Differences

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--- сервис_asterisk [2021/11/17 11:25]
val [Настройка RES_PJSIP каналов]
+++ сервис_asterisk [2023/12/02 08:15]
val [Установка]
@@ Line 8: / Line 8: @@
 ===== Установка =====
+  * [[https://community.asterisk.org/t/install-asterisk-via-apt-or-from-source/99713|Install Asterisk via apt or from source?]]
 ==== Debian/Ubuntu ====
 <code>
@@ Line 38: / Line 39: @@
 </code><code>
 [general]
-;udpbindaddr=0.0.0.0:5061
+;udpbindaddr=0.0.0.0:6050
 transport=udp
 disallow=all
@@ Line 45: / Line 46: @@
 directmedia=no           ; for webinar
 nat=force_rport,comedia  ; for webinar
+qualify=yes              ; for webinar
 ;context=default
@@ Line 72: / Line 74: @@
 ;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
+;directmedia=no
+;nat=force_rport,comedia
+;qualify=yes
 ;[user2]
@@ Line 93: / Line 93: @@
 ==== Использование шаблонов в именах CHAN_SIP каналов ====
 <code>
-root@server# cat /etc/asterisk/sip.conf
+server# cat /etc/asterisk/sip.conf
 </code><code>
 ...
-;[office](!)
+[office](!)
-;type=friend
+type=friend
-;host=dynamic
+host=dynamic
-;deny=0.0.0.0/0.0.0.0
+deny=0.0.0.0/0.0.0.0
-;permit=192.168.X.0/255.255.255.0
+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)
+[403](office)
-;secret=tpassword403
+secret=tpassword403
-;[404](office)
+[404](office)
-;secret=tpassword404
+secret=tpassword404
 ...
 </code>
@@ Line 224: / Line 239: @@
 </code><code>
 ...
-; voip1.un и server.corp.un одна и та же система
+; voip1.un и server.corp.un (курс CGP) одна и та же система
 ; приходится подключаться к ней с разных портов :)
@@ Line 231: / Line 246: @@
 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
@@ Line 343: / Line 359: @@
 </code><code>
 [general]
+...
+disallow=all
+allow=alaw
 ...
 allow=h263
@@ Line 352: / Line 371: @@
 ==== Отладка работы Asterisk ====
+  * [[https://community.asterisk.org/t/help-new-installation-full-of-errors/46614|HELP - New Installation Full of Errors]]
 === Сервисные сообщения ===
@@ Line 389: / Line 410: @@
 ==== TimeBased контексты ====
+  * [[https://wiki.asterisk.org/wiki/display/AST/Contexts%2C+Extensions%2C+and+Priorities#Contexts,Extensions,andPriorities-Dialplansearchorder|Dialplan search order]]
   * [[https://www.voip-info.org/asterisk-tips-openhours/|How to include contexts based on time and date]]
@@ Line 653: / Line 676: @@
 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"
@@ Line 819: / Line 844: @@
 <code>
 http://172.16.1.X/asterisk-cdr-viewer/
+</code>
+  * Спасибо Комлеву Александру Павловичу!
+<code>
+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";
 </code>
 ===== Asterisk realtime =====
@@ Line 1573: / Line 1612: @@
 [general]
 ...
-;featuredigittimeout = 1000
+featuredigittimeout = 3000
 ...
 [featuremap]
@@ Line 1663: / Line 1702: @@
 ...
 pickupexten = *88
-...
-</code><code>
-server# cat /etc/asterisk/extensions.conf
-</code><code>
-...
-exten => _8XXXXXXXXXX,1,Dial(SIP/voip1_00000X/${EXTEN},,T)
-...
-exten => voip1_00000X,1,Dial(SIP/401,,t)
 ...
 </code><code>
@@ Line 1697: / Line 1728: @@
 include => parkedcalls
+...
+exten => _8XXXXXXXXXX,1,Dial(SIP/voip1_00000X/${EXTEN},,T)
+...
+exten => voip1_00000X,1,Dial(SIP/401,,t)
 ...
 </code>
@@ Line 1771: / Line 1806: @@
 ...
-;For IMAP/SMTP
+;;;For IMAP/SMTP
 ;charset=UTF-8
 ;locale=ru_RU.UTF-8
@@ Line 1780: / Line 1815: @@
 ...
-;For IMAP
+;;;For IMAP
-;debian# apt install asterisk-voicemail-imapstorage
+;;;debian# apt install asterisk-voicemail-imapstorage
-;debian# cp /etc/dovecot/private/dovecot.pem /usr/local/share/ca-certificates/dovecot.crt
+;;;debian# cp /etc/dovecot/private/dovecot.pem /usr/local/share/ca-certificates/dovecot.crt
-;debian# update-ca-certificates
+;;;debian# update-ca-certificates
-;debian# service asterisk restart
+;;;debian# service asterisk restart
 ;imapserver=server.corpX.un
 ;imapfolder=INBOX
-;For SMTP
+;;;For SMTP
 ;debian# apt install postfix mailutils
@@ Line 1913: / Line 1948: @@
 <code>
-server# cat 1.call
+# cat call.txt
 </code><code>
-Channel: sip/401
+Channel: Local/voip1_00000X@default
-Extension: 601
+Extension: 89161234567
 Context: default
-CallerId: Conferencia <601>
+CallerId: CorpX<84992636363>
 </code><code>
-server# cat 2.call
+server# grep astspooldir /etc/asterisk/asterisk.conf
-</code><code>
-Channel: sip/402
+server# cp call.txt /var/spool/asterisk/outgoing/
-Extension: 601
-Context: default
+server# ls /var/spool/asterisk/outgoing_done
-CallerId: Conferencia <601>
-</code><code>
-server# cat 3.call
-</code><code>
-Channel: sip/403
-Extension: 601
-Context: default
-CallerId: Conferencia <601>
-</code><code>
-server# cp [123].call /var/spool/asterisk/outgoing/
 </code><code>
 # cat /usr/share/originate_call_file.sh
@@ Line 2025: / Line 2050: @@
 ;[menu2]
 ;exten => s,1,Background(silence/8)
-;exten => s,1,WaitExten(8)    ; работает только после Answer или Background
+;exten => s,n,WaitExten(8)    ; работает только после Answer или Background
 ;exten => _4XX,1,Goto(default,${EXTEN},1)
 ;exten => t,1,Goto(menu,s,1)  ; timeout exceeded, работает только с WaitExten
@@ Line 2257: / Line 2282: @@
 ;number=>89162323232,30
 context=>default
+;context=>to-pstn
 [401]
 number=>89162323232,30
 ;number=>89163434345,30
 context=>default
+;context=>to-pstn
 </code><code>
 server# cat /etc/asterisk/extensions.conf
@@ Line 2420: / Line 2447: @@
 [queue1]
-;для демонстрации на многоканальном телефоне установить call-limit=1
+;timeout=10
-;member => SIP/401  ; в вебинаре (занять можно звонком на 311)
-;member => SIP/403  ; в классе
+;для демонстрации на многоканальном телефоне установить call-limit=1
+;member => SIP/403  ; (можно занять звонком на 311)
+;member => SIP/user1@mailcgp
+;member => SIP/user2@mailcgp
+;member => SIP/user3@mailcgp
 </code><code>
 sever# cat /etc/asterisk/extensions.conf
@@ Line 2473: / Line 2505: @@
 strategy = rrmemory
 musicclass = default
-timeout=10
 announce = queue-markq
@@ Line 2483: / Line 2514: @@
 member => Local/802@agents
 member => Local/803@agents
-;member => SIP/user1@mailcgp
-;member => SIP/user2@mailcgp
-;member => SIP/user3@mailcgp
 </code><code>
 server*CLI> queue show queue1
@@ Line 2511: / Line 2538: @@
 ...
 </code>
-==== Приложение SHELL ====
+==== Функция SHELL ====
   * !!! Название функции только ЗАГЛАВНЫМИ буквами
@@ Line 2806: / Line 2833: @@
 exten => _4XX,n,Hangup()
-exten => _4XX,n(busy),Playback(tt-allbusy)
+exten => _4XX,n(busy),JabberSend(aster,${MY-CALLED-USER-ID},Вам звонил ${CALLERID(all)})
+exten => _4XX,n,Playback(tt-allbusy)
 exten => _4XX,n,Hangup()
 ...
@@ Line 3114: / Line 3143: @@
 ==== Использование хешей паролей для каналов типа user ====
-[[http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret]]
+  * [[http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret]]
+  * [[http://asterisk-support.ru/forum/topics/3032/|SIP авторизация и MD5]]
 <code>
@@ Line 3127: / Line 3157: @@
 [402]
 ;secret=tpassword402
-md5secret=29c91aa064cf462f5146b7ee81e7ba49
+md5secret=a77b8ed36b4be2c14c2d7e4dfad34b46
 ...
 </code>

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