参考
约 23 分钟阅读
Asterisk Reference
Tips
- chan-sccp/chan-sccpGitHubchan-sccp/chan-sccpchan-sccp/chan-sccp · http://chan-sccp-b.sourceforge.net/ · a replacement Channel Driver for chanskinny in the Asterisk Channel Driver Library笔记：参考
- chan-sccp-b.sourceforge.net
- a replacement Channel Driver for chan_skinny in the Asterisk Channel Driver Library
- BUGS
- [#26423]issues.asterisk.org/jira/browse/ASTERISK-26423 res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
Install
FAQ
user , peer, friend
- user 本端做验证, 呼入
- peer 远端做验证, 呼出
- friend 两端都要验证
- agent
- 用户代理
- 终端
- 代理服务
- 用户代理
Macro vs Sub
- 首选 Sub
- Macro 最多 7 层嵌套
Failed to insert call detail record into database
在使用 PJSIP_DIAL_CONTACTS 时, 号码可能非常长, 会导致数据库插入失败
[Sep 5 11:34:50] WARNING[11511]: cel_pgsql.c:351 pgsql_log: Failed to insert call detail record into database!
[Sep 5 11:34:50] WARNING[11511]: cel_pgsql.c:352 pgsql_log: Reason: ERROR: value too long for type character varying(80)对接 O 口网关时, 程序崩溃
可能是由于 UDP 消息截断导致, 打开日志可以看到消息内容应该只有一部分
CDR vs CEL
- 都可以对接后端存储
- CDR
- 相对信息更少
- CEL
- 支持用于账单
- Control over which Asterisk applications are tracked.
- Control over which events should be raised.
- Configurable date format.
- Integration with the Asterisk Manager Interface.
- Integration with RADIUS
- Modules for various logging back-ends including customized CEL output, integration with ODBC, PGSQL, SQLite and TDS.
DAHDi 有持续性的噪音
- 可能是打开了 crc4 导致的, 在
system.conf中关闭即可 - 如果有异常, 那也可能是 crc4 导致的
DAHDi 拨号选项
- channels/chan_dahdi.c#L13167GitHub Fileasterisk/asterisk/channels/chan_dahdi.c#L13167
Dial(DAHDI/pseudo[/extension[/options]])Dial(DAHDI/<channel#>[c|r<cadance#>|d][/extension[/options]])Dial(DAHDI/<subdir>!<channel#>[c|r<cadance#>|d][/extension[/options]])Dial(DAHDI/i<span>[/extension[/options]])Dial(DAHDI/[i<span>-](g|G|r|R)<group#(0-63)>[c|r<cadance#>|d][/extension[/options]])- i - ISDN span channel restriction.
- Used by CC to ensure that the CC recall goes out the same span.
- Also to make ISDN channel names dialable when the sequence number is stripped off. (Used by DTMF attended transfer feature.)
- g - channel group allocation search forward
- G - channel group allocation search backward
- r - channel group allocation round robin search forward
- R - channel group allocation round robin search backward
- c - Wait for DTMF digit to confirm answer
- r<cadance#> - Set distintive ring cadance number
- d - Force bearer capability for ISDN/SS7 call to digital.
PJ ICE Rx error status code: 370401
Core Dump
- Getting a Backtrace
- 需要安装
gdb - 除非编译时带了
DEBUG_THREADS, 否则locks为空 - 可以使用
libbfd, 在编译时加上DONT_OPTIMIZE,BETTER_BACKTRACES以获得更好的转储信息 - 默认转储文件位于当前目录下的
core, 会遵循kernel.core_pattern配置将转储存到指定的地方
sysctl -n kernel.core_pattern
/var/lib/asterisk/scripts/ast_coredumper coreProbation passed 后程序崩溃
- 在 cli.conf 中打开全量日志
res_rtp_asterisk.c: Unsupported payload type received
CEL 数据库写入失败, 字段过长
- 应该是 appdata 字段导致, 可以将数据库的长度改长
DAHDi
- asterisk/dahdi-linuxGitHubasterisk/dahdi-linux是什么？ · Digium Asterisk Hardware Device Interface · Digium 开发的 Asterisk 硬件设备接口 · DAHDI Telephony Interface Driver · dahdi tools 实际操作的是笔记：DAHDI
DTMF/Dual-tone multi-frequency
- DTMFWikipediaDual-tone multi-frequency signaling
- www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode
- www.voip-info.org/wiki/view/DTMF
- 可选模式包括
- inband
- rfs2833
- info
- auto
ADSI/Analog Display Services Interface
- www.voip-info.org/wiki/view/ADSI
- adsi.conf
- asterisk.adsi
HA
Database
# 源码中包含了操作数据库的脚本
cd ./contrib/ast-db-manage
pip install alembic
# 创建配置文件
cp config.ini.sample config.ini
# 调整配置项
# 主要配置 sqlalchemy.url
nano config.in
# 数据库更新到最新结构
alembic -c config.ini upgrade head
# 如果不想操作数据库, 也可以生成 SQL
alembic -c config.ini upgrade head --sql- sqlite
- render_as_batch=True
找不到 ENUM
- ASTERISK-27272 使用的那个版本配置可能有点问题, 在那个文件里添加以下内容即可
from sqlalchemy.dialects.postgresql import ENUM
YESNO_NAME = 'yesno_values'
YESNO_VALUES = ['yes', 'no']实时配置
- Realtime Database Configuration
- 实时模块主要是抽象数据层的访问, 是可以添加自定义的表的
; modules.conf
; 预先加载必须的模块
[modules]
preload => res_odbc.so
preload => res_config_odbc.so
; extconfig.conf
; 定义外部配置
[settings]
; 语法
; file.conf => driver,database[,table[,priority]]
meetme => odbc,general# 获取一条数据
realtime load sippeers name 9009
realtime load queues name marka
# 操作自定义的表
realtime load staffs no 8002cel.postgres.sql
CREATE TABLE cel (
id serial ,
eventtype varchar (30) NOT NULL ,
eventtime timestamp NOT NULL ,
userdeftype varchar(255) NOT NULL ,
cid_name varchar (80) NOT NULL ,
cid_num varchar (80) NOT NULL ,
cid_ani varchar (80) NOT NULL ,
cid_rdnis varchar (80) NOT NULL ,
cid_dnid varchar (80) NOT NULL ,
exten varchar (80) NOT NULL ,
context varchar (80) NOT NULL ,
channame varchar (80) NOT NULL ,
appname varchar (80) NOT NULL ,
appdata varchar (80) NOT NULL ,
amaflags int NOT NULL ,
accountcode varchar (20) NOT NULL ,
peeraccount varchar (20) NOT NULL ,
uniqueid varchar (150) NOT NULL ,
linkedid varchar (150) NOT NULL ,
userfield varchar (255) NOT NULL ,
peer varchar (80) NOT NULL
);Sorcery
- Sorcery
- Asterisk 12 添加
- 数据对象 CURD 抽象层
- Asterisk Database
- Static Configuration Files
- Asterisk Realtime Architecture
- In-Memory
- 提供了缓存服务, 用于从 ARI 推送配置
AST_SORCERY(module_name,object_type,object_id,field_name[,retrieval_method[,retrieval_details]])- 操作函数
- retrieval_method, 默认为 concat
- concat, 当有多条数据时进行拼接, 默认使用
, - single, 当有多条时返回一条记录, 默认为
1
- concat, 当有多条数据时进行拼接, 默认使用
- retrieval_details, 控制 concat 的连接符和 single 的位置
- retrieval_method, 默认为 concat
- 操作函数
- 先配置 extconfig.conf, 再配置 sorcery.conf 使用 extconfig 中定义的信息
func_sorcery.so Get a field from a sorcery object
res_sorcery_astdb.so Sorcery Astdb Object Wizard
res_sorcery_config.so Sorcery Configuration File Object Wizard
res_sorcery_memory.so Sorcery In-Memory Object Wizard
res_sorcery_memory_cache.so Sorcery Memory Cache Object Wizard
res_sorcery_realtime.so Sorcery Realtime Object Wizardpjsip 的默认配置
[res_pjsip]
auth=config,pjsip.conf,criteria=type=auth
domain_alias=config,pjsip.conf,criteria=type=domain_alias
global=config,pjsip.conf,criteria=type=global
system=config,pjsip.conf,criteria=type=system
transport=config,pjsip.conf,criteria=type=transport
aor=config,pjsip.conf,criteria=type=aor
endpoint=config,pjsip.conf,criteria=type=endpoint
contact=astdb,registrator
[res_pjsip_endpoint_identifier_ip]
identify=config,pjsip.conf,criteria=type=identify
[res_pjsip_outbound_publish]
outbound-publish=config,pjsip.conf,criteria=type=outbound-publish
[res_pjsip_outbound_registration]
registration=config,pjsip.conf,criteria=type=registrationpjsip 实时配置
_extconfig.conf
ps_aors => pgsql,asterisk
ps_asterisk_publications => pgsql,asterisk
ps_auths => pgsql,asterisk
ps_contacts => pgsql,asterisk
ps_domain_aliases => pgsql,asterisk
ps_endpoint_id_ips => pgsql,asterisk
ps_endpoints => pgsql,asterisk
ps_globals => pgsql,asterisk
ps_inbound_publications => pgsql,asterisk
ps_outbound_publishes => pgsql,asterisk
ps_registrations => pgsql,asterisk
ps_resource_list => pgsql,asterisk
ps_subscription_persistence => pgsql,asterisk
ps_systems => pgsql,asterisk
ps_transports => pgsql,asterisksorcery.conf
[res_pjsip]
auth =realtime,ps_auths
domain_alias=realtime,ps_domain_aliases
global =realtime,ps_globals
system =realtime,ps_systems
transport =realtime,ps_transports
aor =realtime,ps_aors
endpoint =realtime,ps_endpoints
contact =realtime,ps_contacts
[res_pjsip_endpoint_identifier_ip]
identify =realtime,ps_endpoint_id_ips
[res_pjsip_outbound_publish]
outbound-publish=realtime,ps_outbound_publishes
[res_pjsip_outbound_registration]
registration=realtime,ps_registrationsChannel
# 查看所有的通道类型
core show channeltypeschan_alsa/chan_console/chan_oss
- 同一时间只能启用一个, 默认启用 oss
- ALSA - Advanced Linux Sound Architecture
- OSS - Open Sound System
- Linux 2.6 默认为 ALSA, OSS 标记为废弃
- console - PortAudioWikipediaPortAudio
chan_sip
- 传输支持: tcp,udp,tls,ws,wss
- tcp 和 tls 位于实验阶段
- 5060: This is the standard port for SIP communications
- 8089: This is the standard port for Secure Websockets when used with Asterisk's built-in HTTP sever
- 10000:20000: This is the port range configured in rtp.conf for audio to flow.
QoS
- IP Quality of Service
- 默认未开启 QoS, 所以 Peer 的状态都是显示的未知
res_pjsip
- 总结: 建议使用
res_pjsip, 禁用chan_sip - SIP vs. CHAN_SIP vs. CHAN_PJSIP
- Migrating from chan sip to res pjsip
- 幻灯片
- astricon2015
- 脚本和相关文件配置
- 包含数据库
sippeers转ps_的脚本
- PJSIP: Tuning for Performance
- WHY
- 更好的配置
- 多个 aors -> 单个终端
- 多个终端共振
- NAT 更简单
- 没有 user,peer,friend
- 更好的设备和邮箱状态
- 更简单更快更好的开发
- 配置类型
- transport
- 绑定 res_pjsip 到地址端口
- 可绑定多个
- 不能重载
- endpoint
- 发起和接收通话的设备
- 包含: transport, aor, auth
- 配置的 transport 主要用于发送, 所有的都能接收
- auth
auth_type- 授权类型
- nonce_lifetime
- 单位 秒
- 默认 32
- userpass
- password 存储明文
- md5
- md5_cred 存储密文
- 格式为 账号:asterisk
- asterisk 为 realm, 可以修改
- aor
- Address of Record
- Multiple AORS for 1 device
- AORS can be overwritten or not
- Can be static or dynamic (qualify)
- identify
- Endpoint Identification
res_pjsip_endpoint_identifier_ip- 基于 IP 的认证
- 匹配进入的包 -> 终端
res_pjsip_endpoint_identifier_user- 可以从
From头中提取出用户信息由于验证
- 可以从
- 使用 IP 还是用用户取决于模块加载顺序
- 用于外部线路，直接匹配 IP 来对应 endpoint
- registration
- 将 Asterisk 连接到另外一个 Asterisk
- 以前为
register => username:password@server/context
- acl
- Access Control List
- phoneprov
- Phone Provisioning
- System
- Domain alias
- 域名别名
- 在 AOR 的域名找不到时尝试找别名
- outbound-publish
- transport
- Tips
Dial(PJSIP/${EXTEN})Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})})- 拨打所有设备
PJSIP/9001/sip:[email protected]:33322&PJSIP/9001/sip:[email protected]:58069
Dial(PJSIP/mytrunk/sip:${EXTEN:1})Dial(PJSIP/${EXTEN:1}@mytrunk)
- PJSIP 使用的表前缀为
ps_, 有很多表, 而不像 chan_sip 只有一个 sippeers 表- aors, auths, contacts, endpoints, domain_aliases, endpoint_id_ips, globals, registrations, subscription_persistence, systems, transports
# 问题排查
core set verbose 4
core set debug 4
pjsip set logger on[ts-udp]
type=transport
protocol=udp
bind=0.0.0.0
[ts-tcp]
type=transport
protocol=tcp
bind=0.0.0.0
[trans-more]
type=transport
protocol=udp,tcp,tls,ws,wss
bind=0.0.0.0:5061
local_net=192.0.2.0/2
external_medial_address=20.0.113.1
external_signaling_address=20.0.113.1
[6001]
type=endpoint
context=default
disallow=all
allow=ulw
transport=trans-one
auth=auth6001
aors=6001
; aor static
[6001]
type=aor
contact=sip:[email protected]:5060
contact=sip:[email protected]:5060
; aor dynamic
[6001]
type=aor
default_exporation=3600
maximum_exporation=7200
minimum_exporation=60
max_contacts=1
remove_existing=yes
qualify_frequency=60
qualify_timeout=3.0
[auth6001]
type=auth
auth_type=userpass
password=secret
username=6001
[mytrunk]
type=registration
transport=simpletrans
outbound_auth=mytrunk
server_uri=sip:sip.example.com
client_uri=sip:[email protected]
retry_interval=60
[my-itsip]
type=wizard
sends_auth=yes
sends_registrations=yes
remote_hosts=sip.my-itsp.net
outbound_auth/username=my_username
outbound_auth/password=my_password
endpoint/context=default
aor/qualify_frequency=15
; 简单的用户模板
[user-template](!)
type = wizard
accepts_registrations = yes
accepts_auth = yes
endpoint/context = default
endpoint/allow = !all,ulaw,gsm,g722
aor/max_contacts=5
[9001](user-template)
inbound_auth/username = 9001
inbound_auth/password = 9001
[9002](user-template)
inbound_auth/username = 9002
inbound_auth/password = 9002禁用
noload => res_pjsip.so
noload => res_pjsip_pubsub.so
noload => res_pjsip_session.so
noload => chan_pjsip.so
noload => res_pjsip_exten_state.so
noload => res_pjsip_log_forwarder.sowizard
配置
- qualify_frequency
- QoS
NAT
- rtp_symmetric
- Send media to the address and port from which Asterisk receives it, regardless of where SDP indicates that it should be sent
- force_rport
- Send responses to the source IP address and port as though port were present, even if it's not
- rewrite_contact
- Rewrite SIP Contact to the source address and port of the request so that subsequent requests go to that address and port.
- Configuring res_pjsip to work through NAT
| chan_sip/nat | yes | no | never | route |
|---|---|---|---|---|
| rtp_symmetric | yes | no | no | no |
| force_rport | yes | no | no | yes |
| rewrite_contact | yes | no | no | yes |
Modules
- Tips
module show显示所有模块
app_adsiprog.so Asterisk ADSI Programming Application 0 Running extended
app_agent_pool.so Call center agent pool applications 0 Running core
app_alarmreceiver.so Alarm Receiver for Asterisk 0 Running extended
app_amd.so Answering Machine Detection Application 0 Running extended
app_authenticate.so Authentication Application 0 Running core
app_bridgeaddchan.so Bridge Add Channel Application 0 Running core
app_bridgewait.so Place the channel into a holding bridge 0 Running core
app_cdr.so Tell Asterisk to not maintain a CDR for 0 Running core
app_celgenuserevent.so Generate an User-Defined CEL event 0 Running core
app_chanisavail.so Check channel availability 0 Running extended
app_channelredirect.so Redirects a given channel to a dialplan 0 Running core
app_chanspy.so Listen to the audio of an active channel 0 Running core
app_confbridge.so Conference Bridge Application 0 Running core
app_controlplayback.so Control Playback Application 0 Running core
app_dahdiras.so DAHDI ISDN Remote Access Server 0 Running extended
app_db.so Database Access Functions 0 Running core
app_dial.so Dialing Application 0 Running core
app_dictate.so Virtual Dictation Machine 0 Running extended
app_directed_pickup.so Directed Call Pickup Application 0 Running core
app_directory.so Extension Directory 0 Running core
app_disa.so DISA (Direct Inward System Access) Appli 0 Running core
app_dumpchan.so Dump Info About The Calling Channel 0 Running core
app_echo.so Simple Echo Application 0 Running core
app_exec.so Executes dialplan applications 0 Running core
app_externalivr.so External IVR Interface Application 0 Running extended
app_festival.so Simple Festival Interface 0 Running extended
app_flash.so Flash channel application 0 Running core
app_followme.so Find-Me/Follow-Me Application 0 Running core
app_forkcdr.so Fork The CDR into 2 separate entities 0 Running core
app_getcpeid.so Get ADSI CPE ID 0 Running extended
app_ices.so Encode and Stream via icecast and ices 0 Running extended
app_image.so Image Transmission Application 0 Running extended
app_macro.so Extension Macros 0 Running core
app_meetme.so MeetMe conference bridge 0 Running extended
app_milliwatt.so Digital Milliwatt (mu-law) Test Applicat 0 Running core
app_minivm.so Mini VoiceMail (A minimal Voicemail e-ma 0 Running extended
app_mixmonitor.so Mixed Audio Monitoring Application 0 Running core
app_morsecode.so Morse code 0 Running extended
app_mp3.so Silly MP3 Application 0 Running extended
app_nbscat.so Silly NBS Stream Application 0 Running extended
app_originate.so Originate call 0 Running core
app_page.so Page Multiple Phones 0 Running core
app_playback.so Sound File Playback Application 0 Running core
app_playtones.so Playtones Application 0 Running core
app_privacy.so Require phone number to be entered, if n 0 Running core
app_queue.so True Call Queueing 0 Running core
app_read.so Read Variable Application 0 Running core
app_readexten.so Read and evaluate extension validity 0 Running core
app_record.so Trivial Record Application 0 Running core
app_sayunixtime.so Say time 0 Running core
app_senddtmf.so Send DTMF digits Application 0 Running core
app_sendtext.so Send Text Applications 0 Running core
app_sms.so SMS/PSTN handler 0 Running extended
app_softhangup.so Hangs up the requested channel 0 Running core
app_speech_utils.so Dialplan Speech Applications 0 Running core
app_stack.so Dialplan subroutines (Gosub, Return, etc 0 Running core
app_stasis.so Stasis dialplan application 0 Running core
app_system.so Generic System() application 0 Running core
app_talkdetect.so Playback with Talk Detection 0 Running extended
app_test.so Interface Test Application 0 Running extended
app_transfer.so Transfers a caller to another extension 0 Running core
app_url.so Send URL Applications 0 Running extended
app_userevent.so Custom User Event Application 0 Running core
app_verbose.so Send verbose output 0 Running core
app_voicemail.so Comedian Mail (Voicemail System) with IM 0 Running core
app_waitforring.so Waits until first ring after time 0 Running extended
app_waitforsilence.so Wait For Silence 0 Running extended
app_waituntil.so Wait until specified time 0 Running core
app_while.so While Loops and Conditional Execution 0 Running core
app_zapateller.so Block Telemarketers with Special Informa 0 Running extended
bridge_builtin_features.so Built in bridging features 1 Running core
bridge_builtin_interval_features.so Built in bridging interval features 0 Running core
bridge_holding.so Holding bridge module 0 Running core
bridge_native_rtp.so Native RTP bridging module 0 Running core
bridge_simple.so Simple two channel bridging module 0 Running core
bridge_softmix.so Multi-party software based channel mixin 0 Running core
cdr_csv.so Comma Separated Values CDR Backend 0 Running extended
cdr_custom.so Customizable Comma Separated Values CDR 0 Running core
cdr_manager.so Asterisk Manager Interface CDR Backend 0 Running core
cdr_pgsql.so PostgreSQL CDR Backend 0 Running extended
cdr_syslog.so Customizable syslog CDR Backend 0 Not Running core
cel_manager.so Asterisk Manager Interface CEL Backend 0 Running core
cel_pgsql.so PostgreSQL CEL Backend 0 Running extended
chan_bridge_media.so Bridge Media Channel Driver 0 Running core
chan_dahdi.so DAHDI Telephony w/PRI 0 Running core
chan_iax2.so Inter Asterisk eXchange (Ver 2) 0 Running core
chan_pjsip.so PJSIP Channel Driver 0 Running core
chan_rtp.so RTP Media Channel 0 Running core
codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 Running core
codec_adpcm.so Adaptive Differential PCM Coder/Decoder 0 Running core
codec_alaw.so A-law Coder/Decoder 0 Running core
codec_dahdi.so Generic DAHDI Transcoder Codec Translato 0 Running core
codec_g722.so ITU G.722-64kbps G722 Transcoder 0 Running core
codec_g726.so ITU G.726-32kbps G726 Transcoder 0 Running core
codec_gsm.so GSM Coder/Decoder 0 Running core
codec_ilbc.so iLBC Coder/Decoder 0 Running core
codec_lpc10.so LPC10 2.4kbps Coder/Decoder 0 Running core
codec_resample.so SLIN Resampling Codec 0 Running core
codec_ulaw.so mu-Law Coder/Decoder 0 Running core
format_g719.so ITU G.719 0 Running core
format_g723.so G.723.1 Simple Timestamp File Format 0 Running core
format_g726.so Raw G.726 (16/24/32/40kbps) data 0 Running core
format_g729.so Raw G.729 data 0 Running core
format_gsm.so Raw GSM data 0 Running core
format_h263.so Raw H.263 data 0 Running core
format_h264.so Raw H.264 data 0 Running core
format_ilbc.so Raw iLBC data 0 Running core
format_jpeg.so jpeg (joint picture experts group) image 0 Running extended
format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0 Running core
format_siren14.so ITU G.722.1 Annex C (Siren14, licensed f 0 Running core
format_siren7.so ITU G.722.1 (Siren7, licensed from Polyc 0 Running core
format_sln.so Raw Signed Linear Audio support (SLN) 8k 0 Running core
format_vox.so Dialogic VOX (ADPCM) File Format 0 Running extended
format_wav.so Microsoft WAV/WAV16 format (8kHz/16kHz S 0 Running core
format_wav_gsm.so Microsoft WAV format (Proprietary GSM) 0 Running core
func_aes.so AES dialplan functions 0 Running core
func_audiohookinherit.so Audiohook inheritance placeholder functi 0 Running deprecated
func_base64.so base64 encode/decode dialplan functions 0 Running core
func_blacklist.so Look up Caller*ID name/number from black 0 Running core
func_callcompletion.so Call Control Configuration Function 0 Running core
func_callerid.so Party ID related dialplan functions (Cal 0 Running core
func_cdr.so Call Detail Record (CDR) dialplan functi 0 Running core
func_channel.so Channel information dialplan functions 0 Running core
func_config.so Asterisk configuration file variable acc 0 Running core
func_curl.so Load external URL 0 Running core
func_cut.so Cut out information from a string 0 Running core
func_db.so Database (astdb) related dialplan functi 0 Running core
func_devstate.so Gets or sets a device state in the dialp 0 Running core
func_dialgroup.so Dialgroup dialplan function 0 Running core
func_dialplan.so Dialplan Context/Extension/Priority Chec 0 Running core
func_enum.so ENUM related dialplan functions 0 Running core
func_env.so Environment/filesystem dialplan function 0 Running core
func_extstate.so Gets an extension's state in the dialpla 0 Running core
func_frame_trace.so Frame Trace for internal ast_frame debug 0 Running extended
func_global.so Variable dialplan functions 0 Running core
func_groupcount.so Channel group dialplan functions 0 Running core
func_hangupcause.so HANGUPCAUSE related functions and applic 0 Running core
func_holdintercept.so Hold interception dialplan function 0 Running core
func_iconv.so Charset conversions 0 Running core
func_jitterbuffer.so Jitter buffer for read side of channel. 0 Running core
func_lock.so Dialplan mutexes 0 Running core
func_logic.so Logical dialplan functions 0 Running core
func_math.so Mathematical dialplan function 0 Running core
func_md5.so MD5 digest dialplan functions 0 Running core
func_module.so Checks if Asterisk module is loaded in m 0 Running core
func_periodic_hook.so Periodic dialplan hooks. 0 Running core
func_pitchshift.so Audio Effects Dialplan Functions 0 Running extended
func_pjsip_aor.so Get information about a PJSIP AOR 0 Running core
func_pjsip_contact.so Get information about a PJSIP contact 0 Running core
func_pjsip_endpoint.so Get information about a PJSIP endpoint 0 Running core
func_presencestate.so Gets or sets a presence state in the dia 0 Running core
func_rand.so Random number dialplan function 0 Running core
func_realtime.so Read/Write/Store/Destroy values from a R 0 Running core
func_sha1.so SHA-1 computation dialplan function 0 Running core
func_shell.so Collects the output generated by a comma 0 Running core
func_sorcery.so Get a field from a sorcery object 0 Running core
func_sprintf.so SPRINTF dialplan function 0 Running core
func_srv.so SRV related dialplan functions 0 Running core
func_strings.so String handling dialplan functions 0 Running core
func_sysinfo.so System information related functions 0 Running core
func_talkdetect.so Talk detection dialplan function 0 Running core
func_timeout.so Channel timeout dialplan functions 0 Running core
func_uri.so URI encode/decode dialplan functions 0 Running core
func_version.so Get Asterisk Version/Build Info 0 Running core
func_vmcount.so Indicator for whether a voice mailbox ha 0 Running core
func_volume.so Technology independent volume control 0 Running core
pbx_ael.so Asterisk Extension Language Compiler 0 Not Running extended
pbx_config.so Text Extension Configuration 0 Running core
pbx_loopback.so Loopback Switch 0 Running core
pbx_lua.so Lua PBX Switch 0 Not Running extended
pbx_realtime.so Realtime Switch 0 Running extended
pbx_spool.so Outgoing Spool Support 0 Running core
res_adsi.so ADSI Resource 0 Running core
res_ael_share.so share-able code for AEL 0 Running extended
res_agi.so Asterisk Gateway Interface (AGI) 1 Running core
res_ari.so Asterisk RESTful Interface 10 Running core
res_ari_applications.so RESTful API module - Stasis application 0 Running core
res_ari_asterisk.so RESTful API module - Asterisk resources 0 Running core
res_ari_bridges.so RESTful API module - Bridge resources 0 Running core
res_ari_channels.so RESTful API module - Channel resources 0 Running core
res_ari_device_states.so RESTful API module - Device state resour 0 Running core
res_ari_endpoints.so RESTful API module - Endpoint resources 0 Running core
res_ari_events.so RESTful API module - WebSocket resource 0 Running core
res_ari_model.so ARI Model validators 0 Running core
res_ari_playbacks.so RESTful API module - Playback control re 0 Running core
res_ari_recordings.so RESTful API module - Recording resources 0 Running core
res_ari_sounds.so RESTful API module - Sound resources 0 Running core
res_calendar.so Asterisk Calendar integration 0 Running core
res_clialiases.so CLI Aliases 0 Running core
res_clioriginate.so Call origination and redirection from th 0 Running core
res_config_curl.so Realtime Curl configuration 0 Running core
res_config_pgsql.so PostgreSQL RealTime Configuration Driver 0 Running extended
res_convert.so File format conversion CLI command 0 Running core
res_crypto.so Cryptographic Digital Signatures 1 Running core
res_curl.so cURL Resource Module 0 Running core
res_format_attr_celt.so CELT Format Attribute Module 1 Running core
res_format_attr_g729.so G.729 Format Attribute Module 1 Running core
res_format_attr_h263.so H.263 Format Attribute Module 1 Running core
res_format_attr_h264.so H.264 Format Attribute Module 1 Running core
res_format_attr_ilbc.so iLBC Format Attribute Module 1 Running core
res_format_attr_opus.so Opus Format Attribute Module 1 Running core
res_format_attr_silk.so SILK Format Attribute Module 1 Running core
res_format_attr_siren14.so Siren14 Format Attribute Module 1 Running core
res_format_attr_siren7.so Siren7 Format Attribute Module 1 Running core
res_format_attr_vp8.so VP8 Format Attribute Module 1 Running core
res_hep.so HEPv3 API 0 Running extended
res_hep_pjsip.so PJSIP HEPv3 Logger 0 Running extended
res_hep_rtcp.so RTCP HEPv3 Logger 0 Running unknown
res_http_media_cache.so HTTP Media Cache Backend 1 Running core
res_http_websocket.so HTTP WebSocket Support 2 Running extended
res_limit.so Resource limits 0 Running core
res_manager_devicestate.so Manager Device State Topic Forwarder 0 Running core
res_manager_presencestate.so Manager Presence State Topic Forwarder 0 Running core
res_monitor.so Call Monitoring Resource 0 Running core
res_musiconhold.so Music On Hold Resource 0 Running core
res_mutestream.so Mute audio stream resources 0 Running core
res_parking.so Call Parking Resource 0 Running core
res_phoneprov.so HTTP Phone Provisioning 0 Running extended
res_pjproject.so PJPROJECT Log and Utility Support 1 Running core
res_pjsip.so Basic SIP resource 26 Running core
res_pjsip_acl.so PJSIP ACL Resource 0 Running core
res_pjsip_authenticator_digest.so PJSIP authentication resource 0 Running core
res_pjsip_caller_id.so PJSIP Caller ID Support 0 Running core
res_pjsip_config_wizard.so PJSIP Config Wizard 1 Running core
res_pjsip_dialog_info_body_generator.so PJSIP Extension State Dialog Info+XML Pr 0 Running core
res_pjsip_diversion.so PJSIP Add Diversion Header Support 0 Running core
res_pjsip_dlg_options.so SIP OPTIONS in dialog handler 0 Running unknown
res_pjsip_dtmf_info.so PJSIP DTMF INFO Support 0 Running core
res_pjsip_empty_info.so PJSIP Empty INFO Support 0 Running core
res_pjsip_endpoint_identifier_anonymous.so PJSIP Anonymous endpoint identifier 0 Running core
res_pjsip_endpoint_identifier_ip.so PJSIP IP endpoint identifier 0 Running core
res_pjsip_endpoint_identifier_user.so PJSIP username endpoint identifier0 Running core
res_pjsip_exten_state.so PJSIP Extension State Notifications 0 Running core
res_pjsip_header_funcs.so PJSIP Header Functions 0 Running core
res_pjsip_history.so PJSIP History 0 Running extended
res_pjsip_logger.so PJSIP Packet Logger 0 Running core
res_pjsip_messaging.so PJSIP Messaging Support 0 Running core
res_pjsip_multihomed.so PJSIP Multihomed Routing Support 0 Running core
res_pjsip_mwi.so PJSIP MWI resource 0 Running core
res_pjsip_mwi_body_generator.so PJSIP MWI resource 0 Running core
res_pjsip_nat.so PJSIP NAT Support 0 Running core
res_pjsip_notify.so CLI/AMI PJSIP NOTIFY Support 0 Running core
res_pjsip_one_touch_record_info.so PJSIP INFO One Touch Recording Support 0 Running core
res_pjsip_outbound_authenticator_digest.so PJSIP authentication resource0 Running core
res_pjsip_outbound_publish.so PJSIP Outbound Publish Support 4 Running unknown
res_pjsip_outbound_registration.so PJSIP Outbound Registration Support 0 Running core
res_pjsip_path.so PJSIP Path Header Support 0 Running core
res_pjsip_phoneprov_provider.so PJSIP Phoneprov Provider 0 Running unknown
res_pjsip_pidf_body_generator.so PJSIP Extension State PIDF Provider 0 Running core
res_pjsip_pidf_digium_body_supplement.so PJSIP PIDF Digium presence supplement 0 Running core
res_pjsip_pidf_eyebeam_body_supplement.so PJSIP PIDF Eyebeam supplement 0 Running core
res_pjsip_publish_asterisk.so PJSIP Asterisk Event PUBLISH Support 0 Running unknown
res_pjsip_pubsub.so PJSIP event resource 5 Running core
res_pjsip_refer.so PJSIP Blind and Attended Transfer Suppor 0 Running core
res_pjsip_registrar.so PJSIP Registrar Support 0 Running core
res_pjsip_registrar_expire.so PJSIP Contact Auto-Expiration 0 Running core
res_pjsip_rfc3326.so PJSIP RFC3326 Support 0 Running core
res_pjsip_sdp_rtp.so PJSIP SDP RTP/AVP stream handler 0 Running core
res_pjsip_send_to_voicemail.so PJSIP REFER Send to Voicemail Support 0 Running core
res_pjsip_session.so PJSIP Session resource 25 Running core
res_pjsip_sips_contact.so UAC SIPS Contact support 0 Running core
res_pjsip_t38.so PJSIP T.38 UDPTL Support 0 Running core
res_pjsip_transport_management.so PJSIP Reliable Transport Management 1 Running core
res_pjsip_transport_websocket.so PJSIP WebSocket Transport Support 0 Running core
res_pjsip_xpidf_body_generator.so PJSIP Extension State PIDF Provider 0 Running core
res_realtime.so Realtime Data Lookup/Rewrite 0 Running core
res_rtp_asterisk.so Asterisk RTP Stack 0 Running core
res_rtp_multicast.so Multicast RTP Engine 0 Running core
res_security_log.so Security Event Logging 0 Running core
res_smdi.so Simplified Message Desk Interface (SMDI) 0 Running core
res_sorcery_astdb.so Sorcery Astdb Object Wizard 2 Running core
res_sorcery_config.so Sorcery Configuration File Object Wizard 16 Running core
res_sorcery_memory.so Sorcery In-Memory Object Wizard 8 Running core
res_sorcery_memory_cache.so Sorcery Memory Cache Object Wizard 0 Running core
res_sorcery_realtime.so Sorcery Realtime Object Wizard 0 Running core
res_speech.so Generic Speech Recognition API 0 Running core
res_srtp.so Secure RTP (SRTP) 0 Running core
res_stasis.so Stasis application support 12 Running core
res_stasis_answer.so Stasis application answer support 0 Running core
res_stasis_device_state.so Stasis application device state support 0 Running core
res_stasis_playback.so Stasis application playback support 0 Running core
res_stasis_recording.so Stasis application recording support 0 Running core
res_stasis_snoop.so Stasis application snoop support 0 Running core
res_statsd.so Statsd client support 0 Running extended
res_stun_monitor.so STUN Network Monitor 0 Running core
res_timing_dahdi.so DAHDI Timing Interface 0 Not Running core
res_timing_pthread.so pthread Timing Interface 0 Running extended
res_timing_timerfd.so Timerfd Timing Interface 1 Running core关联信息
反向链接和本文引用的外部资料。
References
GitHub
4 条- asterisk/asterisk/channels/chan_dahdi.c#L13167github.com/asterisk/asterisk/blob/master/channels/chan_dahdi.c
- asterisk/asterisk/res/res_rtp_asterisk.c#L5109github.com/asterisk/asterisk/blob/master/res/res_rtp_asterisk.c
- asterisk/dahdi-linuxgithub.com/asterisk/dahdi-linux
- chan-sccp/chan-sccpgithub.com/chan-sccp/chan-sccp
Wikipedia
2 条- Dual-tone multi-frequency signalingWikipedia:Dual-tone multi-frequency signaling
- PortAudioWikipedia:PortAudio
其他外链
20 条- issues.asterisk.org/jira/browse/ASTERISK-26423issues.asterisk.org/jira/browse/ASTERISK-26423
- asteriskfaqs.org/2017/03/14/asterisk-users/codec-negotiation-or-transcoding-issue.htmlasteriskfaqs.org/2017/03/14/asterisk-users/codec-negotiation-or-transcoding-issue.html
- chan-sccp-b.sourceforge.netchan-sccp-b.sourceforge.net/
- git.alpinelinux.org/cgit/aports/tree/main/asterisk/APKBUILDgit.alpinelinux.org/cgit/aports/tree/main/asterisk/APKBUILD
- issues.asterisk.org/jira/browse/ASTERISK-25676issues.asterisk.org/jira/browse/ASTERISK-25676
- issues.asterisk.org/jira/browse/ASTERISK-27272issues.asterisk.org/jira/browse/ASTERISK-27272
- wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NATwiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT
- wiki.asterisk.org/wiki/display/AST/Getting+a+Backtracewiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
另有 12 个未显示 references。
另有 12 个 references 未显示。