Asterisk Modules

# 模块类型
ls /usr/lib/asterisk/modules | egrep -o '^[a-z]+' | sort | uniq -c | sort -nr
ls /usr/lib/asterisk/modules | xargs -n 1 basename | sed 's/.so$//' | sort | grep func_

# 模块支持状态
grep -rP '^\t<support_level>' $(find . -name '*.c') | sed -re 's#</?support_level>##g' | sort

asterisk -x 'module show'

内置模块

不可以动态 load、unload - 部分可以配置不启用

module	conf	desc
acl	acl.conf	统一 ACL 控制
ccss	ccss.conf	Call Completion Supplementary Services
cdr	cdr.conf	Call Detail Record
cel	cel.conf	Call Event Log
dnsmgr	dnsmgr.conf	内置 DNS 管理器
dsp	dsp.conf
enum	enum.conf	通过 DNS 解析电话号码
extconfig	extconfig.conf	静态和实时外部配置引擎
features	features.conf
http	http.conf	内置 HTTP 服务
indications	indications.conf
logger	logger.conf
manager	manager.conf	AMI
plc		Packet Loss Concealment - 音频丢包补偿
sounds		声音文件索引
udptl	udptl.conf	UDP Transport Layer - 端口配置

dsp - 信号处理
CCSS - Call Completion Supplementary Services - 通话完成补偿服务
- 由 CCBS 和 CCNR 组成
  - CCBS - Call Completion on Busy Subscriber
  - CCNR - Call Completion on No Response
- 例如 A 呼叫 B，但 B 在和 C 通话，开启 CCSS/CCBS 后，可以在 B 和 C 通话完成后通知 A

动态模块

可动态加载

可能不同模块提供相同功能 - 模块互斥

type	desc
app	Dialplan 应用
bridge	三方通话、会议
cdr	CDR+后端配置
cel	CEL+后端配置
chan	通道技术
codec	语音编码
format	语音文件格式
func	Dialplan 函数
pbx	路由
res	资源

chan

chan	conf	desc
chan_audiosocket		简单的 Audio over TCP 协议 - 用于实现外部音频处理
chan_alsa	alsa.conf	Linux 接口
chan_bridge_media
chan_console	console.conf	Linux 接口
chan_dahdi	chan_dahdi.conf	DAHDi 板卡接口
chan_dongle	dongle.conf	GSM 接口
chan_iax2	iax.conf	Inter Asterisk eXchange (Ver 2)
chan_mgcp	mgcp.conf	Media Gateway Control Protocol
chan_oss	oss.conf	Linux 接口
chan_pjsip	pjsip.conf	基于 pjproject 的 sip 实现，推荐使用
chan_rtp	rtp.conf	RTP Media Channel
chan_sip	sip.conf	最早的 sip 实现，⚠️ 不再维护
chan_skinny	skinny.conf	Skinny Client Control Protocol
chan_unistim	unistim.conf	Unified Networks IP Stimulus
chan_mobile	chan_mobile.conf	蓝牙

modules	dependencies
chan_rtp	res_rtp_multicast

Linux 音频接口
- 默认 oss
- alsa - Advanced Linux Sound Architecture
  - 作为替代 OSS 存在
- console - portaudio - audio i/o devices as telephony devices
  - portaudio 实现依赖 alsa
  - portaudio 支持 TCP
  - 有好的用户交互工具
  - 支持复杂的 mix 逻辑
- oss - Open Sound System
  - 最早
skinny - SCCP - Skinny Client Control Protocol
- chan-sccp/chan-sccp
  - 替代 chan_skinny
  - 支持更多功能
mgcp - Media Gateway Control Protocol
unistim - UNIStim - Unified Networks IP Stimulus
- 废弃协议

# PJSIP
# ==========
# lookup AOR, dial first Contact
exten => _6XXX,1,Dial(PJSIP/${EXTEN})
# dial all conacts - 会拼接所有 contacts
exten => _6XXX,1,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})})
# dial SIP - trunk 线路
# 不使用 AOR 和 Contact
exten => _9NXXNXXXXXX,1,Dial(PJSIP/mytrunk/sip:${EXTEN:1}@203.0.113.1:5060)
# dial 预先配置 trunk 上的 AOR 信息
exten => _9NXXNXXXXXX,1,Dial(PJSIP/${EXTEN:1}@mytrunk)

# Local
# ==========
exten => 201,1,Verbose(2,Call desk phone and cellphone but with delay)
exten => 201,n,Dial(Local/deskphone-201@extensions&Local/cellphone-201@extensions,30)

# AudioSocket
# ==========
# Dialplan
exten = 100,1,Verbose("Call to AudioSocket via Dialplan Application")
 same = n,Answer()
 same = n,AudioSocket(40325ec2-5efd-4bd3-805f-53576e581d13,server.example.com:9092)
 same = n,Hangup()
# Chan
exten = 101,1,Verbose("Call to AudioSocket via Channel interface")
 same = n,Answer()
 same = n,Dial(AudioSocket/server.example.com:9092/40325ec2-5efd-4bd3-805f-53576e581d13)
 same = n,Hangup()

pbx

module	conf	desc
pbx_ael	extensions.ael
pbx_config	extensions.conf	最常用
pbx_dundi
pbx_loopback
pbx_lua	extensions.lua
pbx_realtime	db/extension
pbx_spool		将 callfile 放到目录下自动发起拨号

pbx_realtime
- 相当于把 extensions.conf 一行一行存到了数据库, 实际上不太好管理
pbx_lua
- 使用 Lua 来写 dialplain

codec

codec	desc
codec_a_mu
codec_adpcm
codec_alaw
codec_dahdi	/dev/dahdi/transcode - DAHDi 硬件转码
codec_g722
codec_g726
codec_gsm
codec_ilbc
codec_lpc10
codec_resample	从新采样
codec_ulaw

format

format	desc
format_g719
format_g723
format_g726
format_g729
format_gsm
format_h263
format_h264
format_ilbc
format_pcm
format_siren14
format_siren7
format_sln
format_vox
format_wav
format_wav_gsm

cdr

module	conf	desc
cdr_csv
cdr_custom
cdr_manager		AMI 接收 CDR
cdr_mysql
cdr_pgsql
cdr_sqlite3_custom
cdr_syslog
cdr_tds		MS SQL Server

cel

module	conf	desc
cel_custom
cel_manager		AMI 接收 CEL
cel_pgsql
cel_sqlite3_custom
cel_tds		MS SQL Server

bridge

module	desc
bridge_builtin_features
bridge_builtin_interval_features
bridge_holding	store channels in bridge for holding, parking, queues
bridge_native_rtp	Native RTP bridging
bridge_simple
bridge_softmix	多方，软 mix

func

func	desc
func_aes
func_base64
func_blacklist
func_callcompletion
func_callerid
func_cdr
func_channel
func_config
func_curl
func_cut
func_db
func_devstate
func_dialgroup
func_dialplan
func_enum
func_env
func_extstate
func_frame_trace
func_global
func_groupcount
func_hangupcause
func_holdintercept
func_iconv
func_jitterbuffer
func_lock
func_logic
func_math
func_md5
func_module
func_periodic_hook
func_pitchshift
func_pjsip_aor
func_pjsip_contact
func_pjsip_endpoint
func_presencestate
func_rand
func_realtime
func_sha1
func_shell
func_sorcery
func_sprintf
func_srv
func_strings
func_sysinfo
func_talkdetect
func_timeout
func_uri
func_version
func_vmcount
func_volume

app

app	desc
app_adsiprog
app_agent_pool
app_alarmreceiver
app_amd
app_attended_transfer
app_audiosocket
app_authenticate
app_blind_transfer
app_bridgeaddchan
app_bridgewait
app_cdr
app_celgenuserevent
app_chanisavail
app_channelredirect
app_chanspy
app_confbridge	confbridge.conf
app_controlplayback
app_dahdiras
app_db
app_dial
app_dictate
app_directed_pickup
app_directory
app_disa	Direct Inward System Access
app_dumpchan
app_echo	测试应用, 感受延时, 确认是否有声音
app_exec
app_externalivr
app_festival
app_flash
app_followme
app_forkcdr
app_getcpeid
app_ices
app_image
app_meetme	MeetMe conference bridge
app_milliwatt
app_minivm
app_mixmonitor	Mixed Audio Monitoring
app_morsecode
app_mp3
app_nbscat
app_originate
app_page
app_playback
app_playtones
app_privacy
app_queue
app_read
app_readexten
app_record
app_sayunixtime
app_senddtmf
app_sendtext
app_sms	SMS/PSTN
app_softhangup
app_speech_utils
app_stack	Dialplan subroutines - Gosub, Return
app_stasis
app_stream_echo
app_system
app_talkdetect
app_test	Interface Test Application
app_transfer
app_url
app_userevent
app_verbose	VERBOSE
app_voicemail
app_waitforring
app_waitforsilence
app_waituntil
app_while
app_zapateller

res

res module	desc
res_adsi
res_ael_share
res_agi
res_audiosocket
res_calendar
res_clialiases
res_clioriginate
res_config_curl
res_config_pgsql
res_config_sqlite3
res_convert
res_crypto
res_curl
res_format_attr_celt
res_format_attr_g729
res_format_attr_h263
res_format_attr_h264
res_format_attr_ilbc
res_format_attr_opus
res_format_attr_silk
res_format_attr_siren14
res_format_attr_siren7
res_format_attr_vp8
res_hep
res_hep_pjsip
res_hep_rtcp
res_http_media_cache
res_http_websocket
res_limit
res_manager_devicestate
res_manager_presencestate
res_monitor	Monitor 功能, 现在更多推荐使用 app_mixmonitor
res_musiconhold
res_mutestream
res_mwi_devstate
res_parking
res_phoneprov
res_prometheus	prometheus 监控
res_realtime
res_resolver_unbound
res_rtp_asterisk	RTP，RTCP with Symmetric RTP support for NAT
res_rtp_multicast
res_security_log
res_smdi
res_snmp	SNMP 集成
res_speech
res_srtp
res_statsd	stattsd 监控
res_stir_shaken	STIR/SHAKEN
res_stun_monitor
res_timing_dahdi
res_timing_pthread
res_timing_timerfd

module	dependencies
res_agi	res_speech
res_rtp_asterisk	res_pjproject
res_snmp	net-snmp-dev

STIR/SHAKEN
- STIR - Secure Telephone Identity Revisited
- SHAKEN - Signature-based Handling of Asserted Information Using toKENs
- 避免 PTNS CallID 欺骗

res_stasis

stasis 应用模块
基于 WebSocket 实时控制

res_stasis	desc
res_stasis
res_stasis_answer
res_stasis_device_state
res_stasis_playback
res_stasis_recording
res_stasis_snoop

res_sorcery

asterisk 统一的配置访问接口 - CRUD

res_sorcery	desc
res_sorcery_astdb	从 astdb 获取配置
res_sorcery_config	操作配置文件
res_sorcery_memory	内存配置
res_sorcery_memory_cache	缓存远程配置
res_sorcery_realtime	从实时后端获取配置

res_ari

Asterisk RESTful Interface
REST 管理 Asterisk 资源
不同模块提供不同资源接口

ari module	desc
res_ari
res_ari_applications
res_ari_asterisk
res_ari_bridges
res_ari_channels
res_ari_device_states
res_ari_endpoints
res_ari_events
res_ari_model
res_ari_playbacks
res_ari_recordings
res_ari_sounds

res_pjsip

pjsip module	desc
res_pjproject	PJPROJECT Log & Utility
res_pjsip	基础 SIP 资源
res_pjsip_acl	PJSIP ACL
res_pjsip_authenticator_digest	Digest 认证
res_pjsip_caller_id	支持 CallerID
res_pjsip_config_wizard	pjsip_wizard.conf
res_pjsip_dialog_info_body_generator	Extension State Dialog Info+XML
res_pjsip_diversion	Diversion Header
res_pjsip_dlg_options	SIP OPTIONS in dialog
res_pjsip_dtmf_info	支持 DTMF INFO
res_pjsip_empty_info	支持 Empty INFO
res_pjsip_endpoint_identifier_anonymous	匿名终端标识
res_pjsip_endpoint_identifier_ip	IP 终端标识
res_pjsip_endpoint_identifier_user	用户名终端标识
res_pjsip_exten_state	Extension State Notifications
res_pjsip_header_funcs	Header Functions
res_pjsip_history	历史
res_pjsip_logger	包日志
res_pjsip_messaging	支持消息
res_pjsip_mwi	MWI 资源
res_pjsip_mwi_body_generator
res_pjsip_nat	NAT 支持
res_pjsip_notify	支持 CLI/AMI PJSIP NOTIFY
res_pjsip_one_touch_record_info	INFO One Touch Recording
res_pjsip_outbound_authenticator_digest	outbound digest 认证
res_pjsip_outbound_publish
res_pjsip_outbound_registration	outbound 注册
res_pjsip_path	Path 头
res_pjsip_phoneprov_provider
res_pjsip_pidf_body_generator	Extension State PIDF Provider
res_pjsip_pidf_digium_body_supplement	PIDF Sangoma presence supplement
res_pjsip_pidf_eyebeam_body_supplement	PJSIP PIDF Eyebeam supplement
res_pjsip_publish_asterisk	Asterisk Event PUBLISH
res_pjsip_pubsub	event
res_pjsip_refer	Blind and Attended Transfer
res_pjsip_registrar	Registrar
res_pjsip_rfc3326	Reason 头
res_pjsip_sdp_rtp	SDP RTP/AVP stream handler
res_pjsip_send_to_voicemail	REFER Send to Voicemail
res_pjsip_session
res_pjsip_sips_contact	UAC SIPS Contact
res_pjsip_stir_shaken
res_pjsip_t38	PJSIP T.38 UDPTL
res_pjsip_transport_websocket	WebSocket
res_pjsip_xpidf_body_generator	Extension State PIDF Provider

PIDF - Presence Information Data Format
T.38 - FoIP Fax over IP - UDP 协议
rfc3326 - SIP Reason 头
Presence Subscriptions
- res_pjsip_pubsub
- res_pjsip_exten_state
- res_pjsip_pidf_body_generator
- res_pjsip_xpidf_body_generator
- res_pjsip_dialog_info_body_generator
- 特殊驱动支持
  - res_pjsip_pidf_digium_body_supplement
  - res_pjsip_pidf_eyebeam_body_supplement

sip common

load => res_adsi.so
load => res_timing_pthread.so
;load => res_timing_dahdi.so
load => res_agi.so
load => res_crypto.so
load => res_pktccops.so
load => res_smdi.so
load => res_stun_monitor.so
load => res_rtp_asterisk.so
load => res_rtp_msp.so
load => res_curl.so
load => res_clioriginate.so
load => pbx_config.so
load => bridge_multiplexed.so
load => app_dial.so
load => app_exec.so
load => app_system.so
load => app_macro.so
load => app_cdr.so
load => app_chanisavail.so
load => app_grppolicy.so
load => app_mixmonitor.so
load => app_sayunixtime.so
load => app_originate.so
load => app_playback.so
load => app_disa.so
load => app_authenticate.so
load => format_wav.so
load => format_gsm.so
load => func_math.so
load => func_cdr.so
load => func_strings.so
load => func_channel.so
load => func_callerid.so
load => func_timeout.so
load => func_shell.so
load => func_rand.so
load => func_realtime.so
load => func_dialplan.so
load => func_curl.so
load => func_uri.so
load => func_blacklist.so
load => func_db.so
load => func_cut.so
load => cdr_manager.so
load => cdr_syslog.so
load => codec_alaw.so
load => codec_ulaw.so
load => codec_adpcm.so
load => codec_a_mu.so
load => codec_g722.so
load => codec_g723.so
load => codec_g726.so
load => codec_g729.so
load => codec_gsm.so
load => codec_lpc10.so
;load => codec_c1k.so
load => chan_extra.so
load => res_srtp.so
load => chan_sip.so
load => chan_iax2.so
load => res_musiconhold.so
load => func_env.so

res_statsd

statsd.conf

[general]
# 是否启用 statsd 指标
enabled = yes
# statsd 服务端地址
# server[:port] - 默认端口 8125
server = 127.0.0.1
# 指标前缀 - 如果有多个实例则建议添加
prefix = ast-1
# 添加换行
# 方便使用 nc -lu 8125 测试时
;add_newline = no

ast-1.PJSIP.contacts.1876;@47519c5004a0a52468bd2e1570095db4.rtt:19|ms
ast-1.PJSIP.contacts.states.Created:+1|g
ast-1.PJSIP.contacts.states.Created:-1|g
ast-1.PJSIP.contacts.states.Reachable:+1|g

statsd_exporter 映射规则

mappings:
  - match: '([^.]+)[.]PJSIP[.]contacts[.]([^;]+)(;[^.]*)?[.]rtt'
    match_type: regex
    name: 'pjsip_contacts_rtt'
    labels:
      asterisk: '$1'
      conact: '$2'

res_prometheus

res/res_prometheus.c

res_snmp

# 编译需要额外依赖
# ================
apk install net-snmp-dev

net-snmp-config --agent-libs

# ./configure --with-netsnmp

配置

/etc/asterisk/res_snmp.conf

[general]
; 运行为子系统 - 需要 snmpd 启动 agentx
; 非子系统需要 root - 绑定 161 端口
subagent = no
; 启动
enabled = yes

asterisk -x 'module show like snmp'

/etc/snmp/snmpd.conf

# Asterisk user
createUser asteriskUser MD5 "<your password>" DES
rwuser asteriskUser priv

master agentx
agentXSocket	/var/agentx/master
agentXPerms 0660 0550 nobody asterisk

# snmp v2
snmpwalk -v2c -c public 127.0.0.1 1.3.6.1.4.1.22736.1
# snmp v3
snmpwalk -v 3 -u asteriskUser -l authPriv -a MD5 -A password DES -X password ASTERISK-MIB::astVersionString < your > -x < your > 127.0.0.1

常用配置

; debgging
load => app_verbose
load => app_dumpchan

load => res_rtp_multicast
load => chan_rtp
; rtp engine
load => res_rtp_asterisk
load => res_pjproject

load => pbx_config

; channel originate
load => res_clioriginate

load => res_curl
load => res_agi

; bridge
load => chan_bridge_media
load => app_bridgeaddchan
load => app_bridgewait
load => app_confbridge
load => bridge_builtin_features
load => bridge_builtin_interval_features
load => bridge_holding
load => bridge_native_rtp
load => bridge_simple
load => bridge_softmix

; load all codec
load => codec_a_mu
load => codec_adpcm
load => codec_alaw
; load => codec_dahdi ; need dahdi hardware
load => codec_g722
load => codec_g726
load => codec_gsm
load => codec_ilbc
load => codec_lpc10
load => codec_resample
load => codec_ulaw
; load all format
load => format_g719
load => format_g723
load => format_g726
load => format_g729
load => format_gsm
load => format_h263
load => format_h264
load => format_ilbc
load => format_pcm
load => format_siren14
load => format_siren7
load => format_sln
load => format_vox
load => format_wav
load => format_wav_gsm

\n## Reference List\n

Asterisk Modules

Tips

https://pkgs.alpinelinux.org/contents?file=*&path=%2Fusr%2Flib%2Fasterisk%2Fmodules&name=&branch=edge&arch=x86_64

AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Asterisk RESTful Interface", .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, .reload = reload_module, .optional_modules = "res_http_websocket", .requires = "http,res_stasis", .load_pri = AST_MODPRI_APP_DEPEND, );

res_phoneprov https://wiki.asterisk.org/wiki/display/AST/Phone+Provisioning+in+Asterisk https://wiki.asterisk.org/wiki/display/AST/Configuration+of+phoneprov.conf

/asterisk/phoneprov/... => Asterisk HTTP Phone Provisioning Tool

app_adsiprog.so app_agent_pool.so app_alarmreceiver.so app_amd.so Answering Machine Detection Application app_authenticate.so app_bridgeaddchan.so app_bridgewait.so app_cdr.so CDR app_celgenuserevent.so Cel Generate User Event app_chanisavail.so

app_channelredirect.so Channel Redirect app_chanspy.so Channel Spy app_confbridge.so ConfBridge app_controlplayback.so Controll Playback app_dahdiras.so DAHDi RAS app_db.so DB app_dial.so Dial app_dictate.so dicate app_directed_pickup.so app_directory.so app_disa.so app_dumpchan.so app_echo.so app_exec.so app_externalivr.so app_festival.so app_flash.so app_followme.so app_forkcdr.so app_getcpeid.so app_ices.so Encode and Stream via icecast and ices

app_image.so app_macro.so app_meetme.so app_milliwatt.so app_minivm.so app_mixmonitor.so app_morsecode.so app_mp3.so app_nbscat.so app_originate.so app_page.so app_playback.so app_playtones.so app_privacy.so app_queue.so app_read.so app_readexten.so app_record.so app_sayunixtime.so app_senddtmf.so app_sendtext.so app_sms.so app_softhangup.so app_speech_utils.so app_stack.so app_stasis.so app_system.so app_talkdetect.so app_test.so app_transfer.so app_url.so app_userevent.so app_verbose.so app_voicemail.so app_waitforring.so app_waitforsilence.so app_waituntil.so app_while.so app_zapateller.so

bridge_builtin_features.so bridge_builtin_interval_features.so bridge_holding.so bridge_native_rtp.so bridge_simple.so bridge_softmix.so

cdr_adaptive_odbc.so cdr_csv.so cdr_custom.so cdr_manager.so cdr_mysql.so cdr_odbc.so cdr_pgsql.so cdr_sqlite3_custom.so cdr_syslog.so Syslog 集成 cdr_tds.so MS SQL Server 存储

cel_custom.so 自定义 cel_manager.so 管理 cel_odbc.so ODBC 存储 cel_pgsql.so PostgreSQL 存储 cel_sqlite3_custom.so SQLite 自定义存储 cel_tds.so MS SQL Server 存储

chan_alsa.so chan_bridge_media.so chan_console.so chan_dahdi.so chan_dongle.so chan_iax2.so chan_mgcp.so chan_mobile.so chan_oss.so chan_phone.so chan_pjsip.so chan_rtp.so chan_sip.so chan_skinny.so https://en.wikipedia.org/wiki/Skinny_Call_Control_Protocol

chan_unistim.so codec_a_mu.so codec_adpcm.so codec_alaw.so codec_dahdi.so codec_g722.so codec_g726.so codec_gsm.so codec_ilbc.so codec_lpc10.so codec_resample.so codec_speex.so codec_ulaw.so

format_g719.so format_g723.so format_g726.so format_g729.so format_gsm.so format_h263.so format_h264.so format_ilbc.so format_jpeg.so format_ogg_speex.so format_pcm.so format_siren14.so format_siren7.so format_sln.so format_vox.so format_wav.so format_wav_gsm.so

func_aes.so func_audiohookinherit.so func_base64.so func_blacklist.so func_callcompletion.so func_callerid.so func_cdr.so func_channel.so func_config.so func_curl.so func_cut.so func_db.so func_devstate.so func_dialgroup.so func_dialplan.so func_enum.so func_env.so func_extstate.so func_frame_trace.so func_global.so func_groupcount.so func_hangupcause.so func_holdintercept.so func_iconv.so func_jitterbuffer.so func_lock.so func_logic.so func_math.so func_md5.so func_module.so func_odbc.so func_periodic_hook.so func_pitchshift.so func_pjsip_aor.so func_pjsip_contact.so func_pjsip_endpoint.so func_presencestate.so func_rand.so func_realtime.so func_sha1.so func_shell.so func_sorcery.so func_speex.so func_sprintf.so func_srv.so func_strings.so func_sysinfo.so func_talkdetect.so func_timeout.so func_uri.so func_version.so func_vmcount.so func_volume.so

pbx

https://github.com/asterisk/asterisk/blob/master/pbx

pbx_ael.so
pbx_config.so
- extensions.conf 配置文件
pbx_dundi.so
- dundi.conf
- 依赖 zlib 和 crypto
pbx_loopback.so
pbx_lua.so
- 使用 Lua 来写 dialplain
pbx_realtime.so
- 相当于把 extensions.conf 一行一行存到了数据库, 实际上不太好管理
pbx_spool.so
- 监听 spool 来发起通话

res_adsi.so res_ael_share.so res_agi.so

res_calendar.so res_clialiases.so res_clioriginate.so

res_config_curl.so res_config_odbc.so res_config_pgsql.so res_config_sqlite3.so

res_convert.so res_crypto.so res_curl.so

res_fax.so res_fax_spandsp.so

res_format_attr_celt.so res_format_attr_g729.so res_format_attr_h263.so res_format_attr_h264.so res_format_attr_ilbc.so res_format_attr_opus.so res_format_attr_silk.so res_format_attr_siren14.so res_format_attr_siren7.so res_format_attr_vp8.so

res_hep.so res_hep_pjsip.so res_hep_rtcp.so hep.conf

https://github.com/sipcapture/homer/wiki/Examples%3A-Asterisk

[general]
enabled = yes
capture_address = 10.0.0.1:9060
capture_password = foo
capture_id = 1234

res_http_media_cache.so res_http_websocket.so

res_limit.so res_manager_devicestate.so res_manager_presencestate.so res_monitor.so res_musiconhold.so res_mutestream.so res_odbc.so res_odbc_transaction.so res_parking.so res_phoneprov.so

res_realtime.so res_rtp_asterisk.so res_rtp_multicast.so res_security_log.so res_smdi.so

res_sorcery_astdb.so res_sorcery_config.so res_sorcery_memory.so res_sorcery_memory_cache.so res_sorcery_realtime.so

res_speech.so res_srtp.so

res_statsd.so res_stun_monitor.so res_timing_dahdi.so res_timing_pthread.so res_timing_timerfd.so

内置模块​

动态模块​

chan​

pbx​

codec​

format​

cdr​

cel​

bridge​

func​

app​

res​

res_stasis​

res_sorcery​

res_ari​

res_pjsip​

sip common​

res_statsd​

res_prometheus​

res_snmp​

配置​

常用配置​

Asterisk Modules

Tips​

pbx​

内置模块

动态模块

chan

pbx

codec

format

cdr

cel

bridge

func

app

res

res_stasis

res_sorcery

res_ari

res_pjsip

sip common

res_statsd

res_prometheus

res_snmp

配置

常用配置

Tips

pbx