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Asterisk Version

VerTypeReleaseEOL
Asterisk 20LTS2022-10-192027-10-19
Asterisk 19Standard2021-11-022023-11-02
Asterisk 18LTS2020-10-202024-10-20
Asterisk 17Standard2019-10-282021-10-28
Asterisk 16LTS2018-10-092022-10-09
Asterisk 15Standard2017-10-032019-10-03
Asterisk 14Standard2016-09-262018-09-26
Asterisk 13LTS2014-10-242020-10-24
Asterisk 11LTS2012-10-252017-10-25
Asterisk 1.8LTS2010-10-212015-10-21
[Asterisk 1.4]LTS2006-12-232012-04-21
# 获取给的版本源码
curl -O https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18-current.tar.gz

Asterisk 20​

Asterisk 19​

  • debug log categories
    • dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, and stun_packet
    • "simple" record formatting mode
  • app_voicemail - send greeting and instructions
  • astcachedir - /var/cache/asterisk
  • app_confbridge - force the estimated bitrate on an SFU
  • Originate 可设置变量
  • ConfKick
  • app_dial - A
  • app_morsecode
  • CHANGES

Asterisk 18​

Asterisk 17​

  • 废弃 chan_sip
  • ARI 支持应用事件过滤
  • AttendedTransfer - queue up attended transfer to the given extensio
  • BlindTransfer - redirect all channels currently bridged to the caller channel to the specified destination
  • ConfBridge
    • remb_behavior 选项新增 average_all, highest_all, lowest_all
      • bridge 级别而不是每个来源级别控制
  • Dial
    • RINGTIME 和 RINGTIME_MS 变量 - 秒和毫秒 - 创建通话通道和收到第一个 RINGING 信号的间隔
    • PROGRESSTIME 和 PROGRESSTIME_MS - 同上 - 处理 PROGRESS 信号 - 最低值应该为 PDD (Post Dial Delay)
    • DIALEDTIME_MS 和 ANSWEREDTIME_MS - DIALEDTIME 和 ANSWEREDTIME 的毫秒版本
  • ReadExten - 添加 p 选项当用户按 # 时停止
  • RTP/ICE
    • 可以使用 ice_host_candidate 本地地址
  • pbx_dundi - DUNDi 支持 IPv4/IPv6 双绑定
  • 新增 res_mwi_devstate 模块
    • 订阅语音信箱状态事件
  • Prometheus exporter

Asterisk 16​

新增​

  • WebRTC

    • 提示视频质量
    • 支持 REMB
      • 聚合预估每个客户端的可用带宽反馈给发送方 - 发送方可以此来调整包大小
    • 支持 NACK
      • 允许客户端请求重新发送来实现更好的乱序包处理

  • PJSIP

    • 性能提升 - 允许更多的 AOR 且不影响启动时间

  • ConfBridge

    • 文本消息会被中继转发 - 如果开启了 enable_events 还会发送 JSON 事件

  • app_originate 支持 a 参数, 异步拨号, 不等待响应

  • app_macro 模块废弃, 使用 app_stack (Gosub)

  • res_config_sqlite 模块废弃, 使用 res_config_sqlite3

  • res_monitor 模块废弃, 使用 app_mixmonitor

  • cdr_syslog 模块废弃, 并且默认不会构建

  • app_fax 模块废弃, 使用 res_fax

  • 16.7

  • 16.6.2

Asterisk 15​

Asterisk 14​

  • Standard | 2016-09-26 - 2018-09-26

  • 核心 DNS 支持 - libunbound 支持 PJSIP NAPTR SRV

  • 发布 extension 状态到 SIP 订阅服务器 - 例如 Kamailio

    • 能基于设备状态自动在拨号计划里生成 hint

  • 所有应用支持播放 HTTP 媒体

  • ARI 支持批量媒体资源管理

  • ARI 创建的通道可以转交给 Stasis 应用进行外部控制 - 可以在接通前进行额外操作，例如启用特定的媒体场景

  • 14.6.2 - 2017-09-19

  • 14.6.1 - 2017-08-31

    • 5902 res_pjsip: dtmf_mode 添加 auto_info
      • 默认的 auto 会将 dtmf 模式回退到 inband, 该模式是回退为 INFO
    • [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash

  • 14.6.0

    • [ASTERISK-22432] - Async AGI crashes Asterisk when issuing "set variable" command without args
    • [ASTERISK-26978] - rtp: Crash in ast_rtp_codecs_payload_code()
    • [ASTERISK-27016] - Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
    • [ASTERISK-27026] - res_ari: Crash when no ari.conf configuration file exists
    • [ASTERISK-27050] - Crash on Transcoded Audio in PERIODIC_HOOK Function
    • [ASTERISK-27108] - Crash using 'data get' CLI command
    • [ASTERISK-25370] - res_corosync segfaults at startup with corosync version > 2.x
    • [ASTERISK-27046] - res_pjsip_transport_websocket: segfault in get_write_timeout
    • [ASTERISK-27057] - Seg Fault in ast_sorcery_object_get_id at sorcery.c

  • 14.5.0

    • [ASTERISK-21855] - Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
    • [ASTERISK-26692] - res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
    • [ASTERISK-26835] - res_rtp_asterisk: Crash when freeing RTCP address string
    • [ASTERISK-26853] - res_rtp_asterisk: Crash in pjnath when receiving packet
    • [ASTERISK-26926] - func_speex: Crash caused by frame with no datalen
    • [ASTERISK-26927] - pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
    • [ASTERISK-26953] - Asterisk crash if hep.conf have some missing parameters
    • [ASTERISK-26983] - Crash in Manager Reload when TLS Config Changes
    • [ASTERISK-25506] - [patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
    • [ASTERISK-26606] - tcptls: Incorrect OpenSSL function call leads to misleading error report

Asterisk 13​

  • LTS | 2014-10-24 - 2020-10-24
  • REST
  • WebSocket
  • Stasis

Asterisk 12​

  • 新增 ARI 接口
  • 新增 Stasis 消息总线
  • 新增 chan_pjsip

Asterisk 11​

sip.conf

[general]
icesupport=yes

rtp.conf

stunaddr=setyourphaserson.stun.org

turnaddr=4everyseason.turn.org
turnusername=relayme
turnpassword=please

Asterisk 10​

  • 内部数据库从 Berkeley DB 切换为 SQLite 3

Asterisk 1.8​

很多旧的 VoIP 网关使用该版本 - 且可能永远不会升级

  • LTS | 2010-10-21 - 2015-10-21
  • 第二个 LTS 版本 - 大量使用
  • 支持 AMI
  • 支持 AGI
  • SRTP
  • SIP IPv6
  • Connected Party Identification - COLP and CONP
  • 日历集成 - CalDAV, iCal, Exchange, EWS calendars
  • Channel Event Logging - CEL
  • Distributed Device State - Message Waiting Indicator - Jabber/XMPP PubSub
  • 支持 Call Completion Supplementary Services - CCSS
    • Call Completion on Busy Subscriber - CCBS
    • Call Completion on No Response - CCNR
  • Advice of Charge - AOC-S, AOC-D, AOC-E
  • Multicast RTP
  • ISDN Q.SIG Call Rerouting and Call Deflection
  • Google Talk, Google Voice integration
  • Audio Pitch Shifting
  • New in 1.8