Asterisk (PBX)

敬告1: All packages here depend on the package asterisk18!
友示2: All packages have Version

Asterisk Version
名称 依赖 大小 描述
asterisk18 libopenssl, libncurses,
libpopt, libpthread, zlib
997477 Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does VoIP in three protocols, and can inter-operate with almost all standards-based telephony equipment using relatively inexpensive hardware.
asterisk18-format-g726 3780 Support for headerless G.726 16/24/32/40kbps data format in Asterisk.
asterisk18-format-g729 3571 Support for raw headerless G729 data in Asterisk.
asterisk18-codec-g722 6407 A high bit rate 48/56/64Kbps ITU standard CODEC in Asterisk.
asterisk18-codec-g726 5012 Translation between signed linear and ITU G.726-32kbps codecs in Asterisk.
asterisk18-codec-a-mu 2934 Translation between alaw and ulaw codecs in Asterisk.
asterisk18-codec-ulaw 3150 Translation between signed linear and ulaw codecs in Asterisk.
asterisk18-codec-alaw 3069 Translation between signed linear and alaw codecs in Asterisk.
asterisk18-chan-local 11586 An implementation of local proxy channel in Asterisk.
asterisk18-chan-agent 22519 An implementation of agents proxy channel in Asterisk.
asterisk18-chan-gtalk libiksemel 58144 This package provides the channel chan_gtalk and res_jabber for GTalk support to Asterisk.
asterisk18-chan-iax2 asterisk18-res-crypto 133735 This package provides IAX support to Asterisk.
asterisk18-chan-skinny 52033 This package provides the channel chan_skinny support to Asterisk.
asterisk18-chan-mobile bluez-libs 30089 This package provides the channel chan_mobile support to Asterisk.
asterisk18-chan-mgcp 42959 This package provides the channel chan_mgcp support to Asterisk.
asterisk18-pbx-spool 9034 outgoing call spool support in Asterisk.
asterisk18-pbx-ael 4012 support for symbolic Asterisk Extension Logic in Asterisk.
asterisk18-voicemail 595818 This package contains voicemail related modules for Asterisk.
asterisk18-mysql libmysqlclient 36817 This package provides MySQL support to Asterisk.
asterisk18-curl libcurl 8447 This package provides CURL support to Asterisk.
asterisk18-cdr 36202 This package provides Call Detail Record support to Asterisk.
asterisk18-sounds 1058667 This package contains sound files for Asterisk.
asterisk18-app-sms 16204 SMS support (ETSI ES 201 912 protocol 1) in Asterisk.
asterisk18-app-authenticate 3938 support for executing arbitrary authenticate commands in Asterisk.
asterisk18-app-exec 3892 support for application execution in Asterisk.
asterisk18-app-while 4851 a while loop implementation in Asterisk.
asterisk18-app-waituntil 3104 support sleeping until the given epoch in Asterisk.
asterisk18-app-stack asterisk18-res-agi 7980 stack applications Gosub Return etc. in Asterisk.
asterisk18-app-sayunixtime 2890 an application to say Unix time in Asterisk.
asterisk18-app-read 4359 a trivial application to read a variable in Asterisk.
asterisk18-app-directed-pickup 5365 support for directed call pickup in Asterisk.
asterisk18-app-setcallerid 2795 support for setting callerid in Asterisk.
asterisk18-app-originate 3857 originating an outbound call and connecting it to a specified extension or application in Asterisk.
asterisk18-app-system 3312 support for executing system commands in Asterisk.
asterisk18-app-talkdetect 4662 for file playback with audio detect in Asterisk.
asterisk18-app-readexten 4668 a trivial application to read an extension into a variable in Asterisk.
asterisk18-app-verbose 3388 Verbose logging application in Asterisk.
asterisk18-app-alarmreceiver 7605 Central Station Alarm receiver for Ademco Contact ID in Asterisk.
asterisk18-app-chanspy 9776 support for listening in on any channel in Asterisk.
asterisk18-app-chanisavail 3572 support for checking if a channel is available in Asterisk.
asterisk18-app-minivm 27656 a voicemail system in small building blocks working together based on the Comedian Mail voicemail system in Asterisk.
asterisk18-func-uri 2769 Encodes and decodes URI-safe strings in Asterisk.
asterisk18-func-global 4536 global variable dialplan functions in Asterisk.
asterisk18-func-blacklist 2960 looking up the callerid number and see if it is blacklisted in Asterisk.
asterisk18-func-channel 7116 Channel info dialplan function in Asterisk.
asterisk18-func-db 3504 functions for interaction with the database in Asterisk.
asterisk18-func-extstate 2947 retrieving the state of a hinted extension for dialplan control in Asterisk.
asterisk18-func-shell 2860 support for shell execution in Asterisk.
asterisk18-func-devstate 4632 functions for manually controlled blinky lights in Asterisk.
asterisk18-func-vmcount 2787 a vmcount dialplan function in Asterisk.
asterisk18-format-sln16 3471 support for Raw slinear 16 format in Asterisk.
asterisk18-format-sln 3459 support for raw slinear format in Asterisk.
asterisk18-res-musiconhold 17432 This package provides Music On Hold support to Asterisk.
asterisk18-res-agi 26069 support for the Asterisk Gateway Interface extension in Asterisk.
asterisk18-res-crypto 7785 Cryptographic Signature capability in Asterisk.
asterisk18-res-ael-share 42733 support for shareable AEL code mainly between internal and external modules in Asterisk.
libopenssl 1.0.0d-1 zlib 592118 The OpenSSL Project is a collaborative effort to develop a robust, commercial-grade, full-featured, and Open Source toolkit implementing the Secure Sockets Layer (SSL v2/v3) and Transport Layer Security (TLS v1) protocols as well as a full-strength general purpose cryptography library. This package contains the OpenSSL shared libraries, needed by other programs.
libncurses 5.7-5 terminfo 109113 Terminal handling library
libpopt 1.7-5 13796 A command line option parsing library
libpthread 0.9.32-81 30505 POSIX thread library
zlib 1.2.5-1 40519 Library implementing the deflate compression method
terminfo 5.7-5 6056 Terminal Info Database (ncurses)
opkg update
opkg install asterisk18

注意: certain timezone related functions in asterisk assume a valid /etc/localtime file. This normally doesn't exist on OpenWRT, so you may want to set it up yourself to point to the TZ file describing the timezone you need. You can either just copy it from a Linux install from /usr/share/zoneinfo/ or install a limited opkg install zoneinfo-core OpenWRT package.

  • 安装前 检查空间能装得下么?是否有充足空余磁盘空间:
    root@openwrt:/etc$ df
    Filesystem           1K-blocks      Used Available 已用% Mounted on
    /dev/mtdblock3            5248      1344      3904  26% /overlay
    overlayfs:/overlay        5248      1344      3904  26% /
  • 安装后 原来已用26%,现在已用61%,对比一下:
    Filesystem           1K-blocks      Used Available 已用% Mounted on
    /dev/mtdblock3            5248      3192      2056  61% /overlay
    overlayfs:/overlay        5248      3192      2056  61% /

There is no UCI configuration file for the asterisk package, but there is a LuCI based front-end for Asterisk.


If you are using SIP, most of the configuration is done in sip.conf and the dialplan in extensions.conf. If you do not have a SIP phone, you can also use a soft client on your PC.

VoIP traffic is very time dependant, so make sure your internet connection has a delay below 100ms. You may also want to configure QoS (diffserv) to prioritize your SIP and RTP packets, especially if you are heavy using it together with other services.

Most people recommended to setup port forwarding (SIP UDP and RTP UDP) when NAT is used, but i have found it is not always needed.

Here is a very short, basic and tested example for one sip phone and a sip peer.

Example Working Tested Config
Jan 1, 2014
Firmware Version OpenWrt Attitude Adjustment 12.09 / LuCI 0.11.1 Release (0.11.1)
SIP Provider for DID
Use the following command to start asterisk: asterisk -gvvc
NPANXXXXXX is your actual DID
184XXXXXXX is the actual trunk reg from
XXXXX is the actual pwd/secret from

exten => 301, 1, Dial(SIP/301, 20)
exten => 301, 2, VoiceMail(301,u)
exten => 302, 1, Dial(SIP/302, 20)
exten => 303, 1, Dial(SIP/303, 20)
exten => 304, 1, Dial(SIP/304, 20)
esten => 305, 1, Dial(SIP/305, 20)
exten => _1NXXXXXXXXX,1,Dial(SIP/184XXXXXXX/${EXTEN})
exten => *97, 1, VoiceMailMain(${CALLERID(NUM)},s)
exten => *97, 2, Hangup
exten => NPANXXXXXX, 1, Dial(SIP/301, 20)
exten =>NPANXXXXXX, 2, VoiceMail(301,u)

bindport = 5060
bindaddr =
context = others
register =>
callerid=device <301>

  • CLI

The Asterisk command line interface can help you a lot when doing troubleshooting.

To see whats going on, start CLI with asterisk -r and enter core set verbose 3. SIP debugging can be enabled with sip set debug on but this kind of much to read, so you may pipe this to a text file instead: asterisk -vvvr > dump.log.

To show a specific dialplan context: dialplan show yourcontext When you change your dialplan you need to reload it with: dialplan reload. For changes in other files you need to restart your asterisk.

Check if you peers are responsive and online: sip show peers

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  • Last modified: 2015/01/20 01:07
  • by leangjia