• База знаний Астериск
• Файлы конфигурации Asterisk
• xgu.ru wiki Asterisk
• Учебник по Asterisk на asteriskguru.com
• Интересный блог сисадмина из СПб

root@server:~# apt install asterisk

root@server:~# cd /etc/asterisk/

[server:~] # pkg install asterisk13

[server:~] # service asterisk rcvar

[server:~] # cd /usr/local/etc/asterisk/

• Настройка SIP в Asterisk - sip.conf
• Asterisk config sip.conf
• Matching incoming calls to users and peers
• username и auth username - что-то изменилось?
• Модуль SIP канала Asterisk
• How to set the concurrent calls limit on SIP trunk in Asterisk?
• call limit

server# cat sip.conf

[general]
;udpbindaddr=0.0.0.0:5061
transport=udp
disallow=all
allow=alaw
;context=default
;allowguest=no          ;Allow unknown access
;alwaysauthreject=yes   ;Always auth rejects
;autocreatepeer=no      ;AutoCreate Peer
;localnet=192.168.X.0/255.255.255.0
;;;externip=172.16.1.X
;externaddr = 172.16.1.X
;externaddr = 172.16.1.X:6050
;defaultexpiry=60

[401]
secret=tpassword401
type=friend
host=dynamic

[402]
secret=tpassword402
type=friend
host=dynamic
;allowsubscribe=no
;setvar=MY-USER-ID=user2@corpX.un       ;in channel type=user 
;callerid=Petrov Petr Petrovitch<402>
;cid_number=402
;fullname=Petrov Petr Petrovitch
;disallow=all
;allow=gsm
;canreinvite=no ;1.4
;directmedia=no ;1.6
;;;nat=yes
;nat=force_rport,comedia
;qualify=yes
;call-limit=1
;deny=0.0.0.0/0.0.0.0
;permit=172.16.1.0/255.255.255.0

;[user1]
;secret=password1
;type=friend
;host=dynamic

;[corpY]
;type=peer
;context=default
;host=server.corpY.un

root@server# cat sip.conf

...
;[office](!)
;type=friend
;host=dynamic
;deny=0.0.0.0/0.0.0.0
;permit=192.168.X.0/255.255.255.0

;[403](office)
;secret=tpassword403

;[404](office)
;secret=tpassword404
...

• Migrating from chan_sip to res_pjsip
• Описание параметров настройки pjsip в Asterisk
• НОВЫЙ КАНАЛЬНЫЙ ДРАЙВЕР CHAN_PJSIP В ASTERISK 13
• Переменные канала: function 'PJSIP_ENDPOINT'
• Пример настройки подключения Asterisk PJSIP к Zadarma

# cat /etc/asterisk/sip.conf

[general]
udpbindaddr=0.0.0.0:5061

# cat /etc/asterisk/pjsip.conf

[udp-transport-1]
type=transport
protocol=udp
bind=0.0.0.0:5060
;local_net=192.168.X.0/255.255.255.0
;external_media_address=172.16.1.X
;external_signaling_address=172.16.1.X

Базовый вариант

# cat /etc/asterisk/pjsip.conf

...
[401]
type=endpoint
transport=udp-transport-1
context=default
disallow=all
allow=alaw
aors=401
auth=401
;callerid=Ivan I. Ivanov<401>
;set_var=MY-USER-ID=user1@corpX.un
;set_var=MY-PSTN-CALLER-ID=8495123456X
;rtp_symmetric=yes
;force_rport=yes
;rewrite_contact=yes
;direct_media=no

[401]
type=aor
max_contacts=1
remove_existing=yes

[401]
type=auth
auth_type=userpass
password=tpassword401
username=401
...

Wizard вариант

!!! Изменение отдельных параметров требует либо удаление/reload/добавление/reload всего описания точки подключения, либо перезапуска asterisk

# cat /etc/asterisk/pjsip_wizard.conf

[402]
type=wizard
accepts_auth=yes
accepts_registrations=yes
transport=udp-transport-1
inbound_auth/username=402
inbound_auth/password=tpassword402
endpoint/allow=alaw
endpoint/context=default
;endpoint/set_var=MY-USER-ID=user2@corp13.un
;endpoint/callerid=Petr P. Petrov<402>
aor/max_contacts=1
aor/remove_existing=yes

# cat /etc/asterisk/pjsip.conf

...
[voip1_00000X]
type=registration
transport=udp-transport-1
outbound_auth=voip1_00000X
server_uri=sip:voip1.un
client_uri=sip:00000X@voip1.un
;;;retry_interval=10
;;;expiration=10
contact_user=voip1_00000X

[voip1_00000X]
type=auth
auth_type=userpass
password=spasswordX
username=00000X

[voip1_00000X]
type=aor
contact=sip:voip1.un:5060
;qualify_frequency=60

[voip1_00000X]
type=endpoint
transport=udp-transport-1
context=default
disallow=all
allow=alaw
outbound_auth=voip1_00000X
aors=voip1_00000X
from_user=00000X
from_domain=voip1.un
;direct_media=no
;rtp_symmetric=yes
;;;force_rport=yes
;;;rewrite_contact=yes

[voip1_00000X]
type=identify
endpoint=voip1_00000X
match=voip1.un

• Using Variables in Asterisk Dialplans
• Команда диалплана Asterisk "Dial"
• Dialing PJSIP Channels
• The Verbose and NoOp Applications

server# cat extensions.conf

;[general]

;[globals]

[default]

exten => 301,1,Answer()
exten => 301,n,Playback(hello-world)
exten => 301,n,SayNumber(X)
exten => 301,n,Hangup()

exten => 311,1,Answer()
     same => n,Echo()

;exten => 321,1,Answer()
;     same => n,SayDigits(${EXTEN})
;     same => n,Playback(beep)
;     same => n,SayDigits(${CALLERID(num)})
;     same => n,Hangup()

;exten => 331,1,Answer()
;exten => 331,n,NoOp(${MY-USER-ID})
;exten => 331,n,Verbose(1,${MY-USER-ID})
;exten => 331,n,Hangup()

;exten => 401,1,Dial(SIP/401)

;exten => 402,1,Dial(SIP/402)

exten => _4XX,1,Dial(SIP/${EXTEN})
;exten => _4XX,1,Dial(PJSIP/${EXTEN})

server# asterisk -r

server*CLI> sip reload
server*CLI> sip show settings
server*CLI> sip show users
server*CLI> sip show user 401
server*CLI> sip show peers
server*CLI> sip show peer 401
server*CLI> sip unregister 401

server*CLI> pjsip reload
server*CLI> pjsip show endpoints

server*CLI> dialplan reload
server*CLI> dialplan show default

server*CLI> core reload
server*CLI> core restart gracefully

server*CLI> exit

server# asterisk -x 'sip show peers'

• http://www.voip-info.org/wiki/view/Asterisk+video

server# cat sip.conf

[general]
...
allow=h263
allow=h264
videosupport=yes
avpf=yes
...

server# asterisk -rvvv

server*CLI> core set verbose 1

server# cat logger.conf

...
;console => notice,warning,error
console => warning,error,dtmf
messages => notice,warning,verbose(10),error
...

# tail -f /var/log/asterisk/messages

• Asterisk: ngrep, sipgrep, sngrep, protocol diagram

server*CLI> sip set debug peer 401
server*CLI> sip set debug ip 172.16.1.Z
server*CLI> sip set debug off

• How to include contexts based on time and date
• Использование различных контекстов в зависимости от времени суток

server# cat extensions.conf

[default]
...
include=>notwork,18:00-9:00,*,*,*
include=>work,9:00-18:00,*,*,*

[work]
exten => voip1_00000X,1,Dial(SIP/403,10,t)
exten => voip1_00000X,n,Dial(SIP/401&SIP/402)

[notwork]
exten => voip1_00000X,1,Dial(SIP/voip1_00000X/89161234567)

• Настройка Asterisk для связи с VoIP провайдерами

server# cat sip.conf

...
[voip1_00000X]
type=peer
defaultuser=00000X
secret=spasswordX
host=voip1.un
fromuser=00000X
fromdomain=voip1.un

server# cat extensions.conf

...
exten => _8XXXXXXXXXX,1,Dial(SIP/voip1_00000X/${EXTEN})
;exten => _8XXXXXXXXXX,1,Dial(SIP/${EXTEN}@voip1_00000X)
;exten => _8XXXXXXXXXX,1,Dial(PJSIP/${EXTEN}@voip1_00000X)  ;PJSIP только так

server# cat sip.conf

[general]
...
;register => 00000X:spasswordX@voip1.un/voip1_00000X
...
[voip1_00000X]
type=peer
defaultuser=00000X
secret=spasswordX
host=voip1.un
fromuser=00000X
fromdomain=voip1.un
;context=incoming
insecure=invite
callbackextension=voip1_00000X

server# cat extensions.conf

[globals]

;MY-RING-GROUP => SIP/401&SIP/402&SIP/403&SIP/404

;MY-RING-GROUP-84951234567 => SIP/401&SIP/402
;MY-RING-GROUP-84957654321 => SIP/403&SIP/404

[default]
...

;[incoming]
exten => voip1_00000X,1,Dial(SIP/401&SIP/402&SIP/403&SIP/404)

;exten => voip1_00000X,1,Dial(${MY-RING-GROUP})

;exten => 8495XXXXXXX,1,Dial(${MY-RING-GROUP-${EXTEN}})

server*CLI> sip show registry

• ALSA Asterisk Console on RasPBX

phone2# cat extensions.conf

[default]

exten => 301,1,Answer()
exten => 301,n,Playback(hello-world)
exten => 301,n,Hangup()

debian# cat modules.conf

...
load => chan_alsa.so
noload => chan_oss.so
...

pasberry# cat alsa.conf

...
input_device=hw:1,0
output_device=hw:0,0
...

phone2*CLI> module show like alsa

phone2*CLI> channel originate Console/dsp extension 301
phone2*CLI> console answer
phone2*CLI> console hangup

phone2# cat sip.conf

[general]
transport=udp
disallow=all
allow=alaw

[sipproxy]
type=peer
defaultuser=402
secret=tpassword402
host=gate.corpX.un
insecure=invite
callbackextension=sipproxy

phone2# cat extensions.conf

...
exten => _4XX,1,Dial(SIP/sipproxy/${EXTEN})
exten => sipproxy,1,Dial(Console/dsp)

• Подключение Asterisk к VoIP провайдеру по протоколу SIP

root@isp.un:~# cat sip.conf

...
[000001]
type=user
;type=friend
secret=spassword1
host=dynamic
context=from-sip-trunk
...
[000006]
type=user
;type=friend
secret=spassword6
host=dynamic
context=from-sip-trunk

root@isp.un:~# cat /etc/asterisk/extensions.conf

...
[from-sip-trunk]
exten => _8XXXXXXXXXX,1,Dial(SIP/sipnet_00000X/${EXTEN})

server.corpX.un# cat iax.conf

[general]
;bindport=4569
;bindaddr=0.0.0.0
disallow=all
allow=alaw

[corpY]
type=user
;context=default
host=dynamic
secret=apasswordY
auth=md5

[corpY]
type=peer
host=server.corpY.un
username=corpX
secret=apasswordX
auth=md5

;[corpY]
;type=friend
;context=default
;host=server.corpY.un

• CallerID (Настройка CHAN_SIP каналов)

server.corpX.un# cat extensions.conf

[default]
...
exten => _00Y4XX,1,Set(CALLERID(num)=00X${CALLERID(num)})
exten => _00Y4XX,n,Dial(IAX2/corpY/${EXTEN:3})

;;пример универсального шаблона для звонков в офисы с 1-го по 9-й
;exten => _0XX4XX,1,Set(CALLERID(num)=013${CALLERID(num)})
;exten => _0XX4XX,n,Dial(IAX2/corp${EXTEN:2:1}/${EXTEN:3})

server*CLI> iax2 reload

server*CLI> dialplan reload

server*CLI> sip show channels

server*CLI> cdr show active

server*CLI> channel request hangup SIP/<TAB>

# curl -v -X GET "http://localhost:8088/ari/channels?api_key=asterisk:asterisk"

server# cat /etc/asterisk/cdr.conf

...
[csv]
usegmtime=no
...

exten => _4XX,1,Set(CDR(userfield)=${CHANNEL(recvip)})
exten => _4XX,n,Dial(SIP/${EXTEN},,Tt)

server# tail -f /var/log/asterisk/cdr-csv/Master.csv

http://www.voip-info.org/wiki/view/Asterisk+CDR+csv+handling2

• Сервис SYSLOG
• Планирование выполнения заданий в Linux
• Планирование выполнения заданий в FreeBSD

server# :> cdr.conf

server# cat cdr_syslog.conf

[general]
facility=local0

[cdr-simple]
template = "${CDR(clid)}","${CDR(src)}","${CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}"

server# asterisk -x 'core restart gracefully'

server# asterisk -x 'cdr show status'

...
  Mode:                       Simple
...
* Registered Backends
  -------------------
    cdr-syslog
...

server# tail -f /var/log/cisco.log

• Asterisk cdr mysql

# cat cdr.sql

CREATE DATABASE asterisk;
 
GRANT ALL ON asterisk.* TO asterisk@localhost IDENTIFIED BY 'asterisk';

USE asterisk;

CREATE TABLE `cdr` (
 `calldate` datetime NOT NULL default CURRENT_TIMESTAMP,
 `clid` varchar(80) NOT NULL default '',
 `src` varchar(80) NOT NULL default '',
 `dst` varchar(80) NOT NULL default '',
 `dcontext` varchar(80) NOT NULL default '', 
 `channel` varchar(80) NOT NULL default '',
 `dstchannel` varchar(80) NOT NULL default '',
 `lastapp` varchar(80) NOT NULL default '',
 `lastdata` varchar(80) NOT NULL default '',
 `duration` int(11) NOT NULL default '0',
 `billsec` int(11) NOT NULL default '0',
 `disposition` varchar(45) NOT NULL default '', 
 `amaflags` int(11) NOT NULL default '0',
 `accountcode` varchar(20) NOT NULL default '',
 `userfield` varchar(255) NOT NULL default '',
 `uniqueid` VARCHAR(32) NOT NULL default '',
 `linkedid` VARCHAR(32) NOT NULL default '',
 `sequence` VARCHAR(32) NOT NULL default '',
 `peeraccount` VARCHAR(32) NOT NULL default ''
 );

ALTER TABLE `cdr` ADD INDEX ( `calldate` );
ALTER TABLE `cdr` ADD INDEX ( `dst` );
ALTER TABLE `cdr` ADD INDEX ( `accountcode` );

# mysql < cdr.sql

• Asterisk ODBC CDR

• https://code.google.com/p/asterisk-cdr-viewer/wiki/ReadMe
• Web-интерфейс для просмотра и прослушивания записей звонков Asterisk
• Прекрасное приложение, для просмотра статистики и прослушивания записей
• Установка и запуск сервера Apache
• Модуль php
• Модуль php mysql

# cd /var/www/html

# wget http://val.bmstu.ru/unix/voip/asterisk-cdr-viewer-latest.tgz

# tar -xzvf asterisk-cdr-viewer-*.tgz

# cat /var/www/html/asterisk-cdr-viewer/include/config.inc.php

...
$db_user = 'asterisk';
$db_pass = 'asterisk';
$db_name = 'asterisk';
...
$system_auth_require = 0;
...

# cat extensions.conf

...
exten => _4XX,1,Macro(monitor)
exten => _4XX,n,Dial(SIP/${EXTEN})
...
[macro-monitor]
exten => s,1,Set(MONITOR_FILE=/var/spool/asterisk/monitor/${UNIQUEID})
exten => s,n,MixMonitor(${MONITOR_FILE}.wav,b)

# usermod -G asterisk www-data

# service apache2 restart

http://server.corpX.un/asterisk-cdr-viewer/

!!! Найти вариант, при котором для не требуется заполнение поля callbackextension

• sip.conf перенести пользователей в realtime mysql
• Что нам стоит “Asterisk realtime” построить
• Asterisk RealTime SIP
• SIP Realtime, MySQL table structure

# cat sipfriends.sql

CREATE TABLE IF NOT EXISTS `sipfriends` (
      `id` int(11) NOT NULL AUTO_INCREMENT,
      `name` varchar(10) NOT NULL,
      `ipaddr` varchar(15) DEFAULT NULL,
      `port` int(5) DEFAULT NULL,
      `regseconds` int(11) DEFAULT NULL,
      `defaultuser` varchar(10) DEFAULT NULL,
      `fullcontact` varchar(35) DEFAULT NULL,
      `regserver` varchar(20) DEFAULT NULL,
      `useragent` varchar(20) DEFAULT NULL,
      `lastms` int(11) DEFAULT NULL,
      `host` varchar(40) DEFAULT NULL,
      `type` enum('friend','user','peer') DEFAULT NULL,
      `context` varchar(40) DEFAULT NULL,
      `permit` varchar(40) DEFAULT NULL,
      `deny` varchar(40) DEFAULT NULL,
      `secret` varchar(40) DEFAULT NULL,
      `md5secret` varchar(40) DEFAULT NULL,
      `remotesecret` varchar(40) DEFAULT NULL,
      `transport` enum('udp','tcp','udp,tcp','tcp,udp') DEFAULT NULL,
      `dtmfmode` enum('rfc2833','info','shortinfo','inband','auto') DEFAULT NULL,
      `directmedia` enum('yes','no','nonat','update') DEFAULT NULL,
      `nat` enum('yes','no','never','route') DEFAULT NULL,
      `callgroup` varchar(40) DEFAULT NULL,
      `pickupgroup` varchar(40) DEFAULT NULL,
      `language` varchar(40) DEFAULT NULL,
      `allow` varchar(40) DEFAULT NULL,
      `disallow` varchar(40) DEFAULT NULL,
      `insecure` varchar(40) DEFAULT NULL,
      `trustrpid` enum('yes','no') DEFAULT NULL,
      `progressinband` enum('yes','no','never') DEFAULT NULL,
      `promiscredir` enum('yes','no') DEFAULT NULL,
      `useclientcode` enum('yes','no') DEFAULT NULL,
      `accountcode` varchar(40) DEFAULT NULL,
      `setvar` varchar(40) DEFAULT NULL,
      `callerid` varchar(40) DEFAULT NULL,
      `amaflags` varchar(40) DEFAULT NULL,
      `callcounter` enum('yes','no') DEFAULT NULL,
      `busylevel` int(11) DEFAULT NULL,
      `allowoverlap` enum('yes','no') DEFAULT NULL,
      `allowsubscribe` enum('yes','no') DEFAULT NULL,
      `videosupport` enum('yes','no') DEFAULT NULL,
      `maxcallbitrate` int(11) DEFAULT NULL,
      `rfc2833compensate` enum('yes','no') DEFAULT NULL,
      `mailbox` varchar(40) DEFAULT NULL,
      `session-timers` enum('accept','refuse','originate') DEFAULT NULL,
      `session-expires` int(11) DEFAULT NULL,
      `session-minse` int(11) DEFAULT NULL,
      `session-refresher` enum('uac','uas') DEFAULT NULL,
      `t38pt_usertpsource` varchar(40) DEFAULT NULL,
      `regexten` varchar(40) DEFAULT NULL,
      `fromdomain` varchar(40) DEFAULT NULL,
      `fromuser` varchar(40) DEFAULT NULL,
      `qualify` varchar(40) DEFAULT NULL,
      `defaultip` varchar(40) DEFAULT NULL,
      `rtptimeout` int(11) DEFAULT NULL,
      `rtpholdtimeout` int(11) DEFAULT NULL,
      `sendrpid` enum('yes','no') DEFAULT NULL,
      `outboundproxy` varchar(40) DEFAULT NULL,
      `callbackextension` varchar(40) DEFAULT NULL,
      `registertrying` enum('yes','no') DEFAULT NULL,
      `timert1` int(11) DEFAULT NULL,
      `timerb` int(11) DEFAULT NULL,
      `qualifyfreq` int(11) DEFAULT NULL,
      `constantssrc` enum('yes','no') DEFAULT NULL,
      `contactpermit` varchar(40) DEFAULT NULL,
      `contactdeny` varchar(40) DEFAULT NULL,
      `usereqphone` enum('yes','no') DEFAULT NULL,
      `textsupport` enum('yes','no') DEFAULT NULL,
      `faxdetect` enum('yes','no') DEFAULT NULL,
      `buggymwi` enum('yes','no') DEFAULT NULL,
      `auth` varchar(40) DEFAULT NULL,
      `fullname` varchar(40) DEFAULT NULL,
      `trunkname` varchar(40) DEFAULT NULL,
      `cid_number` varchar(40) DEFAULT NULL,
      `callingpres` enum('allowed_not_screened','allowed_passed_screen','allowed_failed_screen','allowed','prohib_not_screened','prohib_passed_screen','prohib_failed_screen','prohib') DEFAULT NULL,
      `mohinterpret` varchar(40) DEFAULT NULL,
      `mohsuggest` varchar(40) DEFAULT NULL,
      `parkinglot` varchar(40) DEFAULT NULL,
      `hasvoicemail` enum('yes','no') DEFAULT NULL,
      `subscribemwi` enum('yes','no') DEFAULT NULL,
      `vmexten` varchar(40) DEFAULT NULL,
      `autoframing` enum('yes','no') DEFAULT NULL,
      `rtpkeepalive` int(11) DEFAULT NULL,
      `call-limit` int(11) DEFAULT NULL,
      `g726nonstandard` enum('yes','no') DEFAULT NULL,
      `ignoresdpversion` enum('yes','no') DEFAULT NULL,
      `allowtransfer` enum('yes','no') DEFAULT NULL,
      `dynamic` enum('yes','no') DEFAULT NULL,
      PRIMARY KEY (`id`),
      UNIQUE KEY `name` (`name`),
      KEY `ipaddr` (`ipaddr`,`port`),
      KEY `host` (`host`,`port`)
) ENGINE=MyISAM;

# mysql asterisk < sipfriends.sql

# mysql asterisk

MariaDB [asterisk]> delete from sipfriends;

insert into sipfriends(name, secret, host, type) values ('403', 'tpassword403', 'dynamic', 'friend');
insert into sipfriends(name, secret, host, type) values ('404', 'tpassword404', 'dynamic', 'friend');

• Asterisk ODBC realtime

# cat sip.conf

[general]
...
rtcachefriends=yes
rtcache=yes
...

# cat extensions.conf

...
exten => _5XX,1,Answer()
exten => _5XX,n,Set(row="${REALTIME(sippeers,name,4${EXTEN:1})}")
exten => _5XX,n,NoOp(ROW is ${row})
exten => _5XX,n,Set(col_name_pair=${CUT(row,",",1)})
exten => _5XX,n,NoOp(${col_name_pair})
exten => _5XX,n,Hangup()
...

debian# apt install asterisk-mysql

debian# service asterisk restart

# asterisk -x 'module show' | grep mysql

# cat cdr_mysql.conf

[global]
hostname=localhost
user=asterisk
password=asterisk
dbname=asterisk
table=cdr

[columns]
alias start => calldate

# service asterisk restart

server*CLI> cdr show status

...
* Registered Backends
  -------------------
    mysql
...

server*CLI> cdr mysql status

...
Wrote NNN records ...

# cat /etc/asterisk/res_config_mysql.conf

[general]
dbhost = 127.0.0.1
dbname = asterisk
dbuser = asterisk
dbpass = asterisk

# cat /etc/asterisk/extconfig.conf

[settings]
sippeers => mysql,general,sipfriends

server*CLI> core reload

server*CLI> realtime mysql status
general connected to asterisk@127.0.0.1, port 3306 with username asterisk for 12 seconds.

# apt install asterisk-mysql

# cat extensions.conf

...
exten => 312,1,Answer()
     same => n,MYSQL(Connect connid localhost asterisk asterisk asterisk)
;     same => n,MYSQL(Query resultid ${connid} SET NAMES 'utf8')
     same => n,MYSQL(Query resultid ${connid} SELECT name, fullname FROM sipfriends)
     same => n(loop),MYSQL(Fetch fetchid ${resultid} NM FN)
     same => n,GotoIf($[${fetchid} = 0]?clear)
     same => n,NoOp(fetchid ${fetchid} ${FN} ${NM})
     same => n,SayDigits(${NM})
;     same => n,Festival(${FN})
     same => n,Goto(312,loop)
     same => n(clear),MYSQL(Clear ${resultid})
     same => n,MYSQL(Disconnect ${connid})
     same => n,Hangup()
...
exten => _6XX,1,MYSQL(Connect connid localhost asterisk asterisk asterisk)
;     same => n,MYSQL(Query resultid ${connid} SET NAMES 'utf8')
     same => n,MYSQL(Query resultid ${connid} SELECT fullname FROM sipfriends where name="4${EXTEN:1}")
     same => n,MYSQL(Fetch fetchid ${resultid} FN)
     same => n,NoOp(fullname ${FN})
;     same => n,Festival(${FN})
     same => n,MYSQL(Clear ${resultid})
     same => n,MYSQL(Disconnect ${connid})
     same => n,Hangup()
...

• Сервис ODBC

• !!! Не совместим с пакетом asterisk-mysql

# cat res_odbc.conf

[asterisk]
enabled => yes
dsn => asterisk_dsn
pre-connect => yes

server*CLI> core reload

server*CLI> odbc show all

ODBC DSN Settings
-----------------

  Name:   asterisk
  DSN:    asterisk_dsn
    Last connection attempt: 1970-01-01 03:00:00
    Number of active connections: 1 (out of 1)

# cat cdr_adaptive_odbc.conf

[default]
connection=asterisk
table=cdr

# service asterisk restart

server*CLI> cdr show status

# cat modules.conf

...
;noload => res_config_odbc.so
...

# cat extconfig.conf

[settings]
sippeers => odbc,asterisk,sipfriends

# cat func_odbc.conf

...
[SIPFRIENDS_BY_NAME]
dsn=asterisk
readsql=SELECT fullname FROM sipfriends where name='${ARG1}'

# cat extensions.conf

...
exten => _6XX,1,Set(FN=${ODBC_SIPFRIENDS_BY_NAME(4${EXTEN:1})})
        same => n,NoOp(fullname ${FN})
;     same => n,Festival(${FN})
...

• Сервис FreeRADIUS

# cat /usr/share/auth.sh

#!/bin/sh

CALL_FROM=$1

# chown asterisk /tmp/auth.log

if echo "User-Name=${CALL_FROM},User-Password=${CALL_FROM},NAS-IP-Address=127.0.0.1" | radclient localhost auth testing123 >>/tmp/auth.log
then
        echo -n YES
else
        echo -n NO
fi

exit 0

# cat /usr/share/acct-start.sh

#!/bin/sh

CALL_ID=$1
CALL_FROM=$2
CALL_TO=$3

echo "User-Name=${CALL_FROM},Acct-Session-Id=${CALL_ID},Called-Station-Id=${CALL_TO},Calling-Station-Id=${CALL_FROM},Acct-Status-Type=Start,NAS-IP-Address=127.0.0.1,NAS-Port=${CALL_FROM}${CALL_TO}"| radclient localhost acct testing123

exit 0

# cat /usr/share/acct-stop.sh

#!/bin/sh

CALL_ID=$1
CALL_FROM=$2
CALL_TO=$3

echo "User-Name=${CALL_FROM},Acct-Session-Id=${CALL_ID},Acct-Status-Type=Stop,NAS-IP-Address=127.0.0.1,NAS-Port=${CALL_FROM}${CALL_TO}"| radclient localhost acct testing123

exit 0

Примечания:

• Помещаем абонента, звонки которого надо тарифицировать, в контекст billing
• Переводим все тарифицируемые вызовы в контекст aaa-call (что-бы отследить событие окончания разговора через контекст «h»)
• В случае MYAUTH!=NO разрешаем абоненту совершить звонок. С помощью макроса «U» программируем вызов кода в момент снятия трубки для начала тарификации. По какой то причине переменные ${CALLERID(num)} ${CALLERID(dnid)} не передаются, а ${UNIQUEID} меняет значение в этом коде, поэтому занчения передаются через аргументы.
• В контексте «h» проверяем факт наличия разговора и только в этом случае тарифицируем его окончание

# cat extensions.conf

...
[billing]
...
exten => _4XX,1,Goto(aaa-call,s,1)
...
[aaa-call]
exten => s,1,Verbose(AUTH CALL_ID=${UNIQUEID} CALL_FROM=${CALLERID(num)} CALL_TO=${CALLERID(dnid)})
exten => s,n,Set(MYAUTH=${SHELL(/usr/share/auth.sh ${CALLERID(num)})});
exten => s,n,GotoIf($["${MYAUTH}" = "NO"]?end_call)
exten => s,n,Dial(SIP/${CALLERID(dnid)},,U(aaa-call-beg-acct,${UNIQUEID},${CALLERID(num)},${CALLERID(dnid)}));
exten => s,n(end_call),Hangup()

exten => h,1,GotoIf($["${DIALSTATUS}" != "ANSWER"]?no-answer)
exten => h,n,Verbose(END ACCT DIALSTATUS=${DIALSTATUS} CALL_ID=${UNIQUEID} CALL_FROM=${CALLERID(num)} CALL_TO=${CALLERID(dnid)})
exten => h,n,System(/usr/share/acct-stop.sh ${UNIQUEID} ${CALLERID(num)} ${CALLERID(dnid)})
exten => h,n(no-answer),NoOp()

[aaa-call-beg-acct]
exten => s,1,Verbose(BEGIN ACCT CALL_ID=${ARG1} CALL_FROM=${ARG2} CALL_TO=${ARG3})
exten => s,n,System(/usr/share/acct-start.sh ${ARG1} ${ARG2} ${ARG3})
exten => s,n,return
...

Настройка прохождения трафика и сигнализации через asterisk и режима тонального набора для всех каналов

server# cat sip.conf

[general]
...
directmedia=no ;1.6
canreinvite=no ;1.4
dtmfmode=rfc2833
...

• С сопровождение и слепой
• Клавиша «flash» аналогового телефона
• Кнопка «transfer» ip телефона
• Программируемая комбинация DTMF

server# cat features.conf

[general]
...
featuredigittimeout = 1000
...
[featuremap]
...
blindxfer => *21
atxfer => *22
...

server# cat extensions.conf

...
exten => _4XX,1,Dial(SIP/${EXTEN},,Tt)
...
exten => voip1_00000X,1,Dial(SIP/403,10,t)
exten => voip1_00000X,n,Dial(SIP/401&SIP/402&SIP/403,,t)

server*CLI> core reload

server*CLI> features show

• My IT-blog hints(Дополнено)
• Стандартные расширения
• Configuring chan_sip for Presence Subscriptions

# cat sip.conf

[general]
...
allowsubscribe=yes
notifyhold=yes
callcounter=yes
notifyringing=yes
limitonpeers=yes
subscribecontext=default
...

# cat extensions.conf

[default]
...
exten => _4XX,hint,SIP/${EXTEN}
...

• DrayTek SIP Softphone
• SIP Phone Panasonic KX-HDVXXX

server*CLI> sip show subscriptions

server*CLI> core show hints

server# cat sip.conf

...
[401]
callgroup=1
pickupgroup=1
...
[403]
callgroup=1
pickupgroup=1

server# cat features.conf

...
[general]
...
pickupexten = *88
...

server*CLI> features show

Сценарий: снимаем трубку чужого, звонящего телефона, просим подождать, набираем *21700, слушаем номер слота, кладем трубку. Со своего телефона набираем номер слота и продолжаем разговор.

asterisk11# less features.conf

asterisk13# less res_parking.conf

...
[default]
parkext => 700
parkpos => 701-720
...
context => parkedcalls
...

server# cat extensions.conf

[default]

include => parkedcalls
...

*CLI> core show codecs

*CLI> core show translation

root@server:~# apt install asterisk-core-sounds-ru-wav

root@server:~# find /usr/share/asterisk/sounds/ru/

server# cd /usr/share/asterisk/sounds/ru/

server# mkdir ru; cd ru/

server# wget http://downloads.asterisk.org/pub/telephony/sounds/asterisk-core-sounds-ru-alaw-current.tar.gz

server# tar -xvf asterisk-core-sounds-ru-alaw-current.tar.gz

server# cat asterisk.conf

...
defaultlanguage = ru
...

# service asterisk restart

server# cat sip.conf

[general]
language=ru
...
[401]
language=ru
...

server# cat extensions.conf

...
exten => 301,1,Answer()
exten => 301,n,Playback(hello-world)
exten => 301,n,Set(CHANNEL(language)=en)
exten => 301,n,Playback(hello-world)
exten => 301,n,Hangup()
...

server# cat voicemail.conf

[general]
...
;debian# apt install asterisk-voicemail-imapstorage
;debian# service asterisk restart
;imapserver=mail.isp.un
;imapfolder=INBOX
;charset=UTF-8
...
[corpX]
;401 => ,,
;402 => 2222,,
;403 => ,Ваше Имя Отчество Фамилия,userX@isp.un,,attache=yes|delete=yes
;404 => 4444,Ваше Имя Отчество Фамилия,,,imapuser=userX|imappassword=passwordX

server*CLI> voicemail reload

server# cat extensions.conf

[default]
...
exten => 390,1,VoiceMailMain(@corpX)
...
exten => _4XX,1,Dial(SIP/${EXTEN},10,Tt)
exten => _4XX,n,VoiceMail(${EXTEN}@corpX)
exten => _4XX,n,Hangup()

...

Альтернативный dialplan для прослушивания голосовой почты (звонок на номер своего телефона)

exten => _4XX,1,GotoIf($["${CALLERID(num)}" = "${EXTEN}"]?MAIL)
exten => _4XX,n,Macro(monitor)
exten => _4XX,n,Dial(SIP/${EXTEN},10)
exten => _4XX,n,VoiceMail(${EXTEN}@corpX)
exten => _4XX,n,Hangup()

exten => _4XX,n(MAIL),VoiceMailMain(${EXTEN}@corpX)
exten => _4XX,n,Hangup()

server# cat sip.conf

...
[401]
;allowsubscribe=no
mailbox=401@corpX
...
[402]
mailbox=402@corpX
...
[403]
mailbox=403@corpX
...
[404]
mailbox=404@corpX
...

server# ls /var/spool/asterisk/voicemail/corpX/

server# ls -l /var/mail/

• Приложение ConfBridge

• Сценарий 1: все желающие участвовать в конференции звонят на заранее обговоренный номер 6NN
• Сценарий 2: ответственный за проведение конференции обзванивает участников и переводит вызовы в конференцию набирая *216NN
• Сценарий 3: ответственный звонит на номер 389, у участников 401 и 403 звонят телефоны и ответственный тоже становится участником конференции.

server# cat extensions.conf

[default]
...
exten => 389/402,1,Originate(SIP/401,exten,default,601,1)
exten => 389/402,n,Originate(SIP/403,exten,default,601,1)
exten => 389/402,n,ConfBridge(601)

exten => _6XX,1,ConfBridge(${EXTEN})
...

server# asterisk -x "confbridge list 1"
...

• Проблемы с CallerID

# asterisk -x "channel originate SIP/402 extension 399@default"

; сначала вызов телефона оператора, затем вызов абонента
server*CLI> channel originate SIP/402 extension 89161234567@default

; сначала вызов абонента, затем телефона оператора 
server*CLI> channel originate SIP/voip1_00000X/89161234567 extension 402@default

; сначала вызов обработчика входящих вызовов (секретарь, IVR, Queue), затем вызов абонента
server*CLI> channel originate Local/voip1_00000X@default extension 89161234567@default

• Сервис atrun

server# cat /usr/share/403.call

Channel: sip/403
Context: default
Extension: 399
CallerId: Conferencia <399>

server# cat /usr/share/401.call

Channel: sip/401
Context: default
Extension: 399
CallerId: Conferencia <399>

server# cat /usr/share/start_conference.sh

#!/bin/sh

cp /usr/share/4* /tmp
chown asterisk /tmp/4*
mv /tmp/4* /var/spool/asterisk/outgoing/

server# chmod +x /usr/share/start_conference.sh

server# /usr/share/start_conference.sh

• Asterisk AMI

• Asterisk ARI

# curl -v -X POST "http://localhost:8088/ari/channels?endpoint=SIP%2F401&extension=301&context=default&callerId=301&api_key=asterisk:asterisk"

или

# curl -v -u asterisk:asterisk -X POST "http://localhost:8088/ari/channels?endpoint=SIP%2F401&extension=301&context=default&callerId=301"

• Asterisk standard extensions

!!! После записи IVR нажмите #

sever# cat extensions.conf

[default]
...
exten => 302,1,Authenticate(1234)
exten => 302,n,Playback(beep)
exten => 302,n,Record(recordings/greeting:alaw)
exten => 302,n,Hangup()

exten => 303,1,Answer()
exten => 303,n,Playback(beep)
exten => 303,n,Playback(recordings/greeting)
exten => 303,n,Playback(beep)
exten => 303,n,Hangup()

exten => 304,1,Goto(menu,s,1) ; for debug
...
;[incoming]
exten => voip1_00000X,1,Goto(menu,s,1)
...
[menu]
exten => s,1,Background(recordings/greeting)
exten => s,2,Goto(s,1)
 
; exten => 1,1,Dial(SIP/401)
; exten => 2,1,Dial(SIP/402)
; exten => 3,1,Dial(SIP/403)
; exten => 4,1,Dial(SIP/404)
; exten => _X,1,Dial(SIP/40${EXTEN})

exten => _X,1,Goto(default,40${EXTEN},1)

;exten => 0,1,Goto(menu2,s,1)
;exten => 0,1,DISA(no-password,default)

;exten => _[34]XX,1,Goto(default,${EXTEN},1)
;exten => _[43]XX,1,Dial(Local/{EXTEN}@default)
 
;exten => t,1,Goto(s,1)  ; timeout exceeded
;exten => i,1,Goto(s,1)  ; error extensions

;[menu2]
;exten => s,1,WaitExten(8)
;exten => _XXX,1,Goto(default,${EXTEN},1)

http://voip.rus.net/tiki-index.php?page=Asterisk+AEL

sever# cat extensions.conf

[default]
...
include => ivr
...
;[incoming]
exten => voip1_00000X,1,Goto(menu,s,1)

sever# cat extensions.ael

context ivr {
  302 => {
    Answer();
    Record(recordings/greeting:alaw);
    Hangup();
  }
  303 => {
    Answer();
    Playback(beep);
    Playback(recordings/greeting);
    Playback(beep);
    Hangup();
  }
  304 => {
    goto menu|s|begin;
  }
}

context menu {
  s => {
    begin:
      Background(recordings/greeting);
      goto begin;
  }
  _X => {
     goto default|40${EXTEN}|1;
  }
}

server# cat extensions.conf

[default]
...
exten => _5XX/402,1,ChanSpy(SIP/4${EXTEN:1})
...

• Создается два файла - in и out (можно смикшировать в один, используя утилиту sox)
• Имена файлов определяется шаблоном auto-${EPOCH}-${TOUCH_MONITOR}

server# cat features.conf

...
[featuremap]
...
automon => *11
...

server# cat extensions.conf

...
exten => _4XX,1,Set(TOUCH_MONITOR=${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-${CALLERID(number)}-${EXTEN})
exten => _4XX,n,Dial(SIP/${EXTEN},,Ww)
...

server*CLI> core reload

server# ls /var/spool/asterisk/monitor/

server# cat extensions.conf

...
[menu]
...
exten => _X,1,Set(MY-FILE-NAME=${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-${CALLERID(number)}-40${EXTEN}.wav)
exten => _X,n,MixMonitor(${MY-FILE-NAME},b)
exten => _X,n,Goto(default,40${EXTEN},1)
...

server*CLI> dialplan reload

server# ls /var/spool/asterisk/monitor/

• The Open Source Enterprise Telephony Recording and Retrieval System

• Convert WAV audio files for use in Asterisk
• Перекодировка mp3 в raw формат для asterisk (Перекодировка звука)

[server:~] # rm /usr/local/share/asterisk/moh/*
[server:~] # cp Knockin_on_Heavens_Door.raw /usr/local/share/asterisk/moh/

root@server:~# rm /usr/share/asterisk/moh/*
root@server:~# cp Knockin_on_Heavens_Door.raw /usr/share/asterisk/moh/

server*CLI> moh reload

server*CLI> moh show files

server# cat extensions.conf

...
exten => 305,1,Answer()
exten => 305,n,MusicOnHold()
...

server# mkdir /usr/share/asterisk/busy

server# cp -v /usr/share/asterisk/sounds/ru/tt-allbusy.* /usr/share/asterisk/busy/

server# cat musiconhold.conf

...
[busy]
mode=files
directory=busy

server# cat extensions.conf

...
exten => ABC,1,Answer()
exten => ABC,n,Dial(SIP/ABC,,m(busy))
...

• Сервис Festival

# asterisk -x 'module show' | grep fest

server# cat extensions.conf

...
exten => 303,1,Answer()
exten => 303,n,Festival('Привет Мир')
exten => 303,n,Hangup()

exten => 304,1,Answer()
exten => 304,n,Set(foo=${FILE(/tmp/text.txt)})
exten => 304,n,Festival(${foo})
exten => 304,n,Hangup()

exten => 305,1,Answer()
;linux   exten => 305,n,Set(foo=${SHELL(uptime | sed -E 's/.* (.*), .*, .*$/\1/' | sed 's/,/ запятая /g')})
exten => 305,n,Festival(${foo})
exten => 305,n,Hangup()
...

server# cat followme.conf

[101+X]
;number=>401,30
;number=>89162323232,30
context=>default
[401]
number=>89162323232,30
;number=>89163434345,30
context=>default

server# cat extensions.conf

;...
exten => _4XX,1,Dial(SIP/${EXTEN},10,t)
exten => _4XX,n,FollowMe(${EXTEN})

• Оптимизируем листинг диалплана Asterisk при помощи MySQL

• Перенаправление звонков

server*CLI> database show

server*CLI> database put num2login 401 user1

server*CLI> database put num2login 402 user2

server*CLI> database put num2login 403 user3

server*CLI> database num2login

# cat extensions.conf

exten => _[a-z].,1,Dial(SIP/${EXTEN})

exten => _4XX,1,Set(MY-LOGIN=${DB(num2login/${EXTEN})})
exten => _4XX,n,NoOp(variable MY-LOGIN set to ${MY-LOGIN})
exten => _4XX,n,Verbose(1,variable MY-LOGIN set to ${MY-LOGIN})
exten => _4XX,n,Dial(SIP/${MY-LOGIN})

server*CLI> database put redirect 403 89166071103

server*CLI> database get redirect 403

server*CLI> database show redirect

server*CLI> database del redirect 403

server# cat extensions.conf

...
[default]
...
exten => 308,1,Answer()
exten => 308,n,Goto(setredirect,s,1)

exten => 309,1,Answer()
exten => 309,n,Playback(beep)
exten => 309,n,NoOp(${DB_DELETE(redirect/${CALLERID(num)})})
exten => 309,n,Hangup()
...
exten => _4XX,1,Dial(SIP/${EXTEN},10,t)

exten => _4XX,n,Set(REDIRNUM=${DB(redirect/${EXTEN})})
exten => _4XX,n,GotoIf($[${LEN(${REDIRNUM})} > 0]?CALLMOBILE)
...
exten => _4XX,n(CALLMOBILE),Goto(default,${REDIRNUM},1)
...
[setredirect]
exten => s,1,Playback(beep)
exten => s,n,WaitExten()
exten => _8XXXXXXXXXX,1,Set(DB(redirect/${CALLERID(num)})=${EXTEN})
exten => _8XXXXXXXXXX,n,Playback(beep)
exten => _8XXXXXXXXXX,n,SayDigits(${EXTEN})
exten => _8XXXXXXXXXX,n,Playback(beep)
exten => _8XXXXXXXXXX,n,Hangup()

• Функция SIPPEER()
• curcalls работает только если для канала установлено ограничение на число одновременных вызовов (call-limit)

server*CLI> core show function SIPPEER

exten => _4XX,1,Set(MY-CALLS-COUNT=${SIPPEER(${EXTEN},curcalls)})
exten => _4XX,n,Verbose(2,*** MY-CALLS-COUNT = ${MY-CALLS-COUNT} ***)
exten => _4XX,n,ExecIf($[ ${MY-CALLS-COUNT} > 0 ]?Set(MY-CLASS-MOH=busy))
exten => _4XX,n,Dial(SIP/${EXTEN},,tTm(${MY-CLASS-MOH}))

• Asterisk call-limit
• Asterisk в примерах: балансировка каналов

exten => _4XX,1,Set(GROUP(MYCALLTO)=${EXTEN})
exten => _4XX,n,Set(MYCALLS=${GROUP_COUNT(${GROUP(MYCALLTO)})})
exten => _4XX,n,Verbose(2,*** ${MYCALLS} ***)
exten => _4XX,n,ExecIf($[ ${MYCALLS} > 1 ]?Set(MYCLASSMOH=busy))
exten => _4XX,n,Verbose(2,(*** ${MYCLASSMOH} ***)
exten => _4XX,n,Dial(SIP/${EXTEN},10,Ttm(${MYCLASSMOH}))

• Переменная канала в Asterisk: DIALSTATUS

[default]

exten => _X!,1,Goto(pbx-invalid,s,1)
...
exten => _4XX,1,Dial(SIP/${EXTEN})
exten => _4XX,n,Verbose(1,${DIALSTATUS})
exten => _4XX,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?pbx-invalid,s,1)
...
[pbx-invalid]
exten => s,1,Playback(pbx-invalid)

Абдирахимов Мансур представил решение «Vip звонок» - определенный номер при звонке на офисный телефон соединялся с определенным номером даже если последний разговаривает.

[office]

...
exten => s,n,Gotoif($["${CALLERID(num)}"="9259280996"]?delta,s,1)
...

[delta];Звонок от Директора
exten => s,1,Playback(priv-trying)
exten => s,n,ExecIf($["${DIALSTATUS}"="${BUSY}"]?Bridge(SIP/854))
exten => s,n,Dial(SIP/854)

• Asterisk настройка очереди - queues.conf

server# cat queues.conf

[general]
persistentmembers = yes

[queue1]
member => SIP/403 ;установить call-limit=1 для linksys

sever# cat extensions.conf

...
;[incoming]
exten => voip1_00000X,1,Answer()
exten => voip1_00000X,n,Queue(queue1)
...

asterisk# agents.conf

[general]

[01]
fullname=Ivan Ivanovitch Ivanov
[02]
fullname=Petr Petrovitch Petrov
[03]
fullname=Your name

asterisk# cat extensions.conf

[default]
...
exten => _6XX,1,AgentLogin(${EXTEN:1})
...
[agents]
exten => _XX,1,AgentRequest(${EXTEN})
     same => n,Hangup()

asterisk# cat queues.conf

[general]
persistentmembers = yes

[queue1]
strategy=rrmemory
member => Local/01@agents
member => Local/02@agents
member => Local/03@agents

server*CLI> queue show queue1

server*CLI> agent show online

• Asterisk Application_System
• Asterisk cmd System
• System and SHELL

server# cat extensions.conf

[default]
...
exten => 398,1,Authenticate(1234)
exten => 398,n,System(/usr/bin/sudo /sbin/init 6)
...

• !!! Название функции только ЗАГЛАВНЫМИ буквами
• Asterisk func shell

# cat /etc/asterisk/extensions.conf

...
exten => _4XX,1,Set(MY-RES=${SHELL(test -e /etc/asterisk/nocall && echo -n DENY || echo -n ALLOW)})
exten => _4XX,n,Verbose(1,MY-RES is ${MY-RES})
exten => _4XX,n,GotoIf($[${MY-RES} = DENY]?end)
exten => _4XX,n,Dial(SIP/${EXTEN})
exten => _4XX,n(end),Hangup()
...

• Asterisk Gateway Interface (AGI)
• Простой пример AGI на PHP
• Синтез и распознавание речи от Google для Asterisk

# cat /usr/share/asterisk/agi-bin/multi_playback.sh

#!/bin/sh

LOG=/tmp/multi_playback.log
SAMPLE_COUNT=0

:>$LOG

while read ARG && [ "$ARG" ]
do
        echo $ARG >> $LOG
        if echo $ARG | grep -q 'agi_arg_'
        then
                SAMPLE_NAME=`echo $ARG | sed -E 's/.*: (.*)$/\1/'`
                echo EXEC Playback $SAMPLE_NAME
                SAMPLE_COUNT=$(($SAMPLE_COUNT+1))
        fi
done

echo SET VARIABLE SAMPLE-COUNT $SAMPLE_COUNT

exit 0

# cat extensions.conf

...
exten => 313,1,AGI(multi_playback.sh,beep,hello-world,beep)
exten => 313,n,Verbose(1,SAMPLE-COUNT ${SAMPLE-COUNT})
exten => 313,n,Hangup()
...

• http://phpagi.sourceforge.net/

# cd /usr/local/

# wget http://val.bmstu.ru/unix/voip/phpagi-2.20.tgz

# tar -xvf phpagi-2.20.tgz

# cat /usr/share/asterisk/agi-bin/example.php

#!/usr/bin/php
<?php
require('/usr/local/phpagi-2.20/phpagi.php');

$agi = new AGI();

$agi->answer();

$agi->stream_file("hello-world");

$agi->hangup();
?>

• Asterisk Managment Interface (AMI)
• Log in, Log out from Asterisk AMI with Telnet
• Asterisk: обратный вызов с использованием AMI и PHP

# cat manager.conf

[general]
enabled = yes
port = 5038
;bindaddr = 127.0.0.1
;bindaddr = 0.0.0.0

#include "manager.d/*.conf"

# cat manager.d/admin.conf

[admin]
secret = admin
read = all
write = all

# asterisk -rx "module reload manager"

# asterisk -rx "manager show connected"

# asterisk -rx "manager show commands"

# telnet 127.0.0.1 5038

Action: Login
ActionID: 1
Username: admin
Secret: admin

Action: Originate
Channel: sip/403
Context: default
Exten: 301
Callerid: 301
Priority: 1

Action: Logoff

• Getting Started with ARI
• Complete reference of the API
• В Asterisk появился REST Interface (ARI)

# cat /etc/asterisk/http.conf

[general]
enabled=yes
;bindaddr=127.0.0.1
;bindaddr=0.0.0.0
bindport=8088

# cat /etc/asterisk/ari.conf

[general]
enabled = yes
pretty = yes
;allowed_origins = http://ari.asterisk.org
;allowed_origins = chrome-extension://hlnmjkbpmnbgeondjeceaomhafdacmlj

[asterisk]
type = user
read_only = no
password = asterisk

• Asterisk REST API Browser

# curl -v -X GET "http://localhost:8088/ari/endpoints?api_key=asterisk:asterisk"

• How to get all dialer events from Asterisk REST API (ARI)?

# curl -sL https://deb.nodesource.com/setup_9.x | sudo -E bash -

# apt install -y nodejs

# npm install -g wscat

# wscat -c "ws://localhost:8088/ari/events?api_key=asterisk:asterisk&app=my-first-app" 

*CLI> ari show apps

# curl -v -u asterisk:asterisk -X POST "http://localhost:8088/ari/applications/my-first-app/subscription?eventSource=endpoint%3ASIP"

*CLI> ari show app my-first-app

exten => 307,1,Answer()
exten => 307,n,Stasis(my-first-app)
exten => 307,n,Hangup()

# curl -v -u asterisk:asterisk -X GET "http://localhost:8088/ari/channels"

...
    "id": "1513594816.47",
...

# ID=1513594816.47

# curl -v -u asterisk:asterisk -X POST "http://localhost:8088/ari/channels/$ID/play?media=sound:hello-world"

# curl -v -u asterisk:asterisk -X POST "http://localhost:8088/ari/channels/$ID/redirect?endpoint=SIP%2F403"

!!! При использовании GSSAPI надо добавить метод PALIN

• Using XMPP (Jabber) with Asterisk
• Сервис OpenFire Управление контактами

# asterisk -x 'module show' | grep xmpp

# cat xmpp.conf

[general]
debug=no

[aster]
type=client
serverhost=server.corpX.un
;username=administrator@corpX.un
;username=admin@corpX.un
secret=Pa$$w0rd
port=5222
usetls=yes

*CLI> module reload res_xmpp

*CLI> xmpp show connections

*CLI> xmpp show buddies

# cat extensions.conf

[default]
...
exten => 306,1,Answer()
exten => 306,n,JabberSend(aster,user3@corpX.un,Hello World)
exten => 306,n,Hangup()
...

;exten => _4XX,1,Set(MY-CALLED-USER-ID=${SIPPEER(${EXTEN},chanvar[MY-USER-ID])})
;exten => _4XX,1,Set(MY-CALLED-USER-ID=${PJSIP_ENDPOINT(${EXTEN},MY-USER-ID)})

exten => _4XX,n,Set(MY-CALLED-USER-STATUS=${JABBER_STATUS(aster,${MY-CALLED-USER-ID}/Spark)})

exten => _4XX,n,GotoIf($[${MY-CALLED-USER-STATUS} > 2]?busy)

exten => _4XX,n,JabberSend(aster,${MY-CALLED-USER-ID},Вам звонит ${CALLERID(all)})

;exten => _4XX,n,Dial(SIP/${EXTEN})
;exten => _4XX,n,Dial(PJSIP/${EXTEN})
exten => _4XX,n,Hangup()

exten => _4XX,n(busy),Playback(tt-allbusy)
exten => _4XX,n,Hangup()
...

• Скрипт на perl формирующий users.conf и extensions.conf
• Связываем Active Directory, Asterisk и OpenFire
• Asterisk LDAP Realtime driver
• Asterisk Realtime Lightweight Directory Access Protocol (LDAP) Driver
• Asterisk sync with Active Directory

• Заполняем атрибут Telephone number и переименовываем пользователя указав Full name на русском языке

# apt install ldap-utils

# cat /root/create_exten_mail_cidname_table.sh

#!/bin/sh

IFS=' '

ldapsearch -x -D "cn=Administrator,cn=Users,dc=corp15,dc=un" -w 'Pa$$w0rd' -H ldap://ad.corp15.un -b "dc=corp15,dc=un" "(telephoneNumber=4*)" cn telephoneNumber mail \
| perl -p00e 's/\r?\n //g' |
while read s
do
        echo $s | grep -vq '^dn:' && continue
        read fn cn
        echo $fn | grep -q '::' && cn=`echo $cn | base64 -d`
        read fn tn
        read fn email
        echo $tn $email $cn
done

# /root/create_exten_mail_cidname_table.sh | tee exten_mail_cidname_table.txt

401 user1@corp15.un Иван Иванович Иванов
402 user2@corp15.un Петр Петрович Петров
403 user3@corp15.un Сидор Сидорович Сидоров
404 user4@corp15.un Василий Муркович Кошкин

• Provisioning

# cat /root/create_ast_users_conf.sh

#!/bin/sh

IFS=' '

cat $1|
while read exten mail name
do
        echo "[$exten]"
        echo "type=friend"
        echo "fullname=$name"
        echo "setvar=MY-USER-ID=$mail"
        secret=`cat $2 | grep "^$exten" | cut -d' ' -f2`
        echo "secret=$secret"
        echo "host=dynamic"
        echo
done

# /root/create_ast_users_conf.sh exten_mail_cidname_table.txt exten_secret_proxy_mac_model_table.txt | tee /etc/asterisk/users.conf

[401]
type=friend
fullname=Иван Иванович Иванов
setvar=MY-USER-ID=user1@corp15.un
secret=tpassword401
host=dynamic
...

# cat /etc/asterisk/sip.conf

[general]
...
#include "/etc/asterisk/users.conf"
...

# cat create_ast_pjsip_wizard.sh

#!/bin/sh

IFS=' '

cat $1|
while read exten mail name
do
        password=`cat $2 | grep "^$exten" | cut -d' ' -f2`
        echo "[$exten]"
        echo "type=wizard"
        echo "accepts_auth=yes"
        echo "accepts_registrations=yes"
        echo "transport=udp-transport-1"
        echo "inbound_auth/username=$exten"
        echo "inbound_auth/password=$password"
        echo "endpoint/allow=alaw"
        echo "endpoint/context=default"
        echo "endpoint/set_var=MY-USER-ID=$mail"
        echo "endpoint/callerid=${name}<${exten}>"
        echo "aor/max_contacts=1"
        echo "aor/remove_existing=yes"
        echo
done

# /root/create_ast_pjsip_wizard.sh exten_mail_cidname_table.txt exten_secret_proxy_mac_model_table.txt | tee /etc/asterisk/pjsip_wizard.conf

# service asterisk restart

• asterisk + AD perl scripts
• из AD берутся номера телефонов и ФИО
• secret для SIP телефонов задается ст статически в sip.conf

# apt install libnet-ldap-perl

# cat users.pl

...
$ADcontroller="ad.corp15.un";
$ADDC="DC=corp15,DC=un";
$ADuser="Administrator";
$ADpass='Pa$$w0rd';
$dialplan="default";
...

# /root/users.pl > /etc/asterisk/users.conf

# cat /etc/asterisk/users.conf

[401]
cid_number = 401
fullname = Ivan I. Ivanov
type = peer

[402]
...

# cat /etc/asterisk/sip.conf

[general]
...
#include "/etc/asterisk/users.conf"

[401]
secret=tpassword401
type=user
host=dynamic

[402]
...

# cat exten_secret_proxy_mac_model_table.txt

401 tpassword401 server.corpX.un 000000000000 Softphone
402 tpassword402 server.corpX.un 000000000000 Softphone
403 tpassword403 server.corpX.un 000E08C190C2 SPA3102
404 tpassword404 server.corpX.un BCC34221709A KX-HDVXXX
...
403 tpassword403 server.corp8.un 000E08C190A2 SPA3102
403 tpassword403 server.corp6.un BCC34221709A KX-HDVXXX
403 tpassword403 server.corp3.un 080023F67118 KX-HDVXXX
403 tpassword403 server.corp1.un 000E08C190A8 SPA3102
403 tpassword403 server.corp4.un 000E08C190B6 SPA3102
403 tpassword403 server.corp2.un BCC3422170AA KX-HDVXXX
403 tpassword403 server.corp7.un 080023E421CD KX-HDVXXX
403 tpassword403 server.corp5.un 000E08C190C2 SPA3102
403 tpassword403 server.corp9.un 000E08C38F1D SPA3102
403 tpassword403 server.corp10.un 080023BF61CD KX-HDVXXX
403 tpassword403 server.corp11.un 000E08C389BF SPA3102
403 tpassword403 server.corp12.un 080023BF61CC KX-HDVXXX

# cat create_prov_phones_conf.sh

#!/bin/sh

wwwdir=/var/www/html/

while read pn secret sipproxy mac model
do
        if [ "$model" = KX-HDVXXX ]
        then
#               echo $pn $secret $mac $model
                cat > ${wwwdir}kx-hdv-$mac.cfg <<FIN
# Panasonic SIP Phone Standard Format File #

## SIP Settings
PHONE_NUMBER_1="$pn"
SIP_AUTHID_1="$pn"
SIP_PASS_1="$secret"
SIP_PRXY_ADDR_1="$sipproxy"
SIP_RGSTR_ADDR_1="$sipproxy"
DIAL_PLAN_1="*xx|[1-4]xx|0xxxxx|8xxxxxxxxxx"

FIN
        elif [ "$model" = SPA3102 ]
        then
#               echo $pn $secret $mac $model
                cat > ${wwwdir}spa-$mac.cfg <<FIN
<flat-profile>
  <Proxy_1_>$sipproxy</Proxy_1_>
  <User_ID_1_>$pn</User_ID_1_>
  <Password_1_>$secret</Password_1_>
  <Dial_Plan_1_>( xxx | 8xxxxxxxxxx | 0xxxxx | *xx )</Dial_Plan_1_>
</flat-profile>

FIN
        fi
done

# /root/create_prov_phones_conf.sh < exten_secret_proxy_mac_model_table.txt

# ls /var/www/html/

• Интеграция Asterisk и SugarCRM
• Сервис SugarCRM
• Сервис SalesPlatform Vtiger CRM Russian

• All Asterisk Versions

• Security Advisories

• http://www.voip-info.org/wiki/view/SIP+TLS
• http://www.sipring.ru/overview/asterisk-pbx-ready/99-asterisk-sip-tls.html
• https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
• https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport
• Можно использовать самоподписанные серитфикаты или CA
• Cертификат сервера или СА необходимо импортировать на телефоны/компьютеры с софтфонами

server# cat server.crt server.key > /etc/asterisk/server.crtkey

server# cat sip.conf

[general]
...
tlsenable=yes
tlsbindaddr=0.0.0.0:5061
tlscertfile=/etc/asterisk/server.crtkey
; tlscafile=/var/www/html/ca.crt ;need if use clietns certs
tlscipher=ALL
tlsclientmethod=tlsv1
tlsdontverifyserver=no
...
[402]
...
transport=tls
encryption=yes
...

# service asterisk restart

# cat sip.conf

[general]
...
useragent=Microsoft Lync
...

http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret

server# echo -n "402:corp:tpassword402" | md5sum

server# cat sip.conf

[general]
...
realm=corp
...
[402]
;secret=tpassword402
md5secret=29c91aa064cf462f5146b7ee81e7ba49
...

• Защита Asterisk или несколько способов как защититься от подбора пароля к SIP аккаунту
• Asterisk: SIP и безопасность
• "Voip hackers runs up a phone bill worth $120,000".

• Установка WebRTC клиента на Asterisk на примере SipML5
• Sipml5 with Asterisk

• !!! Заработало в Firefox

• Создание само подписанного сертификата для системы server.corpX.un

# cat server.crt server.key > /etc/asterisk/server.crtkey

# cat /etc/asterisk/http.conf

[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/etc/asterisk/server.crtkey

# cat /etc/asterisk/sip.conf

[general]
transport=udp,wss
...

[webrtc](!)
avpf = yes
force_avp = yes
icesupport = yes
encryption = yes
dtlsenable = yes
dtlsverify = no
dtlssetup = actpass
dtlscertfile = /etc/asterisk/server.crtkey

[401](webrtc)
...

!!! Проверить и принять сертификат

https://server.corpX.un:8089/ws

• World's first HTML5 SIP client

Display Name: 401
Private Identity*: 401
Public Identity*: sip:401@server.corpX.un
Password: tpassword401
Realm*: server.corpX.un
WebSocket Server URL: wss://server.corpX.un:8089/ws
ICE Servers: []

• Создание самоподписанного сертификата для системы server.corpX.un
• Поддержка протокола HTTPS

# cd /var/www/html/

# wget http://val.bmstu.ru/unix/voip/sipml5-master.zip

# unzip sipml5-master.zip

https://server.corpX.un/sipml5-master/call.htm

https://habr.com/post/225179/ 

key=f609cf87-d8a9-48c3-b11b-e20054ec3bb3

Можно использовать для Raspberry -  arecord aplay 
plughw:1,0 - где 1 микрофон USB камеры - видно в cat /proc/asound/cards

#!/usr/bin/php
<?php
system("arecord --buffer-time=1000000 -f dat -r 16000 -d 4 -D plughw:1,0 send.wav");

///skip
?>

Ок.
Работает распознавание одной строкой из шела, после записи в файл:

curl "asr.yandex.net/asr_xml?key=f609cf87-d8a9-48c3-b11b-e20054ec3bb3&uuid=01ae13cb744628b58fb536d496daa177&topic=notes&lang=ru-RU" -H "Content-Type: audio/x-wav" --data-binary "@test.waw"

Получаем текст, который нужно распарсить, сравнить с базой и выполнить скрипт:

<?xml version="1.0" encoding="utf-8"?>
<recognitionResults success="1">
        <variant confidence="0">раз раз раз раз раз</variant>
</recognitionResults>

Михаил Калинин.

#add to sip_notify.conf

[spa-resync]
Event=>resync
Content-Length=>0

[spa-report]
Event=>report
Content-Length=>0

[spa-reboot]
Event=>reboot
Content-Length=>0

[spa-restart]
Event=>restart
Content-Length=>0

restart asterisk
# service asterisk restart

run sip notify against the extension and see the result
# asterisk -r
asterisk1*CLI> sip debug peer 401
asterisk1*CLI> sip notify spa-resync 401

http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/__nogroup__.group.html

Убедитесь что  trixbox repository выбран.
yum search hud

yum install hudlite-server.i386

yum install tbm-hudadmin

Создайте  директорию  /etc/asterisk/hud 

Создайте пустой файл user.xml в этой директории: cat > user.xml 

Выйдите из директории hud и модифицируйте права chmod –R 766 hud


HUDlite Admin программа будет исследовать ваши настройки и автоматически создавать учетные записи и пароли для ваших пользователей. Вы можете по желанию сменить пароли созданные автоматически и добавить мобильные номера телефонов  и адрес электронной почты ваших пользователей.



Конфигурация HUDLite клиента.

[Download HUDlite client here]

Приинсталляции выберите автозапуск при HUDlite клиента. Примите пользовательскую лицензию. Далее Вам нужно ввести HUDимя пользователя и пароль. Вы можете получить имя пользователя в  HUDlite административной панели. Если ваше имя пользователя 200 ваше имя пользователя будет sip200. Ваш пароль будет восемь случайных символов. Вы  можете изменить пароли по желанию в HUDliteadmin.
Далее введите имя сервера, пароль для обращения к серверу password (общий для всех), и серверный порт.

    Имя сервера IPaddressвашего trixbox.
    Сервер пароль 'password'
    Сервер порт 6600

Если вы пользуетесь MicrosoftOutlookвыберите инсталяцию с разрешением HUDliteиспользовать Ваш список контактов для callerIDи для установки функции ClicktoDialв Outlook. После этого выберите инсталяцию TAPIдрайвера. Это позволит вам выполнять функцию clicktodialиспользуя любую TAPIиспользуемую приложениями.
HUDliteтеперь инсталирован. Вы должны видеть все доступные номера в HUDокне.

asterisk*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
...
00000X/voip1                172.16.1.X       D   N      1024     Unmonitored

asterisk*CLI> database show

asterisk*CLI> database showkey SIP/Registry/00000X

asterisk*CLI> database deltree SIP/Registry/00000X

server# /etc/init.d/asterisk restart

• How to Install Asterisk on CentOS 7

• Работа с исходными текстами

• Installing Asterisk From Source

# apt install libedit-dev uuid-dev libxml2-dev libsqlite3-dev

# wget https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-16-current.tar.gz

# cd /usr/local/src

# tar -xvzf /root/asterisk-16-current.tar.gz

# cd asterisk-16.3.0/

# ./configure --with-jansson-bundled

# cd contrib/scripts

# ./install_prereq install

# ./install_prereq install-unpackaged

# cd -

# make menuselect

# make

# make install

# make samples

# asterisk

# asterisk -rvvv

Задача: Добавление поддержки imap хранилища голосовой почты

• http://blogs.reucon.com/srt/2007/02/20/1171932900000.html
• http://www.asteriskdocs.org/html/ch14s05.html
• http://etel.wiki.oreilly.com/wiki/index.php/Storing_Voicemail_on_an_IMAP_server

Ubuntu

root@server:~# apt-get install libncurses5-dev libssl-dev libxml2-dev
root@server:~# cd /usr/src

root@server:/usr/src# wget ftp://ftp.cac.washington.edu/mail/imap.tar.Z
root@server:/usr/src# wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.2.19.tar.gz

FreeBSD

[server:~] # pkg install perl5 bison m4 gmake gcc libtool libltdl libxml2 speex unixODBC gsm 

[server:~] # cd /usr/src/
[server:/usr/src] # fetch ftp://ftp.cac.washington.edu/mail/imap.tar.Z
[server:/usr/src] # fetch http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.2.19.tar.gz

Ubuntu/FreeBSD

server:/usr/src# gunzip imap.tar.Z
server:/usr/src# tar -xvf imap.tar
server:/usr/src# tar -xvf asterisk-1.6.2.19.tar.gz

server:/usr/src# cd imap-2007f/

root@server:/usr/src/imap-2007f# make slx EXTRACFLAGS="-I/usr/include/openssl"
или
[server:/usr/src/imap-2007f] # make bsf EXTRACFLAGS="-I/usr/include/openssl"

server:/usr/src/imap-2007f# cd ../asterisk-1.6.2.19
server:/usr/src/asterisk-1.6.2.19# ./configure --with-imap=/usr/src/imap-2007f/ --prefix=/usr/local/asterisk/ --disable-xmldoc
server:/usr/src/asterisk-1.6.2.19# make menuconfig
server:/usr/src/asterisk-1.6.2.19# make
server:/usr/src/asterisk-1.6.2.19# make install clean
server:/usr/src/asterisk-1.6.2.19# make samples

server:~# mkdir -p /usr/local/asterisk/var/db/asterisk/

server:~# /usr/local/asterisk/sbin/asterisk

server:~# /usr/local/asterisk/sbin/asterisk -rvvvvvv

# lame --decode Knockin_on_Heavens_Door.mp3

# sox -V Knockin_on_Heavens_Door.mp3.wav -r 8000 -c 1 Knockin_on_Heavens_Door.mp3.raw

# cp Knockin_on_Heavens_Door.mp3.raw /var/lib/asterisk/moh/

# cp Knockin_on_Heavens_Door.mp3.raw /usr/local/share/asterisk/moh/

# cat musiconhold.conf

...
[default]
mode=files
directory=moh
...

# cat extensions.conf

...
exten => 306,1,Answer
exten => 306,2,MusicOnHold()
...