --enable app_queue \ --enable app_sendtext \ --enable app_stack \ --enable app_transfer \ --enable app_system \ --enable app_verbose \ --enable app_voicemail \ --enable app_externalivr \ --enable pbx_config \ --enable pbx_realtime \ --enable res_musiconhold \ --enable res_agi \ --enable res_ari \ --enable res_ari_applications \ --enable res_ari_asterisk \ --enable res_ari_bridges \ --enable res_ari_channels \ --enable res_ari_device_states \ --enable res_ari_endpoints \ --enable res_ari_events \ --enable res_ari_mailboxes \ --enable res_ari_model \ --enable res_ari_playbacks \ --enable res_ari_recordings \ --enable res_ari_sounds \ --enable res_clioriginate \ --enable res_config_curl \ --enable res_config_odbc \ --enable res_curl \ --enable res_format_attr_h263 \ --enable res_format_attr_h264 \ --enable res_format_attr_opus \ --enable res_format_attr_vp8 \ --enable res_http_post \ --enable res_http_websocket \ --enable res_odbc \ --enable res_odbc_transaction \ --enable res_parking \ --enable res_pjproject \ --enable res_pjsip \ --enable res_pjsip_acl \ --enable res_pjsip_authenticator_digest \ --enable res_pjsip_caller_id \ --enable res_pjsip_dialog_info_body_generator \ --enable res_pjsip_diversion \ --enable res_pjsip_dlg_options \ --enable res_pjsip_dtmf_info \ --enable res_pjsip_empty_info \ --enable res_pjsip_endpoint_identifier_anonymous \ --enable res_pjsip_endpoint_identifier_ip \ --enable res_pjsip_endpoint_identifier_user \ --enable res_pjsip_exten_state \ --enable res_pjsip_header_funcs \ --enable res_pjsip_logger \ --enable res_pjsip_messaging \ --enable res_pjsip_mwi \ --enable res_pjsip_mwi_body_generator \ --enable res_pjsip_nat \ --enable res_pjsip_notify \ --enable res_pjsip_one_touch_record_info \ --enable res_pjsip_outbound_authenticator_digest \ --enable res_pjsip_outbound_publish \ --enable res_pjsip_outbound_registration \ --enable res_pjsip_path \ --enable res_pjsip_pidf_body_generator \ --enable res_pjsip_pidf_digium_body_supplement \ --enable res_pjsip_pidf_eyebeam_body_supplement \ --enable res_pjsip_publish_asterisk \ --enable res_pjsip_pubsub \ --enable res_pjsip_refer \ --enable res_pjsip_registrar \ --enable res_pjsip_rfc3326 \ --enable res_pjsip_sdp_rtp \ --enable res_pjsip_send_to_voicemail \ --enable res_pjsip_session \ --enable res_pjsip_sips_contact \ --enable res_pjsip_t38 \ --enable res_pjsip_transport_websocket \ --enable res_pjsip_xpidf_body_generator \ --enable res_realtime \ --enable res_rtp_asterisk \ --enable res_sorcery_astdb \ --enable res_sorcery_config \ --enable res_sorcery_memory \ --enable res_sorcery_memory_cache \ --enable res_sorcery_realtime \ --enable res_srtp \ --enable OPTIONAL_API \ --enable MOH-OPSOUND-WAV \ --enable CORE-SOUNDS-EN-WAV \ menuselect.makeopts RUN make all RUN make install RUN make clean # Postinstall RUN chmod -R 750 /var/spool/asterisk RUN rm -rf /var/lib/apt/lists/* RUN rm -rf /usr/local/src/asterisk # Make own samples WORKDIR /etc/asterisk/ COPY Docker/config/* /etc/asterisk/ # Websockets does not work without TLS RUN apt-get install -y openssl RUN mkdir /etc/asterisk/crt RUN openssl req -new -x509 -days 365 -nodes \ -out /etc/asterisk/crt/certificate.pem \ -keyout /etc/asterisk/crt/private.pem \ -subj "/C=GB/ST=England/L=London/O=Head Office/OU=devops/CN=localhost" # Prepare Browser Phone RUN rm -rf /var/lib/asterisk/static-http/* WORKDIR /usr/local/src # Download Browser-Phone RUN git clone https://github.com/InnovateAsterisk/Browser-Phone.git RUN cp -r /usr/local/src/Browser-Phone/Phone/* /var/lib/asterisk/static-http/ # Set HTTP file permissions RUN chmod -R 744 /var/lib/asterisk/static-http/* EXPOSE 5060/udp 8089/tcp HEALTHCHECK --interval=60s --timeout=10s --retries=3 CMD /usr/sbin/asterisk -rx "core show sysinfo" ENTRYPOINT ["/usr/sbin/asterisk","-f"] CMD ["-v"]