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# service asterisk start
# systemctl disable fail2ban # systemctl mask fail2ban # init 6
!!! Для работы PJSIP необходимо обновить модули (лучше все, занимает, примерно 1 час, некоторые модули обновляются после нескольких итераций, для PJSIP не обязательно)
Admin->Updates->Module Updates Check Online, Download all, Upgrade all, Process
Application-Extensions Add Extensions -> PJSIP User Extension: 402 Display Name: Petr Petrov Secret: tpassword402 Link to a Default User: Create New User Username: user2 Use Custom Username Password For New User: password2 Connectivity -> Trunks Add Trunk -> Add...pjsip... Trunk Name: PSTN Outbound CallerID: 8495NNNNNNN Username: 00000X Secret: spasswordX SIP Server: voip1.un Contact User: 8495NNNNNNN From Domain: voip1.un From User: 00000X Connectivity -> Outbound Routes Route Name: ToPSTN Trunk Sequence for Matched Routes: PSTN Dial Patterns: 8XXXXXXXXXX Connectivity -> Inbound Routes Description: FromPSTN DID Number: 8495NNNNNNN Set Destination: Ext 403
Please provide the core credentials that will be used to administer your system
Username: admin Password: Pa$$w0rd Admin Email address: userX@isp.un
UPDATE `ampusers` SET `password_sha1` = SHA1('12345678') WHERE `username`='admin';
Не активировать, отказаться от SIPStation
Желательно отключить f2ban для локальной сети
Admin->System Admin->Intrusion Detection->Whitelist->172.16.1.0/24
Settings->Asterisk SIP Settings или NAT: no IP Configuration: Public IP или External IP: 172.16.1.X Local Networks: 192.168.1.0/255.255.255.0 Allow SIP Guests: no
В версии 1013 не получается менять имя пользователя при создании канала
Applications->Extensions->Generic CHAN SIP devices Submit User Extension: 401 Display Name: Ivanov Ivan Ivanovitch Device Options secret: tpassword401
Admin->Asterisk CLI Reports->Asterisk Log Files
Удерживая клавишу CTRL
Admin -> Administrator Username: admin2 Password: password2 Admin Access Application->Extensions Apply Changes Bar Add Extensions
Connectivity->Trunks->Add SIP Trunk
Connectivity->Trunks->Add SIP Trunk General Settings Trunk Name: Voip1 00000X Outbound CallerID: 89166071103 Outgoing Settings Trunk Name: voip1_00000X PEER Details: host=voip1.un defaultuser=00000X fromuser=00000X fromdomain=voip1.un secret=spasswordX type=peer
Connectivity->Outbound Routes->Add Route Route Name: Call_To_PSTN Dial Patterns that will use this Route match pattern 89XXXXXXXXX match pattern 8495XXXXXXX match pattern 8499XXXXXXX Trunk Sequence for Matched Routes 0: voip1 00000X
Connectivity->Trunks->Edit SIP Trunk PEER Details: nat=no directmedia=no insecure=invite callbackextension=voip1_00000X
Application->Ring Groups->Add Ring Group Group Description: All Phones Ring Strategy: ringall Extension List: 401 403 Destination if no answer: Terminate Call Hangup
Connectivity->Inbound Routes->Add Incoming Route Description: From voip1 00000X DID Number: voip1_00000X Set Destination: Ring Groups: All Phones
можно использовать цепочки групп используя атрибут групп “Destination if no answer”
Admin->Feature Codes
Admin->Feature Codes In-Call Asterisk Attended Transfer: *2 In-Call Asterisk Blind Transfer ## Setting->General Setting->Asterisk Dial command options: Tt...
Application->Extension->4XX callgroup: 1 pickupgroup: 1
Admin->User Management->Ivanov Ivan Ivanovitch Login Name: user1 Password: password1 Linked Extension: 401
Settings->Asterisk SIP Settings->Chan SIP Language: ru
# cat /etc/asterisk/sip_general_additional.conf
...
# cat /etc/asterisk/sip_additional.conf
...
# cat /etc/asterisk/extensions_additional.conf
...
# cat /etc/asterisk/sip_custom.conf
language=ru
# cat /etc/asterisk/extensions_custom.conf
exten => 301,1,Answer() exten => 301,n,Playback(hello-world) exten => 301,n,SayDigits(X) exten => 301,n,Hangup() [from101] ; no need, use DAHDI Channel exten => s,1,Dial(DAHDI/1,20) exten => s,n,Hangup()