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安装环境 ubuntu 13.10
前往 https://signalwire.com 注册账号
点击自己头像，选择“Personal Access Tokens” 进入个人访问令牌页面，然后点击“+ Add New” 创建令牌。
Token Name 处输入令牌名称，然后点击 “Generate Token” 生成令牌。
PERSONAL ACCESS TOKEN 下面一串字符，就是令牌，请复制出来并保存好。
echo "netkiller" > /etc/yum/vars/signalwireusername echo "pat_hbpgtYbJHMMS5Wcy868dW1o2" > /etc/yum/vars/signalwiretoken dnf install -y https://$(< /etc/yum/vars/signalwireusername):$(< /etc/yum/vars/signalwiretoken)@freeswitch.signalwire.com/repo/yum/centos-release/freeswitch-release-repo-0-1.noarch.rpm epel-release
查看相关包
[root@netkiller ~]# dnf search freeswitch FreeSWITCH Packages for Enterprise Linux 9 - x86_64 174 kB/s | 752 kB 00:04 FreeSWITCH Packages for Enterprise Linux 9 - x86_64 - Debug 168 kB/s | 752 kB 00:04 FreeSWITCH Packages for Enterprise Linux 9 - x86_64 - Source 174 kB/s | 752 kB 00:04 ========================================================================= Name & Summary Matched: freeswitch ========================================================================== freeswitch.src : FreeSWITCH open source telephony platform freeswitch.x86_64 : FreeSWITCH open source telephony platform freeswitch-application-abstraction.x86_64 : FreeSWITCH mod_abstraction freeswitch-application-avmd.x86_64 : FreeSWITCH voicemail detector freeswitch-application-blacklist.x86_64 : FreeSWITCH blacklist module freeswitch-application-callcenter.x86_64 : FreeSWITCH mod_callcenter Call Queuing Application freeswitch-application-cidlookup.x86_64 : FreeSWITCH mod_cidlookup freeswitch-application-conference.x86_64 : FreeSWITCH mod_conference freeswitch-application-curl.x86_64 : FreeSWITCH mod_curl freeswitch-application-db.x86_64 : FreeSWITCH mod_db freeswitch-application-directory.x86_64 : FreeSWITCH mod_directory freeswitch-application-distributor.x86_64 : FreeSWITCH mod_distributor freeswitch-application-easyroute.x86_64 : FreeSWITCH mod_easyroute freeswitch-application-enum.x86_64 : FreeSWITCH mod_enum freeswitch-application-esf.x86_64 : FreeSWITCH mod_esf freeswitch-application-expr.x86_64 : FreeSWITCH mod_expr freeswitch-application-fifo.x86_64 : FreeSWITCH mod_fifo freeswitch-application-fsk.x86_64 : FreeSWITCH mod_fsk freeswitch-application-fsv.x86_64 : FreeSWITCH mod_fsv freeswitch-application-hash.x86_64 : FreeSWITCH mod_hash freeswitch-application-httapi.x86_64 : FreeSWITCH mod_httapi freeswitch-application-http-cache.x86_64 : FreeSWITCH mod_http_cache freeswitch-application-lcr.x86_64 : FreeSWITCH mod_lcr freeswitch-application-limit.x86_64 : FreeSWITCH mod_limit freeswitch-application-memcache.x86_64 : FreeSWITCH mod_memcache freeswitch-application-mongo.x86_64 : FreeSWITCH mod_mongo freeswitch-application-nibblebill.x86_64 : FreeSWITCH mod_nibblebill freeswitch-application-rad_auth.x86_64 : FreeSWITCH mod_rad_auth freeswitch-application-redis.x86_64 : FreeSWITCH mod_redis freeswitch-application-rss.x86_64 : FreeSWITCH mod_rss freeswitch-application-signalwire.x86_64 : FreeSWITCH mod_signalwire freeswitch-application-sms.x86_64 : FreeSWITCH mod_sms freeswitch-application-snapshot.x86_64 : FreeSWITCH mod_snapshot freeswitch-application-snom.x86_64 : FreeSWITCH mod_snom freeswitch-application-soundtouch.x86_64 : FreeSWITCH mod_soundtouch freeswitch-application-spy.x86_64 : FreeSWITCH mod_spy freeswitch-application-stress.x86_64 : FreeSWITCH mod_stress freeswitch-application-translate.x86_64 : FreeSWITCH mod_translate freeswitch-application-valet_parking.x86_64 : FreeSWITCH mod_valet_parking freeswitch-application-video_filter.x86_64 : FreeSWITCH video filter bugs freeswitch-application-voicemail.x86_64 : FreeSWITCH mod_voicemail freeswitch-application-voicemail-ivr.x86_64 : FreeSWITCH mod_voicemail_ivr freeswitch-asrtts-flite.x86_64 : FreeSWITCH mod_flite freeswitch-asrtts-pocketsphinx.x86_64 : FreeSWITCH mod_pocketsphinx freeswitch-asrtts-tts-commandline.x86_64 : FreeSWITCH mod_tts_commandline freeswitch-asrtts-unimrcp.x86_64 : FreeSWITCH mod_unimrcp freeswitch-codec-bv.x86_64 : BroadVoice16 and BroadVoice32 WideBand Codec support for FreeSWITCH open source telephony platform freeswitch-codec-codec2.x86_64 : Codec2 Narrow Band Codec support for FreeSWITCH open source telephony platform freeswitch-codec-h26x.x86_64 : H.263/H.264 Video Codec support for FreeSWITCH open source telephony platform freeswitch-codec-ilbc.x86_64 : iLCB Codec support for FreeSWITCH open source telephony platform freeswitch-codec-isac.x86_64 : iSAC Codec support for FreeSWITCH open source telephony platform freeswitch-codec-mp4v.x86_64 : MP4V Video Codec support for FreeSWITCH open source telephony platform freeswitch-codec-opus.x86_64 : Opus Codec support for FreeSWITCH open source telephony platform freeswitch-codec-passthru-amr.x86_64 : Pass-through AMR Codec support for FreeSWITCH open source telephony platform freeswitch-codec-passthru-amrwb.x86_64 : Pass-through AMR WideBand Codec support for FreeSWITCH open source telephony platform freeswitch-codec-passthru-g723_1.x86_64 : Pass-through g723.1 Codec support for FreeSWITCH open source telephony platform freeswitch-codec-passthru-g729.x86_64 : Pass-through g729 Codec support for FreeSWITCH open source telephony platform freeswitch-codec-silk.x86_64 : Silk Codec support for FreeSWITCH open source telephony platform freeswitch-codec-siren.x86_64 : Siren Codec support for FreeSWITCH open source telephony platform freeswitch-codec-theora.x86_64 : Theora Video Codec support for FreeSWITCH open source telephony platform freeswitch-config-vanilla.x86_64 : Basic vanilla config set for the FreeSWITCH Open Source telephone platform. freeswitch-database-mariadb.x86_64 : MariaDB native support for FreeSWITCH freeswitch-database-pgsql.x86_64 : PostgreSQL native support for FreeSWITCH freeswitch-debuginfo.x86_64 : Debug information for package freeswitch freeswitch-devel.x86_64 : Development package for FreeSWITCH open source telephony platform freeswitch-endpoint-dingaling.x86_64 : Generic XMPP support for FreeSWITCH open source telephony platform freeswitch-endpoint-portaudio.x86_64 : PortAudio endpoint support for FreeSWITCH open source telephony platform freeswitch-endpoint-rtc.x86_64 : Verto endpoint support for FreeSWITCH open source telephony platform freeswitch-endpoint-rtmp.x86_64 : RTPM Endpoint support for FreeSWITCH open source telephony platform freeswitch-endpoint-skinny.x86_64 : Skinny/SCCP endpoint support for FreeSWITCH open source telephony platform freeswitch-endpoint-verto.x86_64 : Verto endpoint support for FreeSWITCH open source telephony platform freeswitch-event-cdr-mongodb.x86_64 : MongoDB CDR Logger for the FreeSWITCH open source telephony platform freeswitch-event-cdr-pg-csv.x86_64 : PostgreSQL CDR Logger for the FreeSWITCH open source telephony platform freeswitch-event-cdr-sqlite.x86_64 : SQLite CDR Logger for the FreeSWITCH open source telephony platform freeswitch-event-erlang-event.x86_64 : Erlang Event Module for the FreeSWITCH open source telephony platform freeswitch-event-format-cdr.x86_64 : JSON and XML Logger for the FreeSWITCH open source telephony platform freeswitch-event-json-cdr.x86_64 : JSON CDR Logger for the FreeSWITCH open source telephony platform freeswitch-event-multicast.x86_64 : Multicast Event System for the FreeSWITCH open source telephony platform freeswitch-event-radius-cdr.x86_64 : RADIUS Logger for the FreeSWITCH open source telephony platform freeswitch-event-rayo.x86_64 : Rayo (XMPP 3PCC) server for the FreeSWITCH open source telephony platform freeswitch-event-snmp.x86_64 : SNMP stats reporter for the FreeSWITCH open source telephony platform freeswitch-format-local-stream.x86_64 : Local File Streamer for the FreeSWITCH open source telephony platform freeswitch-format-mod-shout.x86_64 : Implements Media Steaming from arbitrary shell commands for the FreeSWITCH open source telephony platform freeswitch-format-native-file.x86_64 : Native Media File support for the FreeSWITCH open source telephony platform freeswitch-format-portaudio-stream.x86_64 : PortAudio Media Steam support for the FreeSWITCH open source telephony platform freeswitch-format-shell-stream.x86_64 : Implements Media Steaming from arbitrary shell commands for the FreeSWITCH open source telephony platform freeswitch-format-ssml.x86_64 : Adds Speech Synthesis Markup Language (SSML) parser format for the FreeSWITCH open source telephony platform freeswitch-format-tone-stream.x86_64 : Implements TGML Tone Generation for the FreeSWITCH open source telephony platform freeswitch-freetdm.x86_64 : Provides a unified interface to hardware TDM cards and ss7 stacks for FreeSWITCH freeswitch-kazoo.x86_64 : Kazoo Module for the FreeSWITCH open source telephony platform freeswitch-lang-de.x86_64 : Provides german language dependend modules and speech config for the FreeSWITCH Open Source telephone platform. freeswitch-lang-en.x86_64 : Provides english language dependent modules and speech config for the FreeSWITCH Open Source telephone platform. freeswitch-lang-es.x86_64 : Provides Spanish language dependend modules and speech config for the FreeSWITCH Open Source telephone platform. freeswitch-lang-fr.x86_64 : Provides french language dependend modules and speech config for the FreeSWITCH Open Source telephone platform. freeswitch-lang-he.x86_64 : Provides hebrew language dependend modules and speech config for the FreeSWITCH Open Source telephone platform. freeswitch-lang-pt.x86_64 : Provides Portuguese language dependend modules and speech config for the FreeSWITCH Open Source telephone platform. freeswitch-lang-ru.x86_64 : Provides russian language dependent modules and speech config for the FreeSWITCH Open Source telephone platform. freeswitch-lang-sv.x86_64 : Provides Swedish language dependend modules and speech config for the FreeSWITCH Open Source telephone platform. freeswitch-lua.x86_64 : Lua support for the FreeSWITCH open source telephony platform freeswitch-perl.x86_64 : Perl support for the FreeSWITCH open source telephony platform freeswitch-python.x86_64 : Python support for the FreeSWITCH open source telephony platform freeswitch-release-repo.noarch : FreeSWITCH Packages for Enterprise Linux repository configuration freeswitch-sounds-en-ca-june.noarch : FreeSWITCH fr-CA June prompts freeswitch-sounds-en-ca-june-16000.noarch : FreeSWITCH 16kHz fr CA June prompts freeswitch-sounds-en-ca-june-32000.noarch : FreeSWITCH 32kHz fr CA June prompts freeswitch-sounds-en-ca-june-48000.noarch : FreeSWITCH 48kHz fr CA June prompts freeswitch-sounds-en-ca-june-8000.noarch : FreeSWITCH 8kHz fr CA June prompts freeswitch-sounds-en-ca-june-all.noarch : FreeSWITCH fr CA June prompts freeswitch-sounds-en-us-allison.noarch : FreeSWITCH en-us Allison prompts freeswitch-sounds-en-us-allison-16000.noarch : FreeSWITCH 16kHz en-us Allison prompts freeswitch-sounds-en-us-allison-32000.noarch : FreeSWITCH 32kHz en-us Allison prompts freeswitch-sounds-en-us-allison-48000.noarch : FreeSWITCH 48kHz en-us Allison prompts freeswitch-sounds-en-us-allison-8000.noarch : FreeSWITCH 8kHz en-us Allison prompts freeswitch-sounds-en-us-allison-all.noarch : FreeSWITCH en-us Allison prompts freeswitch-sounds-en-us-callie.noarch : FreeSWITCH en-us Callie prompts freeswitch-sounds-en-us-callie-16000.noarch : FreeSWITCH 16kHz en-us Callie prompts freeswitch-sounds-en-us-callie-32000.noarch : FreeSWITCH 32kHz en-us Callie prompts freeswitch-sounds-en-us-callie-48000.noarch : FreeSWITCH 48kHz en-us Callie prompts freeswitch-sounds-en-us-callie-8000.noarch : FreeSWITCH 8kHz en-us Callie prompts freeswitch-sounds-en-us-callie-all.noarch : FreeSWITCH en-us Callie prompts freeswitch-sounds-fr-ca-june.noarch : FreeSWITCH fr-CA June prompts freeswitch-sounds-fr-ca-june-16000.noarch : FreeSWITCH 16kHz fr CA June prompts freeswitch-sounds-fr-ca-june-32000.noarch : FreeSWITCH 32kHz fr CA June prompts freeswitch-sounds-fr-ca-june-48000.noarch : FreeSWITCH 48kHz fr CA June prompts freeswitch-sounds-fr-ca-june-8000.noarch : FreeSWITCH 8kHz fr CA June prompts freeswitch-sounds-fr-ca-june-all.noarch : FreeSWITCH fr CA June prompts freeswitch-sounds-music.noarch : FreeSWITCH Music on Hold soundfiles freeswitch-sounds-music-16000.noarch : FreeSWITCH 16kHz Music On Hold soundfiles freeswitch-sounds-music-32000.noarch : FreeSWITCH 32kHz Music On Hold soundfiles freeswitch-sounds-music-48000.noarch : FreeSWITCH 48kHz Music On Hold soundfiles freeswitch-sounds-music-8000.noarch : FreeSWITCH 8kHz Music On Hold soundfiles freeswitch-sounds-pt-BR-karina.noarch : FreeSWITCH pt-BR Karina prompts freeswitch-sounds-pt-BR-karina-16000.noarch : FreeSWITCH 16kHz fr BR Karina prompts freeswitch-sounds-pt-BR-karina-32000.noarch : FreeSWITCH 32kHz fr BR Karina prompts freeswitch-sounds-pt-BR-karina-48000.noarch : FreeSWITCH 48kHz fr BR Karina prompts freeswitch-sounds-pt-BR-karina-8000.noarch : FreeSWITCH 8kHz fr BR Karina prompts freeswitch-sounds-pt-BR-karina-all.noarch : FreeSWITCH fr BR Karina prompts freeswitch-sounds-ru-RU-elena.noarch : FreeSWITCH ru-RU Elena prompts freeswitch-sounds-ru-RU-elena-16000.noarch : FreeSWITCH 16kHz ru-RU Elena prompts freeswitch-sounds-ru-RU-elena-32000.noarch : FreeSWITCH 32kHz ru-RU Elena prompts freeswitch-sounds-ru-RU-elena-48000.noarch : FreeSWITCH 48kHz ru-RU Elena prompts freeswitch-sounds-ru-RU-elena-8000.noarch : FreeSWITCH 8kHz ru-RU Elena prompts freeswitch-sounds-ru-RU-elena-all.noarch : FreeSWITCH ru-RU Elena prompts freeswitch-sounds-sv-se-jakob.noarch : FreeSWITCH sv-se Jakob prompts freeswitch-sounds-sv-se-jakob-16000.noarch : FreeSWITCH 16kHz sv-se jakob prompts freeswitch-sounds-sv-se-jakob-32000.noarch : FreeSWITCH 32kHz sv-se jakob prompts freeswitch-sounds-sv-se-jakob-48000.noarch : FreeSWITCH 48kHz sv-se jakob prompts freeswitch-sounds-sv-se-jakob-8000.noarch : FreeSWITCH 8kHz sv-se jakob prompts freeswitch-sounds-sv-se-jakob-all.noarch : FreeSWITCH sv-se jakob prompts freeswitch-timer-posix.x86_64 : Provides posix timer for the FreeSWITCH Open Source telephone platform. freeswitch-xml-cdr.x86_64 : Provides XML CDR interface for the FreeSWITCH Open Source telephone platform. freeswitch-xml-curl.x86_64 : Provides XML Curl interface for the FreeSWITCH Open Source telephone platform. ============================================================================== Name Matched: freeswitch =============================================================================== freeswitch-format-opusfile.x86_64 : Plays Opus encoded files freeswitch-logger-graylog2.x86_64 : GELF logger for Graylog2 and Logstash ============================================================================= Summary Matched: freeswitch ============================================================================= perl-ESL.x86_64 : The Perl ESL module allows for native interaction with FreeSWITCH over the event socket interface. python-ESL.x86_64 : The Python ESL module allows for native interaction with FreeSWITCH over the event socket interface.
查看版本信息
[root@netkiller ~]# dnf info freeswitch
Last metadata expiration check: 0:03:19 ago on Sat 05 Apr 2025 07:50:56 AM CST.
Available Packages
Name : freeswitch
Version : 1.10.11.release.18
Release : 1.el7
Architecture : src
Size : 59 M
Source : None
Repository : freeswitch
Summary : FreeSWITCH open source telephony platform
URL : http://www.freeswitch.org/
License : MPL1.1
Description : FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice
: and chat driven products scaling from a soft-phone up to a soft-switch. It can be used as a
: simple switching engine, a media gateway or a media server to host IVR applications using
: simple scripts or XML to control the callflow.
:
: We support various communication technologies such as SIP, H.323 and GoogleTalk making
: it easy to interface with other open source PBX systems such as sipX, OpenPBX, Bayonne, YATE or Asterisk.
:
: We also support both wide and narrow band codecs making it an ideal solution to bridge legacy
: devices to the future. The voice channels and the conference bridge module all can operate
: at 8, 16 or 32 kilohertz and can bridge channels of different rates.
:
: FreeSWITCH runs on several operating systems including Windows, Max OS X, Linux, BSD and Solaris
: on both 32 and 64 bit platforms.
:
: Our developers are heavily involved in open source and have donated code and other resources to
: other telephony projects including sipXecs, OpenSER, Asterisk, CodeWeaver and OpenPBX.
Name : freeswitch
Version : 1.10.11.release.18
Release : 1.el7
Architecture : src
Size : 59 M
Source : None
Repository : freeswitch-debuginfo
Summary : FreeSWITCH open source telephony platform
URL : http://www.freeswitch.org/
License : MPL1.1
Description : FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice
: and chat driven products scaling from a soft-phone up to a soft-switch. It can be used as a
: simple switching engine, a media gateway or a media server to host IVR applications using
: simple scripts or XML to control the callflow.
:
: We support various communication technologies such as SIP, H.323 and GoogleTalk making
: it easy to interface with other open source PBX systems such as sipX, OpenPBX, Bayonne, YATE or Asterisk.
:
: We also support both wide and narrow band codecs making it an ideal solution to bridge legacy
: devices to the future. The voice channels and the conference bridge module all can operate
: at 8, 16 or 32 kilohertz and can bridge channels of different rates.
:
: FreeSWITCH runs on several operating systems including Windows, Max OS X, Linux, BSD and Solaris
: on both 32 and 64 bit platforms.
:
: Our developers are heavily involved in open source and have donated code and other resources to
: other telephony projects including sipXecs, OpenSER, Asterisk, CodeWeaver and OpenPBX.
Name : freeswitch
Version : 1.10.11.release.18
Release : 1.el7
Architecture : src
Size : 59 M
Source : None
Repository : freeswitch-source
Summary : FreeSWITCH open source telephony platform
URL : http://www.freeswitch.org/
License : MPL1.1
Description : FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice
: and chat driven products scaling from a soft-phone up to a soft-switch. It can be used as a
: simple switching engine, a media gateway or a media server to host IVR applications using
: simple scripts or XML to control the callflow.
:
: We support various communication technologies such as SIP, H.323 and GoogleTalk making
: it easy to interface with other open source PBX systems such as sipX, OpenPBX, Bayonne, YATE or Asterisk.
:
: We also support both wide and narrow band codecs making it an ideal solution to bridge legacy
: devices to the future. The voice channels and the conference bridge module all can operate
: at 8, 16 or 32 kilohertz and can bridge channels of different rates.
:
: FreeSWITCH runs on several operating systems including Windows, Max OS X, Linux, BSD and Solaris
: on both 32 and 64 bit platforms.
:
: Our developers are heavily involved in open source and have donated code and other resources to
: other telephony projects including sipXecs, OpenSER, Asterisk, CodeWeaver and OpenPBX.
Name : freeswitch
Version : 1.10.11.release.18
Release : 1.el7
Architecture : x86_64
Size : 3.2 M
Source : freeswitch-1.10.11.release.18-1.el7.src.rpm
Repository : freeswitch
Summary : FreeSWITCH open source telephony platform
URL : http://www.freeswitch.org/
License : MPL1.1
Description : FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice
: and chat driven products scaling from a soft-phone up to a soft-switch. It can be used as a
: simple switching engine, a media gateway or a media server to host IVR applications using
: simple scripts or XML to control the callflow.
:
: We support various communication technologies such as SIP, H.323 and GoogleTalk making
: it easy to interface with other open source PBX systems such as sipX, OpenPBX, Bayonne, YATE or Asterisk.
:
: We also support both wide and narrow band codecs making it an ideal solution to bridge legacy
: devices to the future. The voice channels and the conference bridge module all can operate
: at 8, 16 or 32 kilohertz and can bridge channels of different rates.
:
: FreeSWITCH runs on several operating systems including Windows, Max OS X, Linux, BSD and Solaris
: on both 32 and 64 bit platforms.
:
: Our developers are heavily involved in open source and have donated code and other resources to
: other telephony projects including sipXecs, OpenSER, Asterisk, CodeWeaver and OpenPBX.
Name : freeswitch
Version : 1.10.11.release.18
Release : 1.el7
Architecture : x86_64
Size : 3.2 M
Source : freeswitch-1.10.11.release.18-1.el7.src.rpm
Repository : freeswitch-debuginfo
Summary : FreeSWITCH open source telephony platform
URL : http://www.freeswitch.org/
License : MPL1.1
Description : FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice
: and chat driven products scaling from a soft-phone up to a soft-switch. It can be used as a
: simple switching engine, a media gateway or a media server to host IVR applications using
: simple scripts or XML to control the callflow.
:
: We support various communication technologies such as SIP, H.323 and GoogleTalk making
: it easy to interface with other open source PBX systems such as sipX, OpenPBX, Bayonne, YATE or Asterisk.
:
: We also support both wide and narrow band codecs making it an ideal solution to bridge legacy
: devices to the future. The voice channels and the conference bridge module all can operate
: at 8, 16 or 32 kilohertz and can bridge channels of different rates.
:
: FreeSWITCH runs on several operating systems including Windows, Max OS X, Linux, BSD and Solaris
: on both 32 and 64 bit platforms.
:
: Our developers are heavily involved in open source and have donated code and other resources to
: other telephony projects including sipXecs, OpenSER, Asterisk, CodeWeaver and OpenPBX.
Name : freeswitch
Version : 1.10.11.release.18
Release : 1.el7
Architecture : x86_64
Size : 3.2 M
Source : freeswitch-1.10.11.release.18-1.el7.src.rpm
Repository : freeswitch-source
Summary : FreeSWITCH open source telephony platform
URL : http://www.freeswitch.org/
License : MPL1.1
Description : FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice
: and chat driven products scaling from a soft-phone up to a soft-switch. It can be used as a
: simple switching engine, a media gateway or a media server to host IVR applications using
: simple scripts or XML to control the callflow.
:
: We support various communication technologies such as SIP, H.323 and GoogleTalk making
: it easy to interface with other open source PBX systems such as sipX, OpenPBX, Bayonne, YATE or Asterisk.
:
: We also support both wide and narrow band codecs making it an ideal solution to bridge legacy
: devices to the future. The voice channels and the conference bridge module all can operate
: at 8, 16 or 32 kilohertz and can bridge channels of different rates.
:
: FreeSWITCH runs on several operating systems including Windows, Max OS X, Linux, BSD and Solaris
: on both 32 and 64 bit platforms.
:
: Our developers are heavily involved in open source and have donated code and other resources to
: other telephony projects including sipXecs, OpenSER, Asterisk, CodeWeaver and OpenPBX.
安装 freeswitch
安装 compat-openssl10 https://pkgs.org/download/compat-openssl10
[root@netkiller ~]# dnf install -y https://pkgs.sysadmins.ws/el8/extras/x86_64/compat-openssl10-1.0.2o-3.el8.x86_64.rpm [root@netkiller ~]# dnf install -y https://repo.almalinux.org/almalinux/9/CRB/x86_64/os/Packages/libmemcached-awesome-1.1.0-12.el9.x86_64.rpm [root@netkiller ~]# dnf install -y https://cdn.amazonlinux.com/2/core/2.0/x86_64/6b0225ccc542f3834c95733dcf321ab9f1e77e6ca6817469771a8af7c49efe6c/../../../../../blobstore/9696af59b58e65548eb6c3256ef10b139190dee9c3efd8a28602db3497a80441/ldns-1.6.16-10.amzn2.x86_64.rpm [root@netkiller ~]# dnf install -y freeswitch-config-vanilla freeswitch-lang-en freeswitch-sounds-en-us-* freeswitch-sounds-music-* freeswitch-codec-opus freeswitch-lua
启动 freeswitch
[root@netkiller ~]# systemctl enable freeswitch [root@netkiller ~]# systemctl start freeswitch [root@netkiller ~]# systemctl status freeswitch
FireWall Ports Network Protocol Application Protocol Description 1719 UDP H.323 Gatekeeper RAS port 1720 TCP H.323 Call Signaling 3478 UDP STUN service Used for NAT traversal 3479 UDP STUN service Used for NAT traversal 5002 TCP MLP protocol server 5003 UDP Neighborhood service 5060 UDP & TCP SIP UAS Used for SIP signaling (Standard SIP Port, for default Internal Profile) 5070 UDP & TCP SIP UAS Used for SIP signaling (For default "NAT" Profile) 5080 UDP & TCP SIP UAS Used for SIP signaling (For default "External" Profile) 8021 TCP ESL Used for mod_event_socket * 16384-32768 UDP RTP/ RTCP multimedia streaming Used for audio/video data in SIP and other protocols 5066 TCP Websocket Used for WebRTC 7443 TCP Websocket Used for WebRTC
fail2ban 自动拦截恶意注册
firewall-cmd --zone=public --add-port=1719/udp --permanent firewall-cmd --zone=public --add-port=1720/tcp --permanent firewall-cmd --zone=public --add-port=3478-3479/udp --permanent firewall-cmd --zone=public --add-port=5002/tcp --permanent firewall-cmd --zone=public --add-port=5003/udp --permanent firewall-cmd --zone=public --add-port=5060/udp --permanent firewall-cmd --zone=public --add-port=5060/tcp --permanent firewall-cmd --zone=public --add-port=5070/udp --permanent firewall-cmd --zone=public --add-port=5080/udp --permanent firewall-cmd --zone=public --add-port=5006/tcp --permanent firewall-cmd --zone=public --add-port=5007/tcp --permanent firewall-cmd --zone=public --add-port=5008/tcp --permanent firewall-cmd --zone=public --add-port=8021/tcp --permanent firewall-cmd --zone=public --add-port=16384-32768/udp --permanent firewall-cmd --zone=public --add-port=5066/tcp --permanent firewall-cmd --zone=public --add-port=7443/tcp --permanent
重启防火墙
firewall-cmd --reload
查看已开放的端口
firewall-cmd --list-ports
[root@netkiller ~]# cp /etc/freeswitch/vars.xml{,.backup}
随机产生密码
[root@netkiller ~]# randpasswd | cut -c -10 NLYPx9JjSx
修改默认密码，将 1234 改为
<X-PRE-PROCESS cmd="set" data="default_password=1234"/> 改为 <X-PRE-PROCESS cmd="set" data="default_password=NLYPx9JjSx"/>
配置公网IP地址
适用场景，公网IP直接在Linux eth0 网卡上配置。
<X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=stun:stun.freeswitch.org"/> 改为 <X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=host:sip.netkiller.cn"/> 或 <X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=IP地址"/>
前面的基本配置，是物理服务器网卡直接配置公网IP地址，很多云主机采用弹性IP机制，将公网IP映射到云主机上。这种模式就需要用到 NAT
配置方法还是修改 /etc/freeswitch/vars.xml 文件
<X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=autonat:你的公网IP地址"/>
<X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=autonat:121.37.25.251"/> <X-PRE-PROCESS cmd="stun-set" data="external_sip_ip=autonat:121.37.25.251"/>
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,H264,VP8"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,G722,PCMU,PCMA,H264,VP8"/>
<X-PRE-PROCESS cmd="set" data="console_loglevel=4"/> 或 <X-PRE-PROCESS cmd="set" data="console_loglevel=info"/>
样本参考
例 168.1. /etc/freeswitch/vars.xml
这是我的配置仅供参考
[root@netkiller ~]# xmlstarlet ed -d '//comment()' /etc/freeswitch/vars.xml
<?xml version="1.0"?>
<include>
<X-PRE-PROCESS cmd="set" data="default_password=******"/>
<X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>
<X-PRE-PROCESS cmd="set" data="domain=sip.aigcsst.com"/>
<X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
<X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
<X-PRE-PROCESS cmd="set" data="use_profile=external"/>
<X-PRE-PROCESS cmd="set" data="rtp_sdes_suites=AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH"/>
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,H264,VP8"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,G722,PCMU,PCMA,H264,VP8"/>
<X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/>
<X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/>
<X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
<X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=autonat:121.37.215.251"/>
<X-PRE-PROCESS cmd="stun-set" data="external_sip_ip=autonat:121.37.215.251"/>
<X-PRE-PROCESS cmd="set" data="unroll_loops=true"/>
<X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
<X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>
<X-PRE-PROCESS cmd="set" data="call_debug=false"/>
<X-PRE-PROCESS cmd="set" data="console_loglevel=info"/>
<X-PRE-PROCESS cmd="set" data="default_areacode=918"/>
<X-PRE-PROCESS cmd="set" data="default_country=US"/>
<X-PRE-PROCESS cmd="set" data="presence_privacy=false"/>
<X-PRE-PROCESS cmd="set" data="au-ring=%(400,200,383,417);%(400,2000,383,417)"/>
<X-PRE-PROCESS cmd="set" data="be-ring=%(1000,3000,425)"/>
<X-PRE-PROCESS cmd="set" data="ca-ring=%(2000,4000,440,480)"/>
<X-PRE-PROCESS cmd="set" data="cn-ring=%(1000,4000,450)"/>
<X-PRE-PROCESS cmd="set" data="cy-ring=%(1500,3000,425)"/>
<X-PRE-PROCESS cmd="set" data="cz-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="de-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="dk-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="dz-ring=%(1500,3500,425)"/>
<X-PRE-PROCESS cmd="set" data="eg-ring=%(2000,1000,475,375)"/>
<X-PRE-PROCESS cmd="set" data="es-ring=%(1500,3000,425)"/>
<X-PRE-PROCESS cmd="set" data="fi-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="fr-ring=%(1500,3500,440)"/>
<X-PRE-PROCESS cmd="set" data="hk-ring=%(400,200,440,480);%(400,3000,440,480)"/>
<X-PRE-PROCESS cmd="set" data="hu-ring=%(1250,3750,425)"/>
<X-PRE-PROCESS cmd="set" data="il-ring=%(1000,3000,400)"/>
<X-PRE-PROCESS cmd="set" data="in-ring=%(400,200,425,375);%(400,2000,425,375)"/>
<X-PRE-PROCESS cmd="set" data="jp-ring=%(1000,2000,420,380)"/>
<X-PRE-PROCESS cmd="set" data="ko-ring=%(1000,2000,440,480)"/>
<X-PRE-PROCESS cmd="set" data="pk-ring=%(1000,2000,400)"/>
<X-PRE-PROCESS cmd="set" data="pl-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="ro-ring=%(1850,4150,475,425)"/>
<X-PRE-PROCESS cmd="set" data="rs-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="ru-ring=%(800,3200,425)"/>
<X-PRE-PROCESS cmd="set" data="sa-ring=%(1200,4600,425)"/>
<X-PRE-PROCESS cmd="set" data="tr-ring=%(2000,4000,450)"/>
<X-PRE-PROCESS cmd="set" data="uk-ring=%(400,200,400,450);%(400,2000,400,450)"/>
<X-PRE-PROCESS cmd="set" data="us-ring=%(2000,4000,440,480)"/>
<X-PRE-PROCESS cmd="set" data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/>
<X-PRE-PROCESS cmd="set" data="beep=%(1000,0,640)"/>
<X-PRE-PROCESS cmd="set" data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/>
<X-PRE-PROCESS cmd="set" data="df_us_ssn=(?!219099999|078051120)(?!666|000|9\d{2})\d{3}(?!00)\d{2}(?!0{4})\d{4}"/>
<X-PRE-PROCESS cmd="set" data="df_luhn=?:4[0-9]{12}(?:[0-9]{3})?|5[1-5][0-9]{14}|3[47][0-9]{13}|3(?:0[0-5]|[68][0-9])[0-9]{11}|6(?:011|5[0-9]{2})[0-9]{12}|(?:2131|1800|35\d{3})\d{11}"/>
<XX-PRE-PROCESS cmd="set" data="digits_dialed_filter=(($${df_luhn})|($${df_us_ssn}))"/>
<X-PRE-PROCESS cmd="set" data="default_provider=example.com"/>
<X-PRE-PROCESS cmd="set" data="default_provider_username=joeuser"/>
<X-PRE-PROCESS cmd="set" data="default_provider_password=password"/>
<X-PRE-PROCESS cmd="set" data="default_provider_from_domain=example.com"/>
<X-PRE-PROCESS cmd="set" data="default_provider_register=false"/>
<X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/>
<X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1,tlsv1.1,tlsv1.2"/>
<X-PRE-PROCESS cmd="set" data="sip_tls_ciphers=ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH"/>
<X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
<X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
<X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
<X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>
<X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
<X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
<X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
<X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>
<X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_in=3mb"/>
<X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_out=3mb"/>
<X-PRE-PROCESS cmd="set" data="suppress_cng=true"/>
<X-PRE-PROCESS cmd="set" data="rtp_liberal_dtmf=true"/>
<X-PRE-PROCESS cmd="set" data="video_mute_png=$${images_dir}/default-mute.png"/>
<X-PRE-PROCESS cmd="set" data="video_no_avatar_png=$${images_dir}/default-avatar.png"/>
</include>
通常在 /etc/freeswitch/vars.xml 中配置，在 internal.xml 和 external.xml 中通过变量引用
[root@netkiller ~]# cp /etc/freeswitch/sip_profiles/internal.xml{,.backup}
[root@netkiller ~]# cp /etc/freeswitch/sip_profiles/external.xml{,.backup}
[root@netkiller ~]# cp /etc/freeswitch/sip_profiles/internal-ipv6.xml{,.backup}
[root@netkiller ~]# cp /etc/freeswitch/sip_profiles/external-ipv6.xml{,.backup}
NAT 配置 172.16.0.10 替换成公网 IP 地址
[root@netkiller ~]# vim /etc/freeswitch/sip_profiles/internal.xml
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
改为
<param name="ext-rtp-ip" value="autonat:172.16.0.10"/>
<param name="ext-sip-ip" value="autonat:172.16.0.10"/>
[root@netkiller ~]# vim /etc/freeswitch/sip_profiles/external.xml
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
改为
<param name="ext-rtp-ip" value="autonat:172.16.0.10"/>
<param name="ext-sip-ip" value="autonat:172.16.0.10"/>
配置 inbound-bypass-media 可以开启旁路模式，在此模式 下FreeSWITCH 只负责 SIP 信令交换，两个 endpoint 话机 RIP 语音通信是点对点方式。这样比较节省流量，但是国内网络复杂，层层 NAT 常常出现局域网正常，部署到广域网就会失败，现象是拨通之后没有语音。
[root@netkiller ~]# grep -r 'inbound-bypass-media' /etc/freeswitch/sip_profiles/internal.xml
<!--<param name="inbound-bypass-media" value="true"/>-->
[root@netkiller ~]# grep 100rel /etc/freeswitch/sip_profiles/internal.xml
There are known issues (asserts and segfaults) when 100rel is enabled.
It is not recommended to enable 100rel at this time.
<!--<param name="enable-100rel" value="true"/>-->
[root@netkiller ~]# vim /etc/freeswitch/sip_profiles/internal.xml
[root@netkiller ~]# systemctl restart freeswitch
[root@netkiller ~]# grep 100rel /etc/freeswitch/sip_profiles/internal.xml
There are known issues (asserts and segfaults) when 100rel is enabled.
It is not recommended to enable 100rel at this time.
<param name="enable-100rel" value="true"/>
[root@netkiller ~]# cp /etc/freeswitch/autoload_configs/switch.conf.xml{,.backup}
[root@netkiller ~]# xmlstarlet ed -d '//comment()' /etc/freeswitch/autoload_configs/switch.conf.xml
配置交换机名称
<param name="switchname" value="netkiller"/>
RTP 端口范围
<!-- RTP port range -->
<!-- <param name="rtp-start-port" value="16384"/> -->
<!-- <param name="rtp-end-port" value="32768"/> -->
日志级别
<param name="loglevel" value="debug"/>
默认是 :: 表示 ipv6 localhost，如果需要远程访问可以改为 listen-ip 0.0.0.0
[root@netkiller ~]# cat /etc/freeswitch/autoload_configs/event_socket.conf.xml
<configuration name="event_socket.conf" description="Socket Client">
<settings>
<param name="nat-map" value="false"/>
<param name="listen-ip" value="::"/>
<param name="listen-port" value="8021"/>
<param name="password" value="ClueCon"/>
<!--<param name="apply-inbound-acl" value="loopback.auto"/>-->
<!--<param name="stop-on-bind-error" value="true"/>-->
</settings>
</configuration>
[root@netkiller ~]# cat /etc/freeswitch/autoload_configs/event_socket.conf.xml
<configuration name="event_socket.conf" description="Socket Client">
<settings>
<param name="nat-map" value="false"/>
<param name="listen-ip" value="127.0.0.1"/>
<param name="listen-port" value="8021"/>
<param name="password" value="netkiller"/>
<!--<param name="apply-inbound-acl" value="loopback.auto"/>-->
<!--<param name="stop-on-bind-error" value="true"/>-->
</settings>
</configuration>
[root@netkiller ~]# fs_cli -p netkiller
参考 /etc/freeswitch/sip_profiles/external/example.xml 文件
[root@netkiller ~]# cat /etc/freeswitch/sip_profiles/external/example.xml <include> <!--<gateway name="asterlink.com">--> <!--/// account username *required* ///--> <!--<param name="username" value="cluecon"/>--> <!--/// auth realm: *optional* same as gateway name, if blank ///--> <!--<param name="realm" value="asterlink.com"/>--> <!--/// username to use in from: *optional* same as username, if blank ///--> <!--<param name="from-user" value="cluecon"/>--> <!--/// domain to use in from: *optional* same as realm, if blank ///--> <!--<param name="from-domain" value="asterlink.com"/>--> <!--/// account password *required* ///--> <!--<param name="password" value="2007"/>--> <!--/// extension for inbound calls: *optional* same as username, if blank ///--> <!--<param name="extension" value="cluecon"/>--> <!--/// proxy host: *optional* same as realm, if blank ///--> <!--<param name="proxy" value="asterlink.com"/>--> <!--/// send register to this proxy: *optional* same as proxy, if blank ///--> <!--<param name="register-proxy" value="mysbc.com"/>--> <!--/// expire in seconds: *optional* 3600, if blank ///--> <!--<param name="expire-seconds" value="60"/>--> <!--/// do not register ///--> <!--<param name="register" value="false"/>--> <!-- which transport to use for register --> <!--<param name="register-transport" value="udp"/>--> <!--How many seconds before a retry when a failure or timeout occurs --> <!--<param name="retry-seconds" value="30"/>--> <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway --> <!--<param name="caller-id-in-from" value="false"/>--> <!--extra sip params to send in the contact--> <!--<param name="contact-params" value=""/>--> <!-- Put the extension in the contact --> <!--<param name="extension-in-contact" value="true"/>--> <!--send an options ping every x seconds, failure will unregister and/or mark it down--> <!--<param name="ping" value="25"/>--> <!--<param name="cid-type" value="rpid"/>--> <!--rfc5626 : Abilitazione rfc5626 ///--> <!--<param name="rfc-5626" value="true"/>--> <!--rfc5626 : extra sip params to send in the contact--> <!--<param name="reg-id" value="1"/>--> <!--</gateway>--> </include>
配置网关
[root@netkiller ~]# cat > /etc/freeswitch/sip_profiles/external/hamsoverip.xml <<EOF <include> <gateway name="hamsoverip"> <param name="username" value="300177"/> <param name="realm" value="pbx-ap.hamsoverip.com:5160"/> <param name="password" value="178aee323ef95"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> </gateway> </include> EOF
配置拨号规则
[root@netkiller ~]# cat > /etc/freeswitch/dialplan/default/hamsoverip.com.xml <<"EOF" <include> <extension name="hamsoverip.com"> <condition field="destination_number" expression="^300(\d+)$"> <action application="bridge" data="sofia/gateway/hamsoverip/300$1"/> </condition> </extension> </include> EOF
[root@netkiller ~]# cp /etc/freeswitch/dialplan/default.xml{,.backup}
[root@netkiller ~]# vim /etc/freeswitch/dialplan/default.xml
<extension name="Local_Extension">
<condition field="destination_number" expression="^(10[01][0-9])$">
<action application="export" data="dialed_extension=$1"/>
<!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
<action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
<action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
<action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/>
<action application="set" data="ringback=${us-ring}"/>
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="set" data="call_timeout=30"/>
<!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> -->
<action application="set" data="hangup_after_bridge=true"/>
<!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
<action application="set" data="continue_on_fail=true"/>
<action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
<action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
<!--<action application="export" data="nolocal:rtp_secure_media=${user_data(${dialed_extension}@${domain_name} var rtp_secure_media)}"/>-->
<action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
<action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
</condition>
</extension>
默认拨号规则是 1000~1019
<condition field="destination_number" expression="^(10[01][0-9])$">
改为 1000~1099
<condition field="destination_number" expression="^(10[0-9][0-9])$">
分机号段为 1000~1999，2000
<condition field="destination_number" expression="^(1[0-9][0-9][0-9]|2000)$">
30000~39999 | 1000~1019
<condition field="destination_number" expression="^(3\d{4}|10[01][0-9])$">
长度 3 或 4 位数 100~199, 1000~1999
<condition field="destination_number" expression="^(1\d{2,3})$">
例 168.2. 拨号规则，配置两个号段 100~199，1000~1999
拨出规则
[root@netkiller ~]# vim /etc/freeswitch/dialplan/default.xml
<extension name="Local_Extension">
<!-- <condition field="destination_number" expression="^(10[01][0-9])$"> -->
<condition field="destination_number" expression="^(1\d{2,3})$">
<action application="export" data="dialed_extension=$1"/>
<!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
<action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
<action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
<action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/>
<action application="set" data="ringback=${us-ring}"/>
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="set" data="call_timeout=30"/>
<!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> -->
<action application="set" data="hangup_after_bridge=true"/>
<!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
<action application="set" data="continue_on_fail=true"/>
<action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
<action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
<!--<action application="export" data="nolocal:rtp_secure_media=${user_data(${dialed_extension}@${domain_name} var rtp_secure_media)}"/>-->
<action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
<action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
</condition>
</extension>
拨入规则
[root@netkiller ~]# cp /etc/freeswitch/dialplan/public.xml{,.backup}
<extension name="public_extensions">
<!-- <condition field="destination_number" expression="^(10[01][0-9])$"> -->
<condition field="destination_number" expression="^(1\d{2,3})$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
[root@netkiller ~]# cat /etc/freeswitch/autoload_configs/logfile.conf.xml
<configuration name="logfile.conf" description="File Logging">
<settings>
<!-- true to auto rotate on HUP, false to open/close -->
<param name="rotate-on-hup" value="true"/>
</settings>
<profiles>
<profile name="default">
<settings>
<!-- File to log to -->
<!--<param name="logfile" value="/var/log/freeswitch.log"/>-->
<!-- At this length in bytes rotate the log file (0 for never) -->
<param name="rollover" value="1048576000"/>
<!-- Maximum number of log files to keep before wrapping -->
<!-- If this parameter is enabled, the log filenames will not include a date stamp -->
<param name="maximum-rotate" value="32"/>
<!-- Prefix all log lines by the session's uuid -->
<param name="uuid" value="true" />
</settings>
<mappings>
<!--
name can be a file name, function name or 'all'
value is one or more of debug,info,notice,warning,err,crit,alert,all
Please see comments in console.conf.xml for more information
-->
<map name="all" value="console,debug,info,notice,warning,err,crit,alert"/>
</mappings>
</profile>
</profiles>
</configuration>
新增一项 proxy_media 配置
echo '<X-PRE-PROCESS cmd="set" data="proxy_media=true"/>' >> /etc/freeswitch/vars.xml [root@netkiller ~]# echo '<X-PRE-PROCESS cmd="set" data="proxy_media=true"/>' >> /etc/freeswitch/vars.xml [root@netkiller ~]# grep 'proxy_media' /etc/freeswitch/vars.xml <X-PRE-PROCESS cmd="set" data="proxy_media=true"/>
去掉 inbound-proxy-media 注释
[root@netkiller ~]# egrep "inbound-proxy-media|inbound-late-negotiation" /etc/freeswitch/sip_profiles/internal.xml
<!--<param name="inbound-proxy-media" value="true"/>-->
<param name="inbound-late-negotiation" value="true"/>
<!--<param name="inbound-proxy-media" value="true"/>-->
修改
<param name="inbound-proxy-media" value="true"/>
[root@netkiller ~]# egrep "inbound-proxy-media|inbound-late-negotiation" /etc/freeswitch/sip_profiles/internal.xml
<param name="inbound-proxy-media" value="true"/>
<param name="inbound-late-negotiation" value="true"/>
确认 voicemail 模块是否启用
[root@netkiller ~]# grep voicemail /etc/freeswitch/autoload_configs/modules.conf.xml
<load module="mod_voicemail"/>
语音信箱存储目录
[root@netkiller ~]# ls /var/lib/freeswitch/storage/voicemail/default/sip.aigcsst.com/ 1000 1001 1002 1003 1004 1005 1006 1007 1009 1010 1011 1012 1013
拨打4000，根据提示输入user id和密码，就可以收听到留言了。
yum install -y mariadb mariadb-server service mariadb start systemctl enable mariadb mysql_secure_installation
[root@netkiller ~]# cat /etc/freeswitch/autoload_configs/pre_load_modules.conf.xml
<configuration name="pre_load_modules.conf" description="Modules">
<modules>
<!-- Databases -->
<!-- <load module="mod_mariadb"/> -->
<load module="mod_pgsql"/>
</modules>
</configuration>
注释 mod_pgsql 启用 mod_mariadb
<load module="mod_mariadb"/>
<!-- <load module="mod_pgsql"/> -->
[root@netkiller ~]# cat /etc/freeswitch/autoload_configs/db.conf.xml
<configuration name="db.conf" description="LIMIT DB Configuration">
<settings>
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
</settings>
</configuration>
<configuration name="db.conf" description="LIMIT DB Configuration">
<settings>
<param name="core-db-dsn" value="mariadb://Server=192.168.0.11;Port=3307;Database=freeswitch;Uid=root;Pwd=123456;" />
</settings>
</configuration>
修改下面文件中的 core-db-dsn
[root@netkiller ~]# vim /etc/freeswitch/autoload_configs/switch.conf.xml
[root@netkiller ~]# cat /etc/freeswitch/autoload_configs/switch.conf.xml | grep dsn
<!-- <param name="core-db-dsn" value="pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='' options='-c client_min_messages=NOTICE'" /> -->
<!-- <param name="core-db-dsn" value="postgresql://freeswitch:@127.0.0.1/freeswitch?options=-c%20client_min_messages%3DNOTICE" /> -->
<param name="core-db-dsn" value="mariadb://Server=localhost;Database=freeswitch;Uid=freeswitch;Pwd=pass;" />
<!-- <param name="core-db-dsn" value="dsn:username:password" /> -->
数据源格式
mariadb://Server=localhost;Database=freeswitch;Uid=freeswitch;Pwd=pass;
修改下面文件中的 odbc-dsn
[root@netkiller ~]# vim /etc/freeswitch/autoload_configs/switch.conf.xml [root@netkiller ~]# vim /etc/freeswitch/autoload_configs/db.conf.xml [root@netkiller ~]# vim /etc/freeswitch/sip_profiles/internal.xml [root@netkiller ~]# vim /etc/freeswitch/sip_profiles/internal-ipv6.xml
freeswitch提供了测试号码
号码 说明 9664 保持音乐 9196 echo，回音测试 9195 echo，回音测试，延迟5秒 9197 milliwatte extension，铃音生成 9198 TGML 铃音生成示例 5000 示例IVR4000听取语音信箱 33xx 电话会议，48K(其中xx可为00-99，下同) 32xx 电话会议，32K 31xx 电话会议，16K 30xx 电话会议，8K 2000-2002 呼叫组 1000-1019 默认分机号
1000-1019 测试账号位置
[root@netkiller ~]# ls /etc/freeswitch/directory/default/ 1000.xml 1002.xml 1004.xml 1006.xml 1008.xml 1010.xml 1012.xml 1014.xml 1016.xml 1018.xml brian.xml example.com.xml 1001.xml 1003.xml 1005.xml 1007.xml 1009.xml 1011.xml 1013.xml 1015.xml 1017.xml 1019.xml default.xml skinny-example.xml
用户配置文件
[root@netkiller ~]# cat /etc/freeswitch/directory/default/1000.xml
<include>
<user id="1000">
<params>
<param name="password" value="$${default_password}"/>
<param name="vm-password" value="1000"/>
</params>
<variables>
<variable name="toll_allow" value="domestic,international,local"/>
<variable name="accountcode" value="1000"/>
<variable name="user_context" value="default"/>
<variable name="effective_caller_id_name" value="Extension 1000"/>
<variable name="effective_caller_id_number" value="1000"/>
<variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
<variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
<variable name="callgroup" value="techsupport"/>
</variables>
</user>
</include>
账号密码在 /etc/freeswitch/vars.xml 配置文件中
<X-PRE-PROCESS cmd="set" data="default_password=eFyS2wcZo1zKZ3KG"/>
用下面方法查看密码
[root@netkiller ~]# cat /etc/freeswitch/vars.xml | grep default_password
YOU SHOULD CHANGE THIS default_password value if you don't want to be subject to any
<X-PRE-PROCESS cmd="set" data="default_password=eFo1zKZyS2wcZ3KG"/>
进入 fs_cli
[root@netkiller ~]# fs_cli .=======================================================. | _____ ____ ____ _ ___ | | | ___/ ___| / ___| | |_ _| | | | |_ \___ \ | | | | | | | | | _| ___) | | |___| |___ | | | | |_| |____/ \____|_____|___| | | | .=======================================================. | Anthony Minessale II, Ken Rice, | | Michael Jerris, Travis Cross | | FreeSWITCH (http://www.freeswitch.org) | | Paypal Donations Appreciated: paypal@freeswitch.org | | Brought to you by ClueCon http://www.cluecon.com/ | .=======================================================. .=======================================================================================================. | _ _ ____ _ ____ | | / \ _ __ _ __ _ _ __ _| | / ___| |_ _ ___ / ___|___ _ __ | | / _ \ | '_ \| '_ \| | | |/ _` | | | | | | | | |/ _ \ | / _ \| '_ \ | | / ___ \| | | | | | | |_| | (_| | | | |___| | |_| | __/ |__| (_) | | | | | | /_/ \_\_| |_|_| |_|\__,_|\__,_|_| \____|_|\__,_|\___|\____\___/|_| |_| | | | | ____ _____ ____ ____ __ | | | _ \_ _/ ___| / ___|___ _ __ / _| ___ _ __ ___ _ __ ___ ___ | | | |_) || || | | | / _ \| '_ \| |_ / _ \ '__/ _ \ '_ \ / __/ _ \ | | | _ < | || |___ | |__| (_) | | | | _| __/ | | __/ | | | (_| __/ | | |_| \_\|_| \____| \____\___/|_| |_|_| \___|_| \___|_| |_|\___\___| | | | | ____ _ ____ | | / ___| |_ _ ___ / ___|___ _ __ ___ ___ _ __ ___ | | | | | | | | |/ _ \ | / _ \| '_ \ / __/ _ \| '_ ` _ \ | | | |___| | |_| | __/ |__| (_) | | | | _ | (_| (_) | | | | | | | | \____|_|\__,_|\___|\____\___/|_| |_| (_) \___\___/|_| |_| |_| | | | .=======================================================================================================. Type /help <enter> to see a list of commands
输入密码，并进入客户端
fs_cli -H 127.0.0.1 -P 8021 -p password
打开sip详细日志 sofia profile internal siptrace on 关闭sip详细日志 sofia profile internal siptrace off
[root@netkiller ~]# fs_cli -x "sofia status"
Name Type Data State
=================================================================================================
external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0)
sip.netkiller.cn alias internal ALIASED
external profile sip:mod_sofia@192.168.0.230:5080 RUNNING (0)
external::example.com gateway sip:joeuser@example.com NOREG
external::hamsoverip gateway sip:300177@pbx-ap.hamsoverip.com:5160 REGED
internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0)
internal profile sip:mod_sofia@192.168.0.230:5060 RUNNING (0)
=================================================================================================
4 profiles 1 alias
freeswitch@-ERR switchname Command not found!> sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT true DBName sofia_reg_internal Pres Hosts sip.aigcsst.com,192.168.0.230 Dialplan XML Context public Challenge Realm auto_from RTP-IP 192.168.0.230 Ext-RTP-IP 121.37.215.251 SIP-IP 192.168.0.230 Ext-SIP-IP 121.37.215.251 URL sip:mod_sofia@192.168.0.230:5060 BIND-URL sip:mod_sofia@192.168.0.230:5060;transport=udp,tcp WS-BIND-URL sip:mod_sofia@192.168.0.230:5066;transport=ws WSS-BIND-URL sips:mod_sofia@192.168.0.230:7443;transport=wss HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN OPUS,G722,G729,PCMU,PCMA,H264,VP8 CODECS OUT OPUS,G722,G729,PCMU,PCMA,H264,VP8 TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 MAX-RECV-RPS 1000 NOMEDIA false LATE-NEG true PROXY-MEDIA true AGGRESSIVENAT false CALLS-IN 195 FAILED-CALLS-IN 99 CALLS-OUT 22 FAILED-CALLS-OUT 14 REGISTRATIONS 8
freeswitch@-ERR switchname Command not found!> sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT true DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 192.168.0.230 Ext-RTP-IP 121.37.215.251 SIP-IP 192.168.0.230 Ext-SIP-IP 121.37.215.251 URL sip:mod_sofia@192.168.0.230:5080 BIND-URL sip:mod_sofia@192.168.0.230:5080;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN OPUS,G722,G729,PCMU,PCMA,H264,VP8 CODECS OUT OPUS,G722,G729,PCMU,PCMA,H264,VP8 TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 MAX-RECV-RPS 1000 NOMEDIA false LATE-NEG true PROXY-MEDIA false AGGRESSIVENAT false CALLS-IN 3 FAILED-CALLS-IN 3 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 0
[root@netkiller ~]# fs_cli -x "sofia status profile internal reg" Registrations: ================================================================================================= Call-ID: 1_4246134458@192.168.123.55 User: 1008@sip.netkiller.cn Contact: "1008" <sip:1008@192.168.123.55:5060;fs_nat=yes;fs_path=sip%3A1008%40223.74.131.87%3A20508> Agent: Yealink SIP-T21P_E2 52.84.0.125 Status: Registered(UDP-NAT)(unknown) EXP(2025-04-06 19:32:38) EXPSECS(2540) Ping-Status: Reachable Ping-Time: 0.00 Host: netkiller IP: 223.74.131.87 Port: 20508 Auth-User: 1008 Auth-Realm: sip.netkiller.cn MWI-Account: 1008@sip.netkiller.cn Call-ID: 212FF149-109A-40AF-A2CF-23003FC53819 User: 1002@sip.netkiller.cn Contact: "BG7NYT" <sip:1002@172.16.0.11:5060;fs_nat=yes;fs_path=sip%3A1002%40112.97.167.132%3A13917> Agent: Jami Daemon 16.0.0-fc3402940f (macOS) Status: Registered(UDP-NAT)(unknown) EXP(2025-04-06 19:39:37) EXPSECS(2959) Ping-Status: Reachable Ping-Time: 0.00 Host: netkiller IP: 112.97.167.132 Port: 13917 Auth-User: 1002 Auth-Realm: sip.netkiller.cn MWI-Account: 1002@sip.netkiller.cn Call-ID: 139311fc-5f0ee947@172.16.0.10 User: 1000@sip.netkiller.cn Contact: "BG7NYT" <sip:1000@172.16.0.10:5060;fs_nat=yes;fs_path=sip%3A1000%40112.97.167.132%3A12689> Agent: Linksys/PAP2T-5.1.6(LS) Status: Registered(UDP-NAT)(unknown) EXP(2025-04-06 19:39:51) EXPSECS(2973) Ping-Status: Reachable Ping-Time: 0.00 Host: netkiller IP: 112.97.167.132 Port: 12689 Auth-User: 1000 Auth-Realm: sip.netkiller.cn MWI-Account: 1000@sip.netkiller.cn Call-ID: 5A7530FB452FDFDC5BDCD7E3A73DF6EEBC4D93D0 User: 1006@sip.netkiller.cn Contact: "" <sip:1006@10.65.4.4:23015;rinstance=18F4361D;fs_nat=yes;fs_path=sip%3A1006%40167.99.119.244%3A23015%3Brinstance%3D18F4361D> Agent: Acrobits SIPIS Status: Registered(UDP-NAT)(unknown) EXP(2025-04-06 18:57:03) EXPSECS(405) Ping-Status: Reachable Ping-Time: 0.00 Host: netkiller IP: 167.99.119.244 Port: 23015 Auth-User: 1006 Auth-Realm: sip.netkiller.cn MWI-Account: 1006@sip.netkiller.cn Total items returned: 4 =================================================================================================
重新扫描加载 sip profile
sofia profile internal rescan sofia profile external rescan
重启 sip profile
sofia profile internal restart sofia profile external restart
# 显示当前呼叫 show calls # 显示呼叫数量 show calls count # 挂断某个呼叫 uuid_kill 59b857d2-d7b8-4c7e-6666-e19be0f16643 # 挂断所有呼叫 hupall # 拨打某个用户并启用echo回音 originate user/1000 &echo
回音测试
originate user/1000 &echo
停泊
originate user/1000 &park # 停泊
保持
originate user/1000 &hold # 保持
播放放音
originate user/1000 &playback(/root/welclome.wav) # 播放音乐
呼叫并录音
originate user/1000 &record(/tmp/vocie_of_alice.wav) # 呼叫并录音
同振与顺振
#经过特定的SIP服务器发起外呼，下面的命令会将INVITE先发送到192.168.2.4:5060上
bgapi originate sofia/external/8005@001.com;fs_path=sip:192.168.2.4:5060 &echo
经过特定SIP服务器
#经过特定的SIP服务器发起外呼，下面的命令会将INVITE先发送到192.168.2.4:5060上
bgapi originate sofia/external/8005@001.com;fs_path=sip:192.168.2.4:5060 &echo
忽略early media
originate {ignore_early_media=true}user/1000 &echo
播放假的early media
originate {transfer_ringback=local_stream://moh}user/1000 &echo
立即播放early media
originate {instant_ringback=true}{transfer_ringback=local_stream://moh}user/1000 &echo
设置外显号码
originate {origination_callee_id_name=7777}user/1000
[ ] freeswitch-sounds-zh-cn-sinmei-8000-1.0.51.tar.gz 2014-10-09 20:21 296K [ ] freeswitch-sounds-zh-cn-sinmei-8000-1.0.51.tar.gz.md5 2014-10-09 20:21 92 [ ] freeswitch-sounds-zh-cn-sinmei-8000-1.0.51.tar.gz.sha1 2014-10-09 20:21 101 [ ] freeswitch-sounds-zh-cn-sinmei-8000-1.0.51.tar.gz.sha256 2014-10-09 20:21 127 [ ] freeswitch-sounds-zh-cn-sinmei-16000-1.0.51.tar.gz 2014-10-09 20:21 583K [ ] freeswitch-sounds-zh-cn-sinmei-16000-1.0.51.tar.gz.md5 2014-10-09 20:21 93 [ ] freeswitch-sounds-zh-cn-sinmei-16000-1.0.51.tar.gz.sha1 2014-10-09 20:21 102 [ ] freeswitch-sounds-zh-cn-sinmei-16000-1.0.51.tar.gz.sha256 2014-10-09 20:21 128 [ ] freeswitch-sounds-zh-cn-sinmei-32000-1.0.51.tar.gz 2014-10-09 20:21 1.1M [ ] freeswitch-sounds-zh-cn-sinmei-32000-1.0.51.tar.gz.md5 2014-10-09 20:21 93 [ ] freeswitch-sounds-zh-cn-sinmei-32000-1.0.51.tar.gz.sha1 2014-10-09 20:21 102 [ ] freeswitch-sounds-zh-cn-sinmei-32000-1.0.51.tar.gz.sha256 2014-10-09 20:21 128 [ ] freeswitch-sounds-zh-cn-sinmei-48000-1.0.51.tar.gz 2014-10-09 20:21 1.6M [ ] freeswitch-sounds-zh-cn-sinmei-48000-1.0.51.tar.gz.md5 2014-10-09 20:21 93 [ ] freeswitch-sounds-zh-cn-sinmei-48000-1.0.51.tar.gz.sha1 2014-10-09 20:21 102 [ ] freeswitch-sounds-zh-cn-sinmei-48000-1.0.51.tar.gz.sha256 2014-10-09 20:21 128
编译 mod_say_zh 模块
cd /usr/local/src/freeswitch/src/mod/say/mod_say_zh make && make install
autoload_configs/modules.conf.xml
<!-- Say -->
<load module="mod_say_en"/>
<load module="mod_say_zh"/>
cd /usr/local/freeswitch/conf/lang/ cp -fr en zh cd zh mv en.xml zh.xml
zh.xml
<language name="zh" say-module="zh" sound-prefix="$${sounds_dir}/zh/cn/link" tts-engine="mod_tts_commandline" tts-voice="link">
vars.xml
<X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/zh/cn/link"/>
<X-PRE-PROCESS cmd="set" data="default_language=zh"/>
<X-PRE-PROCESS cmd="set" data="default_dialect=cn"/>
<X-PRE-PROCESS cmd="set" data="default_voice=link"/>
freeswitch.xml
<X-PRE-PROCESS cmd="include" data="lang/zh/*.xml"/>
fs_cli 手动加载模块
load mod_say_zh reloadxml
配置 Dialplan 拨号计划 dialplan/default.xml
<!--say测试-->
<extension name="socket_767_example">
<condition field="destination_number" expression="^767\d+$">
<action application="answer"/>
<action application="say" data="zh name_spelled intered 3456789"></action>
<action application="say" data="en NUMBER intered 3456789"></action>
<action application="say" data="zh TELEPHONE_NUMBER intered 13781655437"></action>
<action application="playback" data="voicemail/vm-goodbye.wav"></action>
</condition>
</extension>
单个用户配置 1001.xml
<variable name="language" value="zh"/> <variable name="default_language" value="zh"/>
dialplan中配置中文
<extension name="ivr_demo">
<condition field="destination_number" expression="^5000$">
<action application="set" data="language=zh"/>
<action application="answer"/>
<action application="sleep" data="2000"/>
<action application="ivr" data="demo_ivr"/>
</condition>
</extension>
# freeswitch 用户管理
## 安装依赖
pip install -r requirements.txt -i https://pypi.tuna.tsinghua.edu.cn/simple
## 帮助信息
```shell
PS D:\GitHub\freeswitch> python.exe .\freeswitch.py -h
usage: freeswitch.py [-h] [-a <number> <callsign> <callgroup> [<number> <callsign> <callgroup> ...]] [-r 1000] [-l] [-s 1000] [-d]
FreeSWITCH 用户管理工具
options:
-h, --help show this help message and exit
-a <number> <callsign> <callgroup> [<number> <callsign> <callgroup> ...], --add <number> <callsign> <callgroup> [<number> <callsign> <callgroup> ...]
添加用户
-r 1000, --remove 1000
删除用户
-l, --list 列出用户
-s 1000, --show 1000 查看用户
-d, --debug 调试模式
Author: netkiller - https://www.netkiller.cn/linux/
```
## 添加用户
```shell
PS D:\GitHub\freeswitch> python.exe .\freeswitch.py -a 1000 BG7NYT admin
```
## 查看用于
```shell
PS D:\GitHub\freeswitch> python.exe .\freeswitch.py -s 1000
<include>
<user id="1000">
<params>
<param name="password" value="B5AbNCYj"/>
<param name="vm-password" value="1000"/>
</params>
<variables>
<variable name="toll_allow" value="domestic,international,local"/>
<variable name="accountcode" value="1000"/>
<variable name="user_context" value="default"/>
<variable name="effective_caller_id_name" value="BG7NYT"/>
<variable name="effective_caller_id_number" value="1000"/>
<variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
<variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
<variable name="callgroup" value="admin"/>
</variables>
</user>
</include>
```
## 列出所有用户
```shell
PS D:\GitHub\freeswitch> python.exe .\freeswitch.py -l
+----------+--------+----------+----------+--------+
| 电话号码 | 呼号 | 密码 | 语音信箱 | 呼叫组 |
+==========+========+==========+==========+========+
| 1000 | BG7NYT | HbM3imgb | 2031 | admin |
+----------+--------+----------+----------+--------+
| 1003 | BG7NYT | 1u3Fc6t4 | 5927 | 33333 |
+----------+--------+----------+----------+--------+
```
## 删除用户
```shell
PS D:\GitHub\freeswitch> python.exe .\freeswitch.py -r 1000
```