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最後更新: 2024-11-07
目錄
- CLI Configure File
- ...
- Scripts
CLI Configure File
cli.conf
; automatically executed core set verbose 3 = yes core set debug 3 = yes
cli_aliases.conf
[general] template = friendly ;template = asterisk_1dot4 ; Asterisk 1.4 style syntax [friendly] hangup request=channel request hangup originate=channel originate help=core show help pri intense debug span=pri set debug intense span reload=module reload pjsip reload=module reload res_pjsip.so res_pjsip_authenticator_digest.so ........
Linux 權限
asterisk 是在 dialout 及 audio Group 內的 !!
dialout:x:20:asterisk audio:x:29:asterisk
Help
asterisk -h
server*CLI> help ???
e.g. help sip
啟動 asterisk daemon
asterisk -p -U <user>
- -p # attempt to run with realtime priority for increased performance
- -U <user> # Run as a user other than the caller
Default: /usr/sbin/asterisk -p -U asterisk
Command
check version
asterisk -V
Asterisk 1.8.13.1~dfsg1-3+deb7u3
Get Shell
- -r Connect to Asterisk on this machine
- -v Increase verbosity
asterisk -rv
server*CLI>
Run Command Directly
# -x cmd Execute command cmd (implies -r)
e.g.
asterisk -x "sip reload"
asterisk -x "sip show peers"
CLI
aliases
cli show aliases
Alias Command Real Command reload module reload help core show help hangup request channel request hangup pri intense debug span pri set debug intense span originate channel originate
Core
Voip*CLI> core show version
Asterisk 1.8.7.1
Voip*CLI> core show uptime
System uptime: 1 hour, 53 minutes, 19 seconds Last reload: 1 hour, 53 minutes, 19 seconds
Voip*CLI> core show channels
Channel Location State Application(Data) 0 active channels 0 active calls 5 calls processed
Voip*CLI> core show calls
0 active calls 5 calls processed
Voip*CLI> core show hints
-= Registered Asterisk Dial Plan Hints =-
203@default : SIP/203 State:Idle Watchers 0
202@default : SIP/202 State:Unavailable Watchers 0
201@default : SIP/201 State:Unavailable Watchers 0
302@default : SIP/302 State:Unavailable Watchers 0
301@default : SIP/301 State:Unavailable Watchers 0
224@default : SIP/224 State:Unavailable Watchers 0
221@default : SIP/221 State:Unavailable Watchers 0
220@default : SIP/220 State:Unavailable Watchers 0
223@default : SIP/223 State:Unavailable Watchers 0
222@default : SIP/222 State:Unavailable Watchers 0
----------------
- 10 hints registeredVoip*CLI> core show functions
Installed Custom Functions: ---------------------------- CALLERID TIMEOUT ................. 44 custom functions installed.
codecs
# show codecs
ls -l /usr/lib/asterisk/modules/ | grep codec_
OR
server*CLI> core show codecs
It does not indicate anything about your configuration.
INT BINARY HEX TYPE NAME DESCRIPTION
-----------------------------------------------------------------------------------
1 (1 << 0) (0x1) audio g723 (G.723.1)
2 (1 << 1) (0x2) audio gsm (GSM)
4 (1 << 2) (0x4) audio ulaw (G.711 u-law)
8 (1 << 3) (0x8) audio alaw (G.711 A-law)
16 (1 << 4) (0x10) audio g726aal2 (G.726 AAL2)
32 (1 << 5) (0x20) audio adpcm (ADPCM)
64 (1 << 6) (0x40) audio slin (16 bit Signed Linear PCM)
128 (1 << 7) (0x80) audio lpc10 (LPC10)
256 (1 << 8) (0x100) audio g729 (G.729A)
512 (1 << 9) (0x200) audio speex (SpeeX)
1024 (1 << 10) (0x400) audio ilbc (iLBC)
2048 (1 << 11) (0x800) audio g726 (G.726 RFC3551)
4096 (1 << 12) (0x1000) audio g722 (G722)
8192 (1 << 13) (0x2000) audio siren7 (ITU G.722.1 (Siren7, licensed from Polycom))
16384 (1 << 14) (0x4000) audio siren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))
32768 (1 << 15) (0x8000) audio slin16 (16 bit Signed Linear PCM (16kHz))
65536 (1 << 16) (0x10000) image jpeg (JPEG image)
131072 (1 << 17) (0x20000) image png (PNG image)
dialplan
Voip*CLI> dialplan show
Voip*CLI> dialplan reload
database
Voip*CLI> database show
/SIP/Registry/201 : 192.168.88.171:5060:60:201:sip:[email protected]:5060 /SIP/Registry/202 : 192.168.88.172:5060:60:202:sip:[email protected]:5060 /SIP/Registry/301 : 192.168.88.32:5060:60:301:sip:[email protected]:5060 /SIP/Registry/302 : 192.168.88.30:5060:60:302:sip:[email protected]:5060 4 results found.
SIP
# 重新載入 sip.conf setting
Voip*CLI> sip reload
# 查看什麼人連到 Server
Voip*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status 201/201 192.168.88.171 D N 5060 Unmonitored 202/202 192.168.88.172 D N 5060 Unmonitored 301/301 192.168.88.32 D N 5060 Unmonitored 302/302 192.168.88.30 D N 5060 Unmonitored 4 sip peers [Monitored: 0 online, 0 offline Unmonitored: 4 online, 0 offline]
Voip*CLI> sip show peer 202
* Name : 202 Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : context-user-202 Subscr.Cont. : <Not set> Language : AMA flags : Unknown Transfer mode: open .......................
Voip*CLI> sip show users
Username Secret Accountcode Def.Context ACL ForcerPort
202 ?????? context-user-20 No Yes
201 ?????? context-user-20 No Yes# 會顥示 clear text password !!
Voip*CLI> sip show user 202
* Name : 202 Secret : <Set> # 只會顯然 <Set> 或 <Not set> MD5Secret : <Not set> Context : context-user-202 Language : AMA flags : Unknown Transfer mode: open MaxCallBR : 384 kbps CallingPres : Presentation Allowed, Not Screened Call limit : 0 Callgroup : Pickupgroup : Callerid : "Tim" <> ACL : No Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Sess-Min-SE : 90 secs RTP Engine : asterisk Codec Order : (ulaw:20,alaw:20) Auto-Framing: No
sip show registry
Voip*CLI> sip notify <type> <peer>
# types are defined in sip_notify.conf
Voip*CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer 192.168.88.172 (None) 2f0d209b06d7b83 0x0 (nothing) No Rx: REGISTER <guest> 192.168.88.30 (None) 054980911c97b24 0x0 (nothing) No Rx: REGISTER <guest> 192.168.88.32 (None) 434f2f4403a83be 0x0 (nothing) No Rx: REGISTER <guest> 192.168.88.171 (None) 6c2cf3487b18337 0x0 (nothing) No Rx: REGISTER <guest>
Voip*CLI> sip show inuse
* Peer name In use Limit
Voip*CLI> sip show settings
Global Settings: ---------------- UDP Bindaddress: 0.0.0.0:5060 TCP SIP Bindaddress: 0.0.0.0:5060 TLS SIP Bindaddress: Disabled ....................................... Network QoS Settings: --------------------------- IP ToS SIP: CS0 IP ToS RTP audio: CS0 IP ToS RTP video: CS0 IP ToS RTP text: CS0 802.1p CoS SIP: 4 802.1p CoS RTP audio: 5 802.1p CoS RTP video: 6 802.1p CoS RTP text: 5 Jitterbuffer enabled: No Network Settings: --------------------------- SIP address remapping: Enabled using externaddr Externhost: <none> Externaddr: x.x.x.x:0 Externrefresh: 10 Localnet: 192.168.88.0/255.255.255.0 Global Signalling Settings: --------------------------- Codecs: 0x0 (nothing) Codec Order: none Relax DTMF: No RFC2833 Compensation: No Symmetric RTP: Yes Compact SIP headers: No ....................................... Default Settings: ----------------- Allowed transports: UDP Outbound transport: UDP Context: default Force rport: Yes DTMF: rfc2833 .......................................
# 重選部份 setting
sip reload
# Debug
sip set debug on
Cheatsheet
Voip*CLI> sip show settings
Voip*CLI> sip show inuse
Voip*CLI> sip show peers
Voip*CLI> sip show users
Scripts
status.sh
echo "######## T1"
asterisk -x 'dahdi show channels' | awk '{print $1,$4,$5}'
echo "######## SIP"
asterisk -x "sip show peers"
