The SIP protocol is widely used today in VoIP telephony. The so-called SIP-providers that make it possible to make calls at favorable rates enjoy popularity among the users.
It is only half the battle to enter into a contract with an operator and get his credentials. Although the default SIP support is built into Asterisk, you need to set up incoming calls for asterisk.
To do this, you should open the file /etc/asterisk/sip.conf, and set definition in the format "register => user [: password [: name]] @host [: port] [/ extension]"in the "General" section register. Let us consider each of the options:
- User is the account number through which the SIP-server authenticates the user.
- Password is the password the user will use to pass authentication.
- Name - имя пользователя, используемое для идентификации SIP-сервером (необязательно).
- Name is the user name used to identify the SIP-server (optional).
- Host is the domain name of the SIP-server.
- Port is the port number to log on to host (the default value is 5060).
- Extension is the extension number to receive calls to your goip asterisk version.
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You need to specify settings for each SIP server, by pre-creating the section for each server in /etc/asterisk/sip.conf.
Outgoing calls must be configured via the extra digit in /etc/asterisk/extensions.conf file. It is necessary to register a line of this type: exten => _5, 1, Dial (SIP / $ {EXTEN: 1} @ operator, 30, r), where _5 is an additional figure. If you have multiple SIP-numbers, in order to save money, outgoing calls can be divided into groups by area code.
To configure incoming calls, you need to take of the existing SIP-number in the context in the format [name-in], and outgoing calls in the format of [name-out]. Next, you need to register the whole algorithm of the call, including the inner room, getting a call in the queue, voice message in the case of "non-response", an offer to leave a message, send a message to the voicemail box, a farewell to the subscriber and call completion.
To independently configure Asterisk is not an easy task for a beginner in the VoIP-technology. This procedure requires knowledge of specific technical nuances. It is better to seek professional help from professionals who have the appropriate qualifications.
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