Сервис Asterisk

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

• Install Asterisk via apt or from source?

root@server:~# apt update

root@server:~# apt install asterisk

# apt purge asterisk-config asterisk

# apt autoremove

• 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 /etc/asterisk/sip.conf

[general]
;udpbindaddr=0.0.0.0:6050
transport=udp
disallow=all
allow=alaw

directmedia=no           ; for webinar
nat=force_rport,comedia  ; for webinar
qualify=yes              ; for webinar

;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=Петр Петров<402>
;cid_number=402
;fullname=Петр Петров
;disallow=all
;allow=gsm
;call-limit=1
;deny=0.0.0.0/0.0.0.0
;permit=172.16.1.0/255.255.255.0
;directmedia=no
;nat=force_rport,comedia
;qualify=yes

;[user2]
;secret=password2
;...

;[corpY]
;type=peer
;context=default
;host=server.corpY.un
;directmedia=no        ;need for video?

server# cat /etc/asterisk/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

[internet](!)
type=friend
host=dynamic
permit=0.0.0.0/0.0.0.0
nat=force_rport,comedia
qualify=yes
directmedia=no
;call-limit=1

[401](office)
secret=tpassword401

[402](internet)
secret=tpassword402

[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

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=Иван Иванов<401>
;set_var=MY-USER-ID=user1@corpX.un
;set_var=MY-PSTN-CALLER-ID=8495123456X
;force_rport=yes
;rewrite_contact=yes
;rtp_symmetric=yes
;direct_media=no

[401]
type=aor
max_contacts=1
remove_existing=yes
;qualify_frequency=60

[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

[corp]
type=endpoint
transport=udp-transport-1
context=default
disallow=all
allow=alaw
aors=corp
;force_rport=yes
;rewrite_contact=yes
;rtp_symmetric=yes
;direct_media=no

[corp]
type=aor
contact=sip:server.corp.un:5060
;qualify_frequency=60

[corp]
type=identify
endpoint=corp
match=server.corp.un

# cat /etc/asterisk/pjsip.conf

...
; voip1.un и server.corp.un (курс CGP) одна и та же система
; приходится подключаться к ней с разных портов :)

[udp-transport-2]
type=transport
protocol=udp
bind=0.0.0.0:5062
;local_net=192.168.1.0/255.255.255.0
;local_net=192.168.X.0/255.255.255.0
;external_media_address=172.16.1.X
;external_signaling_address=172.16.1.X

[voip1_00000X]
type=endpoint
transport=udp-transport-2
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=auth
auth_type=userpass
password=spasswordX
username=00000X

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

[voip1_00000X]
type=registration
transport=udp-transport-2
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=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 /etc/asterisk/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 => 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 /etc/asterisk/sip.conf

[general]
...
disallow=all
allow=alaw
...
allow=h263
allow=h264
videosupport=yes
avpf=yes
...

• HELP - New Installation Full of Errors

server# asterisk -rvvv

server*CLI> core set verbose 1

server# cat /etc/asterisk/logger.conf

...
[logfiles]
...
console => warning,error,dtmf
messages => notice,warning,error,verbose(3)
...

# 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

server*CLI> rtp set debug on 

• Dialplan search order

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

server# cat /etc/asterisk/extensions.conf

[default]

...

;[incoming]

include=>incoming_work_time,10:00-16:59,mon-fri,*,*
include=>incoming_work_time,10:00-13:59,sat,*,*

include=>incoming_not_work_time,17:00-9:59,mon-fri,*,*
include=>incoming_not_work_time,14:00-9:59,sat,*,*
include=>incoming_not_work_time,*,sun,*,*


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

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

• Настройка Asterisk для связи с VoIP провайдерами

server# cat /etc/asterisk/sip.conf

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

server# cat /etc/asterisk/extensions.conf

...
exten => _8XXXXXXXXXX,1,Dial(SIP/voip1_00000X/${EXTEN})

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

;exten => _[+789]XXXXXXXXXX!,1,GoSub(sub-monitor,s,1)
;exten => _[+789]XXXXXXXXXX!,n,Dial(SIP/voip1_00000X/8${EXTEN:-10})

server# cat /etc/asterisk/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 /etc/asterisk/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

server*CLI> pjsip show registrations

• ALSA Asterisk Console on RasPBX

phone2# cat /etc/asterisk/extensions.conf

[default]

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

debian# cat /etc/asterisk/modules.conf

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

rasberry# 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 /etc/asterisk/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 /etc/asterisk/extensions.conf

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

• Подключение Asterisk к VoIP провайдеру по протоколу SIP

root@voip1.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@voip1.un:~# cat extensions.conf

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

server.corpX.un# cat /etc/asterisk/iax.conf

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

[corp]
type=user
;context=default
host=dynamic
secret=apassword
auth=md5

[corp]
type=peer
host=server.corp.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 /etc/asterisk/extensions.conf

[default]
...
exten => _000XXX,1,Set(CALLERID(num)=00X${CALLERID(num)})
exten => _000XXX,n,Dial(IAX2/corp/${EXTEN:3})
;exten => _000XXX,n,Dial(PJSIP/${EXTEN:3}@corp)

;exten => _00Y[34]XX,1,Set(CALLERID(num)=00X${CALLERID(num)})
;exten => _00Y[34]XX,n,Dial(IAX2/corpY/${EXTEN:3})

• Сервис Ansible Использование шаблонов

;;пример универсального шаблона для звонков в офисы 
;;с 1-го по 9-й и c 10-го по 13-й

exten => _00X[34]XX,1,Set(CALLERID(num)=00X${CALLERID(num)})
exten => _00X[34]XX,n,Dial(IAX2/corp${EXTEN:2:1}/${EXTEN:3})

exten => _01[0-3][34]XX,1,Set(CALLERID(num)=00X${CALLERID(num)})
exten => _01[0-3][34]XX,n,Dial(IAX2/corp${EXTEN:1:2}/${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>

# watch -d "asterisk -x 'cdr show active'"

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

server# cat /etc/asterisk/cdr.conf

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

• Asterisk 13 Function_CHANNEL

...
     same => n,Set(CDR(userfield)=${CHANNEL(recvip)})             ;SIP
...
     same => n,Set(CDR(userfield)=${CHANNEL(pjsip,remote_addr)})  ;PJSIP
...

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# :> /etc/asterisk/cdr.conf

server# cat /etc/asterisk/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/syslog

• MySQL CDR Backend

# 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

# cat /etc/asterisk/extensions.conf

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

• https://code.google.com/p/asterisk-cdr-viewer/wiki/ReadMe
• Значительно переработанная версия Web-интерфейса 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;
...

# usermod -G asterisk www-data

# service apache2 restart

http://172.16.1.X/asterisk-cdr-viewer/

• Спасибо Комлеву Александру Павловичу!

Чтобы запись разговора воспроизводилась прямо на странице CDR Viewer, нужно в файле 
/var/www/html/asterisk-cdr-viewer/include/functions.inc.php
заменить  строку:

echo "    <td class=\"record_col\"><a href=\"download.php?audio=$recorded_file.$system_audio_format\" title=\"Listen to call recording\"><img src=\"templates/images/sound.png\" alt=\"Call recording\" /></a></td>\n";

на такую:

echo "    <td class=\"record_col\"><a href=\"download.php?audio=$recorded_file.$system_audio_format\" title=\"Listen to call recording\"><img src=\"templates/images/sound.png\" alt=\"Call recording\" /></a><audio controls><source src=\"download.php?audio=$recorded_file.$system_audio_format\"></audio></td>\n";

• Asterisk PJSIP Realtime (asterisk-pbx.ru)
• Setting up PJSIP Realtime (wiki.asterisk.org)

CREATE TABLE `ps_aors` (
  `id` varchar(40) NOT NULL,
  `contact` varchar(255) DEFAULT NULL,
  `default_expiration` int(11) DEFAULT NULL,
  `mailboxes` varchar(80) DEFAULT NULL,
  `max_contacts` int(11) DEFAULT NULL,
  `minimum_expiration` int(11) DEFAULT NULL,
  `remove_existing` enum('yes','no') DEFAULT NULL,
  `qualify_frequency` int(11) DEFAULT NULL,
  `authenticate_qualify` enum('yes','no') DEFAULT NULL,
  `maximum_expiration` int(11) DEFAULT NULL,
  `outbound_proxy` varchar(40) DEFAULT NULL,
  `support_path` enum('yes','no') DEFAULT NULL,
  `qualify_timeout` float DEFAULT NULL,
  `voicemail_extension` varchar(40) DEFAULT NULL,
  UNIQUE KEY `id` (`id`),
  KEY `ps_aors_id` (`id`),
  KEY `ps_aors_qualifyfreq_contact` (`qualify_frequency`,`contact`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8;

CREATE TABLE `ps_auths` (
  `id` varchar(40) NOT NULL,
  `auth_type` enum('md5','userpass') DEFAULT NULL,
  `nonce_lifetime` int(11) DEFAULT NULL,
  `md5_cred` varchar(40) DEFAULT NULL,
  `password` varchar(80) DEFAULT NULL,
  `realm` varchar(40) DEFAULT NULL,
  `username` varchar(40) DEFAULT NULL,
  UNIQUE KEY `id` (`id`),
  KEY `ps_auths_id` (`id`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8;

CREATE TABLE `ps_contacts` (
  `id` varchar(255) DEFAULT NULL,
  `uri` varchar(255) DEFAULT NULL,
  `expiration_time` bigint(20) DEFAULT NULL,
  `qualify_frequency` int(11) DEFAULT NULL,
  `outbound_proxy` varchar(40) DEFAULT NULL,
  `path` text,
  `user_agent` varchar(255) DEFAULT NULL,
  `qualify_timeout` float DEFAULT NULL,
  `reg_server` varchar(20) DEFAULT NULL,
  `authenticate_qualify` enum('yes','no') DEFAULT NULL,
  `via_addr` varchar(40) DEFAULT NULL,
  `via_port` int(11) DEFAULT NULL,
  `call_id` varchar(255) DEFAULT NULL,
  `endpoint` varchar(40) DEFAULT NULL,
  `prune_on_boot` enum('yes','no') DEFAULT NULL,
  UNIQUE KEY `id` (`id`),
  UNIQUE KEY `ps_contacts_uq` (`id`,`reg_server`),
  KEY `ps_contacts_id` (`id`),
  KEY `ps_contacts_qualifyfreq_exp` (`qualify_frequency`,`expiration_time`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8;

CREATE TABLE `ps_domain_aliases` (
  `id` varchar(40) NOT NULL,
  `domain` varchar(80) DEFAULT NULL,
  UNIQUE KEY `id` (`id`),
  KEY `ps_domain_aliases_id` (`id`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8;


CREATE TABLE `ps_endpoint_id_ips` (
  `id` varchar(40) NOT NULL,
  `endpoint` varchar(40) DEFAULT NULL,
  `match` varchar(80) DEFAULT NULL,
  UNIQUE KEY `id` (`id`),
  KEY `ps_endpoint_id_ips_id` (`id`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8;


CREATE TABLE `ps_endpoints` (
  `id` varchar(40) NOT NULL,
  `transport` varchar(40) DEFAULT NULL,
  `aors` varchar(200) DEFAULT NULL,
  `auth` varchar(40) DEFAULT NULL,
  `context` varchar(40) DEFAULT NULL,
  `disallow` varchar(200) DEFAULT NULL,
  `allow` varchar(200) DEFAULT NULL,
  `direct_media` enum('yes','no') DEFAULT NULL,
  `connected_line_method` enum('invite','reinvite','update') DEFAULT NULL,
  `direct_media_method` enum('invite','reinvite','update') DEFAULT NULL,
  `direct_media_glare_mitigation` enum('none','outgoing','incoming') DEFAULT NULL,
  `disable_direct_media_on_nat` enum('yes','no') DEFAULT NULL,
  `dtmf_mode` enum('rfc4733','inband','info','auto') DEFAULT NULL,
  `external_media_address` varchar(40) DEFAULT NULL,
  `force_rport` enum('yes','no') DEFAULT NULL,
  `ice_support` enum('yes','no') DEFAULT NULL,
  `identify_by` enum('username','auth_username') DEFAULT NULL,
  `mailboxes` varchar(40) DEFAULT NULL,
  `moh_suggest` varchar(40) DEFAULT NULL,
  `outbound_auth` varchar(40) DEFAULT NULL,
  `outbound_proxy` varchar(40) DEFAULT NULL,
  `rewrite_contact` enum('yes','no') DEFAULT NULL,
  `rtp_ipv6` enum('yes','no') DEFAULT NULL,
  `rtp_symmetric` enum('yes','no') DEFAULT NULL,
  `send_diversion` enum('yes','no') DEFAULT NULL,
  `send_pai` enum('yes','no') DEFAULT NULL,
  `send_rpid` enum('yes','no') DEFAULT NULL,
  `timers_min_se` int(11) DEFAULT NULL,
  `timers` enum('forced','no','required','yes') DEFAULT NULL,
  `timers_sess_expires` int(11) DEFAULT NULL,
  `callerid` varchar(40) DEFAULT NULL,
  `callerid_privacy` enum('allowed_not_screened','allowed_passed_screened','allowed_failed_screened','allowed','prohib_not_screened','prohib_passed_screened','prohib_failed_screened','prohib','unavailable') DEFAULT NULL,
  `callerid_tag` varchar(40) DEFAULT NULL,
  `100rel` enum('no','required','yes') DEFAULT NULL,
  `aggregate_mwi` enum('yes','no') DEFAULT NULL,
  `trust_id_inbound` enum('yes','no') DEFAULT NULL,
  `trust_id_outbound` enum('yes','no') DEFAULT NULL,
  `use_ptime` enum('yes','no') DEFAULT NULL,
  `use_avpf` enum('yes','no') DEFAULT NULL,
  `media_encryption` enum('no','sdes','dtls') DEFAULT NULL,
  `inband_progress` enum('yes','no') DEFAULT NULL,
  `call_group` varchar(40) DEFAULT NULL,
  `pickup_group` varchar(40) DEFAULT NULL,
  `named_call_group` varchar(40) DEFAULT NULL,
  `named_pickup_group` varchar(40) DEFAULT NULL,
  `device_state_busy_at` int(11) DEFAULT NULL,
  `fax_detect` enum('yes','no') DEFAULT NULL,
  `t38_udptl` enum('yes','no') DEFAULT NULL,
  `t38_udptl_ec` enum('none','fec','redundancy') DEFAULT NULL,
  `t38_udptl_maxdatagram` int(11) DEFAULT NULL,
  `t38_udptl_nat` enum('yes','no') DEFAULT NULL,
  `t38_udptl_ipv6` enum('yes','no') DEFAULT NULL,
  `tone_zone` varchar(40) DEFAULT NULL,
  `language` varchar(40) DEFAULT NULL,
  `one_touch_recording` enum('yes','no') DEFAULT NULL,
  `record_on_feature` varchar(40) DEFAULT NULL,
  `record_off_feature` varchar(40) DEFAULT NULL,
  `rtp_engine` varchar(40) DEFAULT NULL,
  `allow_transfer` enum('yes','no') DEFAULT NULL,
  `allow_subscribe` enum('yes','no') DEFAULT NULL,
  `sdp_owner` varchar(40) DEFAULT NULL,
  `sdp_session` varchar(40) DEFAULT NULL,
  `tos_audio` varchar(10) DEFAULT NULL,
  `tos_video` varchar(10) DEFAULT NULL,
  `sub_min_expiry` int(11) DEFAULT NULL,
  `from_domain` varchar(40) DEFAULT NULL,
  `from_user` varchar(40) DEFAULT NULL,
  `mwi_from_user` varchar(40) DEFAULT NULL,
  `dtls_verify` varchar(40) DEFAULT NULL,
  `dtls_rekey` varchar(40) DEFAULT NULL,
  `dtls_cert_file` varchar(200) DEFAULT NULL,
  `dtls_private_key` varchar(200) DEFAULT NULL,
  `dtls_cipher` varchar(200) DEFAULT NULL,
  `dtls_ca_file` varchar(200) DEFAULT NULL,
  `dtls_ca_path` varchar(200) DEFAULT NULL,
  `dtls_setup` enum('active','passive','actpass') DEFAULT NULL,
  `srtp_tag_32` enum('yes','no') DEFAULT NULL,
  `media_address` varchar(40) DEFAULT NULL,
  `redirect_method` enum('user','uri_core','uri_pjsip') DEFAULT NULL,
  `set_var` text,
  `cos_audio` int(11) DEFAULT NULL,
  `cos_video` int(11) DEFAULT NULL,
  `message_context` varchar(40) DEFAULT NULL,
  `force_avp` enum('yes','no') DEFAULT NULL,
  `media_use_received_transport` enum('yes','no') DEFAULT NULL,
  `accountcode` varchar(80) DEFAULT NULL,
  `user_eq_phone` enum('yes','no') DEFAULT NULL,
  `moh_passthrough` enum('yes','no') DEFAULT NULL,
  `media_encryption_optimistic` enum('yes','no') DEFAULT NULL,
  `rpid_immediate` enum('yes','no') DEFAULT NULL,
  `g726_non_standard` enum('yes','no') DEFAULT NULL,
  `rtp_keepalive` int(11) DEFAULT NULL,
  `rtp_timeout` int(11) DEFAULT NULL,
  `rtp_timeout_hold` int(11) DEFAULT NULL,
  `bind_rtp_to_media_address` enum('yes','no') DEFAULT NULL,
  `voicemail_extension` varchar(40) DEFAULT NULL,
  `mwi_subscribe_replaces_unsolicited` int(11) DEFAULT NULL,
  `deny` varchar(95) DEFAULT NULL,
  `permit` varchar(95) DEFAULT NULL,
  `acl` varchar(40) DEFAULT NULL,
  `contact_deny` varchar(95) DEFAULT NULL,
  `contact_permit` varchar(95) DEFAULT NULL,
  `contact_acl` varchar(40) DEFAULT NULL,
  `subscribe_context` varchar(40) DEFAULT NULL,
  `fax_detect_timeout` int(11) DEFAULT NULL,
  `contact_user` varchar(80) DEFAULT NULL,
  `asymmetric_rtp_codec` enum('yes','no') DEFAULT NULL,
  UNIQUE KEY `id` (`id`),
  KEY `ps_endpoints_id` (`id`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8;


CREATE TABLE `ps_globals` (
  `id` varchar(40) NOT NULL,
  `max_forwards` int(11) DEFAULT NULL,
  `user_agent` varchar(255) DEFAULT NULL,
  `default_outbound_endpoint` varchar(40) DEFAULT NULL,
  `debug` varchar(40) DEFAULT NULL,
  `endpoint_identifier_order` varchar(40) DEFAULT NULL,
  `max_initial_qualify_time` int(11) DEFAULT NULL,
  `default_from_user` varchar(80) DEFAULT NULL,
  `keep_alive_interval` int(11) DEFAULT NULL,
  `regcontext` varchar(80) DEFAULT NULL,
  `contact_expiration_check_interval` int(11) DEFAULT NULL,
  `default_voicemail_extension` varchar(40) DEFAULT NULL,
  `disable_multi_domain` enum('yes','no') DEFAULT NULL,
  `unidentified_request_count` int(11) DEFAULT NULL,
  `unidentified_request_period` int(11) DEFAULT NULL,
  `unidentified_request_prune_interval` int(11) DEFAULT NULL,
  `default_realm` varchar(40) DEFAULT NULL,
  `mwi_tps_queue_high` int(11) DEFAULT NULL,
  `mwi_tps_queue_low` int(11) DEFAULT NULL,
  `mwi_disable_initial_unsolicited` enum('yes','no') DEFAULT NULL,
  `ignore_uri_user_options` enum('yes','no') DEFAULT NULL,
  UNIQUE KEY `id` (`id`),
  KEY `ps_globals_id` (`id`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8;


CREATE TABLE `ps_registrations` (
  `id` varchar(40) NOT NULL,
  `auth_rejection_permanent` enum('yes','no') DEFAULT NULL,
  `client_uri` varchar(255) DEFAULT NULL,
  `contact_user` varchar(40) DEFAULT NULL,
  `expiration` int(11) DEFAULT NULL,
  `max_retries` int(11) DEFAULT NULL,
  `outbound_auth` varchar(40) DEFAULT NULL,
  `outbound_proxy` varchar(40) DEFAULT NULL,
  `retry_interval` int(11) DEFAULT NULL,
  `forbidden_retry_interval` int(11) DEFAULT NULL,
  `server_uri` varchar(255) DEFAULT NULL,
  `transport` varchar(40) DEFAULT NULL,
  `support_path` enum('yes','no') DEFAULT NULL,
  `fatal_retry_interval` int(11) DEFAULT NULL,
  `line` enum('yes','no') DEFAULT NULL,
  `endpoint` varchar(40) DEFAULT NULL,
  UNIQUE KEY `id` (`id`),
  KEY `ps_registrations_id` (`id`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8;



CREATE TABLE `ps_subscription_persistence` (
  `id` varchar(40) NOT NULL,
  `packet` varchar(2048) DEFAULT NULL,
  `src_name` varchar(128) DEFAULT NULL,
  `src_port` int(11) DEFAULT NULL,
  `transport_key` varchar(64) DEFAULT NULL,
  `local_name` varchar(128) DEFAULT NULL,
  `local_port` int(11) DEFAULT NULL,
  `cseq` int(11) DEFAULT NULL,
  `tag` varchar(128) DEFAULT NULL,
  `endpoint` varchar(40) DEFAULT NULL,
  `expires` int(11) DEFAULT NULL,
  UNIQUE KEY `id` (`id`),
  KEY `ps_subscription_persistence_id` (`id`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8;


CREATE TABLE `ps_systems` (
  `id` varchar(40) NOT NULL,
  `timer_t1` int(11) DEFAULT NULL,
  `timer_b` int(11) DEFAULT NULL,
  `compact_headers` enum('yes','no') DEFAULT NULL,
  `threadpool_initial_size` int(11) DEFAULT NULL,
  `threadpool_auto_increment` int(11) DEFAULT NULL,
  `threadpool_idle_timeout` int(11) DEFAULT NULL,
  `threadpool_max_size` int(11) DEFAULT NULL,
  `disable_tcp_switch` enum('yes','no') DEFAULT NULL,
  UNIQUE KEY `id` (`id`),
  KEY `ps_systems_id` (`id`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8;



CREATE TABLE `ps_transports` (
  `id` varchar(40) NOT NULL,
  `async_operations` int(11) DEFAULT NULL,
  `bind` varchar(40) DEFAULT NULL,
  `ca_list_file` varchar(200) DEFAULT NULL,
  `cert_file` varchar(200) DEFAULT NULL,
  `cipher` varchar(200) DEFAULT NULL,
  `domain` varchar(40) DEFAULT NULL,
  `external_media_address` varchar(40) DEFAULT NULL,
  `external_signaling_address` varchar(40) DEFAULT NULL,
  `external_signaling_port` int(11) DEFAULT NULL,
  `method` enum('default','unspecified','tlsv1','sslv2','sslv3','sslv23') DEFAULT NULL,
  `local_net` varchar(40) DEFAULT NULL,
  `password` varchar(40) DEFAULT NULL,
  `priv_key_file` varchar(200) DEFAULT NULL,
  `protocol` enum('udp','tcp','tls','ws','wss') DEFAULT NULL,
  `require_client_cert` enum('yes','no') DEFAULT NULL,
  `verify_client` enum('yes','no') DEFAULT NULL,
  `verify_server` enum('yes','no') DEFAULT NULL,
  `tos` varchar(10) DEFAULT NULL,
  `cos` int(11) DEFAULT NULL,
  `allow_reload` enum('yes','no') DEFAULT NULL,
  UNIQUE KEY `id` (`id`),
  KEY `ps_transports_id` (`id`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8;

# cat /etc/asterisk/sorcery.conf

[res_pjsip]
endpoint=realtime,ps_endpoints
auth=realtime,ps_auths
aor=realtime,ps_aors
domain_alias=realtime,ps_domain_aliases
contact=realtime,ps_contacts

[res_pjsip_endpoint_identifier_ip]
identify=realtime,ps_endpoint_id_ips

insert into ps_aors (id, max_contacts) values (403, 1);
insert into ps_aors (id, max_contacts) values (404, 1);
insert into ps_auths (id, auth_type, password, username) values (403, 'userpass', 'tpassword403', 403);
insert into ps_auths (id, auth_type, password, username) values (404, 'userpass', 'tpassword404', 404);
insert into ps_endpoints (id, transport, aors, auth, context, disallow, allow, direct_media) values (403, 'udp-transport-1', '403', '403', 'default', 'all', 'alaw', 'no');
insert into ps_endpoints (id, transport, aors, auth, context, disallow, allow, direct_media) values (404, 'udp-transport-1', '404', '404', 'default', 'all', 'alaw', 'no');

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

# cat chan_sip_realtime.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 < chan_sip_realtime.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 /etc/asterisk/sip.conf

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

# cat /etc/asterisk/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

# asterisk -x 'module show' | grep odbc

# cat /etc/asterisk/res_odbc.conf

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

server*CLI> core reload

server*CLI> odbc show all

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

  Name:   asterisk
  DSN:    asterisk
    Number of active connections: 1 (out of 1)

# cat /etc/asterisk/cdr_adaptive_odbc.conf

[default]
connection=asterisk
table=cdr

# service asterisk restart

server*CLI> cdr show status   !!! Ничего не изменилось, поискать другую команду

# cat /etc/asterisk/modules.conf

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

# cat /etc/asterisk/extconfig.conf

[settings]
sippeers => odbc,asterisk,sipfriends

ps_endpoints => odbc,asterisk
ps_auths => odbc,asterisk
ps_aors => odbc,asterisk
ps_domain_aliases => odbc,asterisk
ps_endpoint_id_ips => odbc,asterisk
ps_contacts => odbc,asterisk

# cat /etc/asterisk/func_odbc.conf

...

[LAST_DSTCHANNEL_BY_CALLERID]
dsn=asterisk
readsql=SELECT dstchannel FROM cdr WHERE src='${ARG1}' AND disposition='ANSWERED' ORDER BY calldate DESC LIMIT 1;

# cat /etc/asterisk/extensions.conf

...
exten => voip1_00000X,1,Set(MY-DST-CHAN=${ODBC_LAST_DSTCHANNEL_BY_CALLERID(${CALLERID(num)})})
exten => voip1_00000X,n,Set(MY-DST-CHAN=${CUT(MY-DST-CHAN,"-",1)})
exten => voip1_00000X,n,ExecIf($[${LEN(${MY-DST-CHAN})} = 0]?Set(MY-DST-CHAN=SIP/401&SIP/402&SIP/403&SIP/404))
exten => voip1_00000X,n,Dial(${MY-DST-CHAN},10)
exten => voip1_00000X,n,Dial(SIP/401&SIP/402&SIP/403&SIP/404)
...

• 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 /etc/asterisk/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 /etc/asterisk/sip.conf

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

• С сопровождение и слепой
• Клавиша “flash” аналогового телефона
• Кнопка “transfer” ip телефона
• Программируемая комбинация DTMF

server# cat /etc/asterisk/features.conf

[general]
...
featuredigittimeout = 3000
...
[featuremap]
...
blindxfer => *21
atxfer => *22
...

server# cat /etc/asterisk/extensions.conf

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

exten => voip1_00000X,1,Dial(SIP/401,10,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 /etc/asterisk/sip.conf

[general]
...
allowsubscribe=yes
notifyhold=yes
callcounter=yes
notifyringing=yes
limitonpeers=yes
subscribecontext=default
...
[4NN]
...
;allowsubscribe=no
...

# cat /etc/asterisk/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

• Call Pickup

server# cat /etc/asterisk/sip.conf

...
[401]
callgroup=1
pickupgroup=1
;namedcallgroup=gr1,gr2,gr3
;namedpickupgroup=gr2,gr3,gr4
...
[403]
callgroup=1
pickupgroup=1

server# cat /etc/asterisk/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 /etc/asterisk/features.conf

asterisk13# less /etc/asterisk/res_parking.conf

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

server# cat /etc/asterisk/extensions.conf

[default]

include => parkedcalls
...
exten => _8XXXXXXXXXX,1,Dial(SIP/voip1_00000X/${EXTEN},,T)
...
exten => voip1_00000X,1,Dial(SIP/401,,t)
...

*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

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 /etc/asterisk/asterisk.conf

...
[options]
...
defaultlanguage = ru
...

# service asterisk restart

server# cat /etc/asterisk/sip.conf

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

server# cat /etc/asterisk/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 /etc/asterisk/voicemail.conf

[general]

...

;;;For IMAP/SMTP
;charset=UTF-8
;locale=ru_RU.UTF-8
;emailsubject=Голосовое сообщение от абонента ${VM_CALLERID}
;emailbody=Здравствуйте ${VM_NAME}\n\n${VM_DATE} получено голосовое сообщение от абонента ${VM_CALLERID}
;emaildateformat=%A, %B %d, %Y в %r

...

;;;For IMAP
;;;debian# apt install asterisk-voicemail-imapstorage
;;;debian# cp /etc/dovecot/private/dovecot.pem /usr/local/share/ca-certificates/dovecot.crt
;;;debian# update-ca-certificates
;;;debian# service asterisk restart
;imapserver=server.corpX.un
;imapfolder=INBOX

;;;For SMTP
;debian# apt install postfix mailutils

...

[corpX]
;401 => ,,
;402 => 2222,,
;403 => ,Ваше Имя Отчество Фамилия,user3@corpX.un,,attache=yes|delete=yes
;404 => 4444,Ваше Имя Отчество Фамилия,,,imapuser=student|imappassword=password

server*CLI> voicemail reload

server# cat /etc/asterisk/extensions.conf

[default]
...
exten => 390,1,VoiceMailMain(@corpX)

;exten => 391,1,VoiceMailMain(${CALLERID(num)}@corpX)
;exten => asterisk,1,VoiceMailMain(${CALLERID(num)}@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)}" = "${EXTEN}"]?MAIL)
exten => _4XX,n,Dial(SIP/${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 /etc/asterisk/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
• Asterisk app: Originate

• 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 /etc/asterisk/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 601"
...

• Проблемы с CallerID

# asterisk -x "channel originate SIP/401 extension 601@default"
# asterisk -x "channel originate SIP/402 extension 601@default"
# asterisk -x "channel originate SIP/403 extension 601@default"

; сначала вызов телефона оператора, затем вызов абонента
server*CLI> channel originate SIP/403 extension 89161234567@default

; Сначала вызов абонента, затем обработчика (например - IVR)
server*CLI> channel originate SIP/voip1_00000X/89161234567 extension voip1_00000X@default
server*CLI> channel originate PJSIP/89161234567@voip1_00000X extension voip1_00000X@default

; сначала вызов обработчика (например - все телефоны в офисе или оператор/агент из членов очереди), затем вызов абонента
server*CLI> channel originate Local/voip1_00000X@default extension 89161234567@default

# cat /usr/share/originate_cli.sh

#!/bin/sh

asterisk -x "channel originate SIP/$1 extension $2@default"

# /usr/share/originate_cli.sh 403 601

• Asterisk Call Files
• Сервис atrun

# cat call.txt

Channel: Local/voip1_00000X@default
Extension: 89161234567
Context: default
CallerId: CorpX<84992636363>

server# grep astspooldir /etc/asterisk/asterisk.conf

server# cp call.txt /var/spool/asterisk/outgoing/

server# ls /var/spool/asterisk/outgoing_done

# cat /usr/share/originate_call_file.sh

#!/bin/sh

cat <<EOF >/var/spool/asterisk/outgoing/$1$2$3
Channel: SIP/$1
Extension: $2
Context: default
CallerId: $2<$3>
EOF

# chmod +x /usr/share/originate_call_file.sh

# /usr/share/originate_call_file.sh 401 601 Conferencia
# /usr/share/originate_call_file.sh 402 601 Conferencia
# /usr/share/originate_call_file.sh 401 601 Conferencia

• Asterisk AMI

• Asterisk ARI

# curl -v -X POST "http://localhost:8088/ari/channels?endpoint=SIP%2F403&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%2F403&extension=301&context=default&callerId=301"

sever# cat /etc/asterisk/extensions.conf

...
exten => 302,1,Authenticate(1234)
exten => 302,n,Playback(vm-intro)
exten => 302,n,Record(recordings/speech.alaw,,,xk)
exten => 302,n,Hangup()

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

# ls -lh /usr/share/asterisk/sounds/recordings/

• Asterisk standard extensions

sever# cat /etc/asterisk/extensions.conf

...
exten => 304,1,Goto(menu,s,1) ; for debug
...
exten => voip1_00000X,1,Goto(menu,s,1)
...
[menu]
exten => s,1,Background(recordings/speech)
exten => s,n,Goto(s,1)
 
; exten => 1,1,Dial(SIP/401)
; exten => 2,1,Dial(SIP/402)
; exten => 3,1,Dial(SIP/403)

; exten => _X,1,Dial(SIP/40${EXTEN})

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

exten => _4XX,1,Goto(default,${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)

;[menu2]
;exten => s,1,Background(silence/8)
;exten => s,n,WaitExten(8)    ; работает только после Answer или Background
;exten => _4XX,1,Goto(default,${EXTEN},1)
;exten => t,1,Goto(menu,s,1)  ; timeout exceeded, работает только с WaitExten
;exten => i,1,Goto(menu,s,1)  ; error extensions

• The Asterisk Extension Language (Язык Расширения Астериска)

sever# cat /etc/asterisk/extensions.conf

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

sever# cat /etc/asterisk/extensions.ael

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

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

• МОДУЛЬ CHANSPY В ASTERISK

server# cat /etc/asterisk/extensions.conf

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

• Если создается два файла - in и out, то (можно смикшировать в один, используя утилиту sox)
• touch mix monitor префикс

server# cat /etc/asterisk/features.conf

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

server# cat /etc/asterisk/extensions.conf

...
exten => _4XX,n,Set(TOUCH_MONITOR=${STRFTIME(${EPOCH},,%Y.%m.%d-%H.%M)}_${CALLERID(number)}_${EXTEN})
exten => _4XX,n,Dial(SIP/${EXTEN},,wW)
...
exten => _8XXXXXXXXXX,1,Set(TOUCH_MONITOR=${STRFTIME(${EPOCH},,%Y.%m.%d-%H.%M)}_${CALLERID(number)}_${EXTEN})
exten => _8XXXXXXXXXX,n,Dial(SIP/voip1_00000X/${EXTEN},,W)
..

server*CLI> core reload

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

server# cat /etc/asterisk/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)

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 /etc/asterisk/extensions.conf

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

server# mkdir /usr/share/asterisk/incoming

server# cp -v /usr/share/asterisk/sounds/recordings/speech.alaw /usr/share/asterisk/incoming/

server# mkdir /usr/share/asterisk/busy/

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

server# mkdir /usr/share/asterisk/silence/

server# cp -v /usr/share/asterisk/sounds/en/silence/10.* /usr/share/asterisk/silence/

server# cat /etc/asterisk/musiconhold.conf

...
[incoming]
mode=files
directory=incoming

[busy]
mode=files
directory=busy

[silence]
mode=files
directory=silence

server# cat /etc/asterisk/extensions.conf

...
exten => voip1_00000X,1,Dial(SIP/4NN,,m(incoming)t)

• Сервис Festival

# asterisk -x 'module show' | grep fest

server# cat /etc/asterisk/extensions.conf

...
exten => 324,1,Answer()
;exten => 324,n,Festival(Hello World)
;exten => 324,n,Festival(Привет Мир)
exten => 324,n,Hangup()

;Плохо воспроизводит длинный текст
;exten => 325,1,Answer()
;exten => 325,n,Set(ZABBIX-MESSAGE=${FILE(/tmp/zabbix.txt)})
;exten => 325,n,Festival(${ZABBIX-MESSAGE})
;exten => 325,n,Hangup()

;exten => 326,1,Answer()
;exten => 326,n,Wait(3)
;exten => 326,n,Playback(/tmp/zabbix)
;exten => 326,n,Playback(/tmp/zabbix)
;exten => 326,n,Playback(/tmp/zabbix)
;exten => 326,n,Hangup()

;exten => 327,1,Answer()
;exten => 327,n,Set(TOP-CPU=${SHELL(ps ax -o pcpu,command --sort pcpu | cut -d' ' -f3 | tail -n5 | xargs -I CMD basename CMD)})
;exten => 327,n,Festival(${TOP-CPU})
;exten => 327,n,Hangup()

...

exten => _4XX,1,Dial(SIP/${EXTEN},10)

;exten => _4XX,n,Set(MY-CALLED-USER-NAME=${ODBC_SIPFRIENDS_BY_NAME(${EXTEN})})
exten => _4XX,n,Set(MY-CALLED-USER-NAME=${SIPPEER(${EXTEN},callerid_name)})
;exten => _4XX,n,Set(MY-CALLED-USER-NAME=${PJSIP_ENDPOINT(${EXTEN},callerid)})

exten => _4XX,n,Festival(${MY-CALLED-USER-NAME} не может ответить)

server# cat /etc/asterisk/followme.conf

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

server# cat /etc/asterisk/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 show num2login

# cat /etc/asterisk/extensions.conf

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

exten => _4XX,1,Set(MY-LOGIN=${DB(num2login/${EXTEN})})
exten => _4XX,n,ExecIf($[ "${MY-LOGIN}" = "" ]?Set(MY-LOGIN=${EXTEN}))
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 /etc/asterisk/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(MY-REDIRNUM=${DB(redirect/${EXTEN})})
exten => _4XX,n,GotoIf(${MY-REDIRNUM}?CALLMOBILE)
...
exten => _4XX,n(CALLMOBILE),Goto(default,${MY-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},,Tt)
exten => _4XX,n,Verbose(1,Значение DIALSTATUS равно ${DIALSTATUS})
exten => _4XX,n,GotoIf($["${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)

Кузьмин Алексей предложил решение, если требуется только уведомить звонящего

exten => _X.,1,Dial(PJSIP/${EXTEN},60)
exten => _X.,n,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER, BUSY, CHANUNAVAIL, CONGESTION, ANSWER)
exten => s-BUSY,1,Playback(ru/custom/busy_ru)
exten => s-BUSY,n,Hangup()
exten => s-NOANSWER,1,Playback(ru/custom/noanswer_ru)
exten => s-NOANSWER,n,Hangup()
exten => s-CHANUNAVAIL,1,Playback(ru/custom/invalid_ru)
exten => s-CHANUNAVAIL,n,Hangup()

• Asterisk настройка очереди - queues.conf
• Команда Asterisk: Queue

server# cat /etc/asterisk/queues.conf

[general]
persistentmembers = yes

[queue1]
;timeout=10

;для демонстрации на многоканальном телефоне установить call-limit=1 
;member => SIP/403  ; (можно занять звонком на 311)

;member => SIP/user1@mailcgp
;member => SIP/user2@mailcgp
;member => SIP/user3@mailcgp

sever# cat /etc/asterisk/extensions.conf

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

• Asterisk. Конфигурационный файл agents.conf

asterisk# /etc/asterisk/agents.conf

[general]

[my-agents](!)
;musiconhold=silence

[801](my-agents)
fullname=Ivan Ivanovitch Ivanov
[802](my-agents)
fullname=Petr Petrovitch Petrov
[803](my-agents)
fullname=Your name

asterisk# cat /etc/asterisk/extensions.conf

[default]
...
exten => _8XX,1,AgentLogin(${EXTEN})
...
[agents]
exten => _8XX,1,AgentRequest(${EXTEN})
     same => n,Hangup()

asterisk# cat /etc/asterisk/queues.conf

[general]
persistentmembers = yes

[queue1]
strategy = rrmemory
musicclass = default

announce = queue-markq
announce-frequency = 30
announce-position = yes
announce-position-limit = 1

member => Local/801@agents
member => Local/802@agents
member => Local/803@agents

server*CLI> queue show queue1

server*CLI> agent show online

• Asterisk Application_System
• Asterisk cmd System
• System and SHELL
• Пакет sudo

server# cat /etc/asterisk/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
• Сервис HTTP Управление кодировкой
• http://172.16.1.X/asterisk/addrbook.txt

# mkdir /var/www/html/asterisk/

# cat /var/www/html/asterisk/addrbook.txt

Вячеслав Лохтуров<+74951234569>
Вячеслав Лохтуров<+74951234560>
Ваше ФИО<+7ВАШМОБИЛЬНЫЙ>

# cat /usr/local/bin/get-customer-callerid-all.sh

#!/bin/sh

PHONE=$1

ADDR_BOOK=/var/www/html/asterisk/addrbook.txt

grep $PHONE $ADDR_BOOK || echo $PHONE

# chmod +x /usr/local/bin/get-customer-callerid-all.sh

# cat /etc/asterisk/extensions.conf

...
exten => _4XX,n,ExecIf($["${CALLERID(name)}"=""]?Set(CALLERID(all)=${SHELL(/usr/local/bin/get-customer-callerid-all.sh ${CALLERID(num)})}))
...
exten => voip1_00000X,1,Set(CALLERID(all)=${SHELL(/usr/local/bin/get-customer-callerid-all.sh ${CALLERID(num)})})
exten => voip1_00000X,n,Dial(SIP/401&SIP/402&SIP/403&SIP/404)
...

• 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

# chmod +x /usr/share/asterisk/agi-bin/multi_playback.sh

# Демонстрировать лучше построчно
# /usr/share/asterisk/agi-bin/multi_playback.sh

agi_request: multi_playback.sh
agi_escho_kucha_parametrov: ...
agi_arg_1: beep
agi_arg_2: hello-world
agi_arg_3: beep

# rm /tmp/multi_playback.log

# cat /etc/asterisk/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()
...

Решение от Кузьмина Алексея, скрипты взаимоисключающие, один только переводит полное ФИО в фамилию + инициалы, второй еще транслитирует

Место хранения /var/lib/asterisk/agi-bin

Сделать исполняемыми chmod +x

exten => _X.,1,AGI(shortname.sh,${CALLERID(name)},${CALLERID(num)})
exten => _X.,1,AGI(translite.sh,${CALLERID(name)},${CALLERID(num)})

shortname.sh

#!/bin/bash
fio=""
words=($1)
num=$2
m=${#words[@]}
for ((i=0;i<=m-1;i++)); do
    word=${words[$i]}
    if [ $i -gt 0 ]; then
        word=${word:0:1}.
    fi
    if [ $i -eq 0 ]; then
        fio="${word} "
    else
        fio="${fio}${word}"
    fi
done
echo "SET CALLERID \"$fio\"<$num>"

translite.sh

#!/bin/bash
fio=""
words=($1)
num=$2
m=${#words[@]}
for ((i=0;i<=m-1;i++)); do
    word=${words[$i]}
    if [ $i -gt 0 ]; then
        word=${word:0:1}.
    fi
    str=`echo $word | sed "y/абвгдеёзийклмнопрстуфыэ/abvgdeeziyklmnoprstufye/"`
    str=`echo $str | sed "y/АБВГДЕЁЗИЙКЛМНОПРСТУФЫЭ/ABVGDEEZIYKLMNOPRSTUFYE/"`
    str=${str//ж/zh} str=${str//Ж/Zh};
    str=${str//х/kh} str=${str//Х/Kh};
    str=${str//ц/ts} str=${str//Ц/Ts};
    str=${str//ч/ch} str=${str//Ч/Ch};
    str=${str//ш/sh} str=${str//Ш/Sh};
    str=${str//щ/sch} str=${str//Щ/Sch};
    str=${str//ъ/} str=${str//Ъ/};
    str=${str//ь/} str=${str//Ь/};
    str=${str//ю/yu} str=${str//Ю/Yu};
    str=${str//я/ya} str=${str//Я/Ya};
    if [ $i -eq 0 ]; then
        fio="${str} "
    else
        fio="${fio}${str}"
    fi
done
echo "SET CALLERID \"$fio\"<$num>"

• 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
• Программирование диалогов expect Asterisk AMI

# less /etc/asterisk/manager.conf

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

#include "manager.d/*.conf"

# cat /etc/asterisk/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: HelloWorld<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 = *

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

• Asterisk REST API Browser

# curl http://localhost:8088/ari/endpoints?api_key=asterisk:asterisk

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

debian9# curl -sL https://deb.nodesource.com/setup_9.x | sudo -E bash -

# apt install -y nodejs npm

# 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 => 314,1,Answer()
exten => 314,n,Stasis(my-first-app)
exten => 314,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 /etc/asterisk/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
;usetls=no

*CLI> module reload res_xmpp

*CLI> xmpp show connections

*CLI> xmpp show buddies

# cat /etc/asterisk/extensions.conf

[default]
...
exten => 307,1,Answer()
exten => 307,n,JabberSend(aster,userN@corpX.un,Hello World)
exten => 307,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})})

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

• Видео урок: Открытое решение IM и Presence в инфраструктуре Microsoft AD

# apt install ldap-utils

# cat /root/create_exten_mail_cidname_table.sh

#!/bin/sh

IFS=' '

ldapsearch -x -D "cn=Administrator,cn=Users,dc=corp,dc=un" -w 'Pa$$w0rd' -H ldap://ad.corp.un -b "dc=corp,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

# chmod +x /root/create_exten_mail_cidname_table.sh

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

401 user1@corp.un Иван Иванович Иванов
402 user2@corp.un Петр Петрович Петров
403 user3@corp.un Сидор Сидорович Сидоров
404 user4@corp.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=$secret"
        echo "host=dynamic"
        echo mailbox=${exten}'@'$(echo $mail | cut -d '@' -f2 | cut -d'.' -f1)
        echo
done

# chmod +x /root/create_ast_users_conf.sh

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

# cat /etc/asterisk/users.conf

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

# 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.corp.un";
$ADDC="DC=corp,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
405 tpassword405 server.corpX.un 080023F67118 KX-HDVXXX
...
403 tpassword403 server.corp1.un 000E08C190A8 SPA3102
403 tpassword403 server.corp2.un BCC34221709A KX-HDVXXX
403 tpassword403 server.corp3.un 080023F67118 KX-HDVXXX
403 tpassword403 server.corp4.un 000E08C38F1D SPA3102
403 tpassword403 server.corp5.un 000E08C190A2 SPA3102
403 tpassword403 server.corp5.un 000E08C190C2 SPA3102
403 tpassword403 server.corp6.un BCC3422170AA KX-HDVXXX
403 tpassword403 server.corp7.un 080023E421CD KX-HDVXXX
403 tpassword403 server.corp8.un 000E08C190B6 SPA3102
403 tpassword403 server.corp9.un 000E08C190BB 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 /etc/asterisk/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 /etc/asterisk/sip.conf

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

• http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret
• SIP авторизация и MD5

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

server# cat /etc/asterisk/sip.conf

[general]
...
realm=corp
...
[402]
;secret=tpassword402
md5secret=a77b8ed36b4be2c14c2d7e4dfad34b46
...

• 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
• SIPML5 — SIP клиент для браузера на базе WebRTC
• Установка и настройка Asterisk для работы с WebRTC
• https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
• Настройка WebRTC в Asterisk 13

• !!! 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

• DoubangoTelecom/sipml5

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

• Распознавание речи в Asterisk с использованием Yandex SpeechKit HTTP API

• Голосовые сервисы помогут голосовой почте

Сотрудник звонит на внешний номер, номер не отвечает или занят.

Внешний номер перезванивает на внешний номер организации (с которого у него пропущенный), где по умолчанию его должен встретить IVR.

Если в течении последних Х минут в cdr есть запись звонка на этот внешний номер с состоянием номер не отвечает или занят, то соединить с внутренним номер с которого ему звонили.

ДОП.: Если при этом в cdr есть запись что с этим внешним номером звонок состоялся(по времени после не отвеченного, направление не важно, внешний номер сам перезвонил и дозвонился или до него дозвонились со второго раза), то переключить на IVR.

 

Решение собственное, если есть что улучшить буду рад.

Адаптировано под вашу методичку, но лучше проверить.

 

/etc/asterisk/func_odbc.conf

 

[NOTANSWERED_BY_CALLERID]

dsn=asterisk

readsql=SELECT cnum FROM cdr WHERE dst='${ARG1}' AND disposition<>'ANSWERED' AND (calldate >= (SELECT COALESCE(MAX(calldate), now() - interval ${ARG2} minute) AS calldate FROM cdr WHERE (dst='${ARG1}' OR src='${ARG1}') AND disposition='ANSWERED' AND (calldate >= now() - interval ${ARG2} minute) ORDER BY calldate DESC LIMIT 1)) ORDER BY calldate DESC LIMIT 1;

 

/etc/asterisk/extensions.conf

 

exten => voip1_00000X,1,Set(NOTANSWERED=${ODBC_NOTANSWERED_BY_CALLERID(${CALLERID(num)},5)}) ; External number, # minutes to search not answered call

same => n,GotoIf($[${LEN(${NOTANSWERED})} != 0]?recall)

same => n,Goto(ivr,s,1)

same => n(recall),Set(CALLERID(name)=RECALL ${CALLERID(name)})

same => n,Dial(PJSIP/${NOTANSWERED},60)

 

Вызывается функция ODBC с передачей ей номера звонящего и количества минут за которые искать записи.

В SQL запросе вначале выполняется поиск за указанный интервал в минутах отвеченных звонков с участием внешнего номера и возвращается дата самого последнего звонка, если он есть, либо дата начала указанного интервала поиска (now() – 5 минут, например) и после этого выполняется поиск не отвеченных звонков на указанный внешний номер за указанный интервал.

После этого к CALLERID(name) звонящего добавляется RECALL, чтобы уведомить сотрудника, что ему перезванивают, и осуществляется соединение с номером сотрудника от которого был последний не отвеченный звонок, или переключение на IVR, если не отвеченных звонков не было.

 

Алексей Михайлович Кузьмин

#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