Asterisk cơ bản

Posted on by Admin
Nội dung
Hide
Nội dung
×

Asterisk cơ bản

Trong bài viết này mình chia sẻ về asterisk cơ bản các bạn tham khảo. Hiện tại chan_sip đã cũ và trong một số version được thay thế bằng pjsip Trong Asterisk tất cả cấu hình sip, trunk, channel được định nghĩa /etc/asterisk/sip.conf. Dialplan được định nghĩa trong /etc/asterisk/extensions.conf. Tìm kiếm cấu hình khác của Asterisk trong /etc/asterisk/.  Voicemail và ghi âm lưu trong /var/spool/asterisk/. Asterisk logs /var/log/asterisk/.

Kết nối đến Asterisk daemon:

asterisk -rvvv

sip.conf reload từ Asterisk console

CLI> module reload chan_sip.so

/etc/asterisk/modules.conf file load module. Mặc định line ‘autoload=yes’

Dialplan

Các định nghĩa về dialplan:

  • Contexts:
    • syntax: a section starting with a name in square brackets, like “[local]”
    • act as namespaces/scopes
    • the [general] and [globals] contexts are special
    • The context defined for a channel (in sip.conf) defines where it enters the dialplan.
    • Since a context limits scope, it can be used to restrict features (e.g. allow long distance calling only in some contexts).
    • One context can be included in another: include => myContext
  • Extensions
    • syntax: an extension starts like “exten => ” exten => name,priority,application()
    • not just (sometimes not at all) a numeric ID tied to a phone
    • might triggered by an incoming call or digits dialed on a channel
    • really a script/function/series of steps, with each step containing an application
    • Each step in an extension has three parts:
      • name/number of the extension
      • priority (i.e. then number of this step in the extension)
      • the application/command executed by the step
  • Priorities
    • the order number of each step of an extension
    • numbered (and executed) sequentially
    • The special “n” priority takes the number of the previous priority and increments it. (But an extension must at least always have a step “1”, even if the rest are “n”.)
    • Priorities can be labeled, which is handy when using the “n” priority: exten => 123,n(myPriorityLabel),app()
  • Applications
    • Performs an action on the channel (e.g. play a sound, dial another channel, hangup, etc.)
    • Common applications: Answer(), Playback(), Hangup(), Goto(), Background(), WaitExten() Playback(/home/foo/sounds/ding) (Plays ‘ding.wav’.) Goto(myContext,myExtension,myPriority)

The “same =>” shorthand operator:

Ví dụ:

exten => 123,1,Answer()
exten => 123,n,doStuff()
exten => 123,n,doMore()
exten => 123,n,Hangup()

Hoặc ngắn gọn hơn:

exten => 123,1,Answer()
	same => n,doStuff()
	same => n,doMore()
	same => n,Hangup()

Biến asterisk:

PI=3.141592
Dial(${PI})

Biến mặc định asterisk :

${EXTEN}    Current extension
${EXTEN:1}  Extension omitting first numeral (subscript one to end of number)
${EXTEN:-4:4}  Start four digits from end, and return four digits (i.e. last 4 of number)

Pattern matching:

    X    matches 0-9
    Z    matches 1-9
    N    matches 2-9
    [372] matches three or seven or two
    .    matches one or more character
    !    matches zero or more characters
Match a ten-digit phone number:
    _NXXNXXXXXX
Match an international number (from North America):
    _011.

Asterisk CLI

CLI> module reload chan_sip.so
CLI> dialplan reload
CLI> sip show peers
CLI> sip show peer twilio
CLI> sip show users
CLI> core show channels
CLI> core show channels verbose
CLI> core show channel PJSIP/111-00000024
CLI> channel request hangup PJSIP/111-00000024
CLI> module show
CLI> module reload app_voicemail.so
CLI> voicemail show users
CLI> module reload features
CLI> core restart now
CLI> core restart gracefully
CLI> core show uptime
CLI> logger mute
CLI> logger set level NOTICE off
CLI> core show hints

Debugging:

CLI> core set verbose 4
CLI> core set debug 4
CLI> sip set debug on

Kiểm tra port asterisk đang chạy:

netstat -panu

Kiểm tra user đang online hay offline

asterisk -x 'pjsip show endpoints' | grep 130
watch "asterisk -x 'core show channels'"

Restarting:

  • core restart now restarts the Asterisk service immediately, ending any calls in progress.
  • core restart gracefully prevents new calls from starting up in Asterisk, but allows calls in progress to continue. When all the calls have finished, Asterisk restarts.
  • core restart when convenient waits until Asterisk has no calls in progress, then it restarts the service. It does not prevent new calls from entering the system.

File /etc/asterisk/sip.conf

; /etc/asterisk/sip.conf
; See /etc/asterisk/sip.conf.dist for helpful comments.

[general]
context=unauthenticated
allowguest=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp
srvlookup=no
language=en
nat=force_rport,comedia
localnet=10.0.0.0/255.255.255.0
externaddr=nn.nnn.nn.nnn

[codecs](!)
disallow=all
allow=g722,ulaw

[twilio](!)
type=peer
context=incoming
username=AsteriskUser
secret=xxxxxxxxxxxxxxxxxx
insecure=port,invite
dtmfmode=auto
host=foo.pstn.us1.twilio.com



    +000 000 000
  

(!,codecs)
type=friend
context=local
host=dynamic
dtmfmode=auto
rtp_symmetric=yes
rewrite_contact=yes
aggregate_mwi=yes
directmedia=nonat

[101](phone)
description=101 Polycom SoundPoint IP 335
secret=5Q0-#5th
mailbox=100@local

[102](phone)
description=102 Yealink T21 E2
secret=a137$a9tY
mailbox=100@local

[103](phone)
description=102 Yealink T21 E2
secret=a137$a9tY
mailbox=100@local

[103](phone)
description=102 Yealink T21 E2
secret=a137$a9tY
mailbox=100@local

File /etc/asterisk/extensions.config

; /etc/asterisk/extensions.config
; See /etc/asterisk/extenstions.conf.dist for helpful comments.

[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]
MY_AREA_CODE=555
MY_NUMBER=15555555555
RING_TIME=25

[emergency]
exten => 911,1,Verbose(1,Call initiated to 911!)
    same => n,Dial(SIP/911)

[local]

include => emergency

include => parkedcalls

exten => welcome,1,Answer()
    same => n,Playback(enter-ext-of-person)
    same => n,WaitExten()

exten => 101,1,Dial(SIP/101,,tT)

exten => 102,1,Dial(SIP/102,,tT)

exten => 103,1,Dial(SIP/103,,tT)

exten => 104,1,Dial(SIP/104,,tT)

; *97 is arbitrary. We can set the "voicemail" button key mapping on the phones.
exten => *97,1,VoiceMailMain(100@local,s)

exten => 199,1,Answer()
    same => n,Playback(hello-world)
    same => n,Hangup()

; Block toll numbers
exten => _900XXXXXXX,1,Hangup()
exten => _1900XXXXXXX,1,Hangup()
exten => _NXX976XXXX,1,Hangup()
exten => _1NXX976XXXX,1,Hangup()

; Block international
exten => _011.,1,Hangup()

exten => _1NXXNXXXXXX,1,Set(CALLERID(all)="+${MY_NUMBER}")
    same => n,Dial(SIP/+${EXTEN}@twilio)

exten => _NXXNXXXXXX,1,Set(CALLERID(all)="+${MY_NUMBER}")
    same => n,Dial(SIP/+1${EXTEN}@twilio)

exten => _NXXXXXX,1,Set(CALLERID(all)="+${MY_NUMBER}")
    same => n,Dial(SIP/+1${MY_AREA_CODE}${EXTEN}@twilio)

; Handle invalid entries:
exten => i,1,Playback(pbx-invalid)
    same => n,Goto(local,welcome,1)

; Handle time-outs
exten => t,1,Playback(vm-goodbye)
    same => n,Hangup()

[incoming]

exten => welcome,1,Answer()
    same => n,Background(/usr/share/asterisk/sounds/extra/enter-ext-of-person)
    same => n,WaitExten(8)
    same => n,Dial(allphones)

exten => _+X.,1,Goto(incoming,welcome,1)

exten => allphones,1,Dial(SIP/101&SIP/102&SIP/103&SIP/104,${RING_TIME},,tT)
    same => n,VoiceMail(100@local,u)

exten => 101,1,Dial(SIP/101,,tT)

exten => 102,1,Dial(SIP/102,,tT)

exten => 103,1,Dial(SIP/103,,tT)

exten => 104,1,Dial(SIP/104,,tT)

; Handle invalid entries
exten => i,1,Playback(pbx-invalid)
    same => n,Goto(incoming,welcome,1)

; Handle time-outs
exten => t,1,Playback(vm-goodbye)
    same => n,Hangup()

File /etc/asterisk/voicemail.conf

; /etc/asterisk/voicemail.conf
; See /etc/asterisk/voicemail.conf.dist for detailed comments

[general]

format=wav49|gsm|wav
serveremail=asterisk
attach=no
maxmsg=100
maxsecs=300
minsecs=1
maxgreet=120
skipms=3000
maxsilence=7
silencethreshold=128
maxlogins=2
charset=UTF-8
tz=eastern
locale=en_US.utf8
callback=fromvm

[zonemessages]

eastern=America/New_York|'vm-received' Q 'digits/at' IMp 
central=America/Chicago|'vm-received' Q 'digits/at' IMp 
central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM

[local]

100 => 9931,My Mailbox,[email protected]

## /etc/askterisk/features.conf ##

; /etc/askterisk/features.conf
; See /etc/askterisk/features.conf for helpful comments

[general]

courtesytone = beep
parkext => 700 
parkpos => 701-750
context => parkedcalls
parkingtime =>  90

PJSIP

PJSIP là channel driver thay thế cho chan_sip driver và ‘sip.conf’ thay bằng ‘pjsip.conf’.

Một vài cấu hình mẫu:

[201]
type=endpoint
context=default
disallow=all
allow=ulaw
transport=simpletrans
auth=auth201
aors=201

AoR Address of Record links a Contact to an Endpoint. AoR objects can also set associations with mailboxes. Example:

[201]
type=aor
max_contacts=1

Contact holds info about inbound registrations, and aliases SIP URI’s. May be associated with an individual SIP user agent. Although they can be created manually, a contact is automatically created upon registration to an AoR. Example of manual contact in an AoR section:

[202]
type=aor
contact=sip:[email protected]:5060

Registration sets up outbound registration with another system (e.g. a local peer or provider’s trunk). Like an endpoint, a registration can also link to transport or auth sections. Example:

[mytrunk]
type=registration
transport=simpletrans
outbound_auth=mytrunk
server_uri=sip:[email protected]:5060
client_uri=sip:[email protected]:5060
retry_interval=60

Transport transport layer options, like tcp, udp, websockets, tls. Mulitple endpoints can share the same transport. Example:

[simpletrans]
type=transport
protocol=udp
bind=0.0.0.0

Auth holds inbound or outbound authentication options and credentials. Example

[auth201]
type=auth
auth_type=userpass
password=201
username=201

Identify controls how we identify from which endpoint a packet originates (by IP or by user). If we don’t define an Identify section, user identity is based on the packet’s SIP “From” header. Example:

[201]
type=identify
endpoint=201
match=203.0.113.200

Dialplan mẫu:

exten => _2XX,1,Dial(PJSIP/${EXTEN})

Thay vì:

exten => _2XX,1,Dial(SIP/${EXTEN})

Pjsip CLI commands:

CLI> pjsip show endpoints

Thay vì:

CLI> sip show peers

Troubleshooting

CLI> core show channels
CLI> channel request hangup all

sudo watch "asterisk -x 'core show channels'"
sudo watch "asterisk -x 'core show channels verbose'"
sudo asterisk -rx 'core show channels concise'
sudo asterisk -rx 'core show channel PJSIP/xx_trunk-00000b8d'

RTCP may gather some call quality stats:

sudo asterisk -rx 'pjsip show channelstats'

Programming và Automation

CLI> channel originate PJSIP/126 application Playback hello-world

exten => 666,1,NoOp(-------- Voice Changer --------)
	same => n,Answer(1)
	same => n,Set(pranker=${CALLERID(all)})
	same => n,Set(PITCH_SHIFT(both)=lower)
	same => n,Read(prankee,enter-ext-of-person, 4)
	same => n,Log(NOTICE, ${pranker} pranked ${prankee} (voice change))
	same => n,Set(CALLERID(all)="April Furst" <667>)
	same => n,Dial(PJSIP/${prankee:0:3}, 20) 
	same => n,Voicemail(${prankee:0:3})
	same => n,Hangup()

exten => 664,1,NoOp(-------- Mysterious Numbers Station --------)
	same => n,Answer()
	same => n,Set(PITCH_SHIFT(tx)=lower)
	same => n,Set(i=0)
	same => n(while_start),While($[${i} < 64])
	same => n,Playback(silence/1)
	same => n,GotoIf($[${RAND(0,10)} < 8]?:letter)
	same => n,SayDigits(${RAND(0,9)})
	same => n,Goto(while_end)
	same => n(letter),SayPhonetic(${SHELL(echo ${RAND(0,6)} | tr 0-6 A-F)})
	same => n(while_end),Set(i=$[${i} + 1])
	same => n,EndWhile
	same => n,Playback(activated)
	same => n,Hangup()

One thing to note is that once we call the Dial() function, execution of the dialplan doesn’t continue until the function returns (i.e. one of the parties hangs up). If we want to do stuff during the call, use a macro with the “M()” option to Dial():

exten => 699,1,NoOp(-------- Test Dial() macros --------)
	same => n,Dial(PJSIP/+12485552222@my_trunk,20,M(mystery-call))

[macro-mystery-call]
exten => s,1,Wait(5)
	same => n,SayPhonetic(MYSTERY)
	same => n,Hangup()

If we’re dialing a local extension, do something like this instead:

exten => 699,1,NoOp(-------- Test Dial() macros --------)
	same => n,Dial(Local/123@my_context,20,M(mystery-call))

Debugging SIP

Bản tin SIP cơ bản:

A to B:  INVITE
B to A:  100 Trying
B to A:  180 Ringing
B to A:  180 Ringing
B to A:  200 OK
A to B:  ACK
...(RTP media packets flow but no SIP packets, even in a several-minute conversation)...
B to A:  BYE
A to B:  200 OK

Tool hỗ trợ:

tcpdump -i eth0 'host asterisk.example.com and port 5060' -vv -w 20170316-serverA-serverB-sip-port-5060.pcap

Call Parking

CLI> parking show default

List config

sudo asterisk -x 'config list' | grep 'core'
core                 /etc/asterisk/asterisk.conf
core                 /etc/asterisk/cdr.conf
core                 /etc/asterisk/cel.conf

Bài viết mới nhất

Bảo mật Nginx với Lets Encrypt trên Ubuntu 20.04 với Certbot
Sep 15, 2023
Cài đặt Nginx Reverse Proxy để triển khai nhiều ứng dụng sử dụng Docker
Aug 24, 2023
Build một ứng dụng web đơn giản với NodeJS
Aug 3, 2023

Error: Contact form not found.