04-Oct-2016 08:39

Setting up Incoming/Outgoing Calls in the Asterisk PBX

ejointech

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.
Are you interested in VoIP technology? Are you looking for a reliable start-up in the telecommunications sector? You will be interested in the opportunity to start a GSM termination business . You can get the maximum profit making the minimal investment! We offer a turnkey GoAntiFraud solution for beginners, which includes opportunities for efficient VoIP termination, as well as a set of equipment by GoIP, EjoinTech & China Skyline at low cost.

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.

GoAntiFraud offers you to start a profitable GSM termination business! If you are interested in VoIP technology, we will help you start your own business, yielding a stable income. By purchasing our comprehensive New Business package , you will start making money immediately! We will give you full technical support at all stages of business.
business
04-Oct-2016 08:39 Setting up Incoming/Outgoing Calls in the Asterisk PBX

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.
Are you interested in VoIP technology? Are you looking for a reliable start-up in the telecommunications sector? You will be interested in the opportunity to start a GSM termination business . You can get the maximum profit making the minimal investment! We offer a turnkey GoAntiFraud solution for beginners, which includes opportunities for efficient VoIP termination, as well as a set of equipment by GoIP, EjoinTech & China Skyline at low cost.

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.

GoAntiFraud offers you to start a profitable GSM termination business! If you are interested in VoIP technology, we will help you start your own business, yielding a stable income. By purchasing our comprehensive New Business package , you will start making money immediately! We will give you full technical support at all stages of business.
Setting up Incoming/Outgoing Calls in the Asterisk PBX thumb.png Author2 04-10-2016 Setting up Incoming/Outgoing Calls in the Asterisk PBX

Related Articles

GoAntiFraud

Repeaters to Amplify GSM Signals

02 Aug 2016 13:42
It is known that the terminator should provide a high quality of the route to sell it profitably. The route will be in demand among the originators only if the ACD and ASR performance are up to par. These figures describe the number of successful calls, that is, those that last for more than 0 secon...
GoAntiFraud

Start of Work With VoIP Equipment

22 Aug 2016 15:08
Beginners, who are familiarizing with VoIP technologies, do not yet know many nuances of the organization of the Internet telephony. Some of them are willing to save and connect the Huawei 3G modem of series E1550, E160g, E173, and E1752C. But, according to experts, it is better to install the gatew...
GoAntiFraud

Subtleties of VoIP-gateway Selection

06 Jun 2016 11:29
Those who are engaged in GSM voice traffic termination face the challenge of VoIP-gateway selection. If you do not approach the issue wisely, the equipment you have purchased can negate all your efforts on business development. We consider the basic criteria which you should look for when choosing t...
GoAntiFraud

The Methods of Organizing an Inexpensive VoIP Call-center

08 Aug 2016 11:24
Many business owners are faced with the problem of the expansion of telephone lines, the need to increase the communication load on the network and other difficulties in expanding the staff and hiring call-center operators. Most often, business owners solve their problems by installing a PBX that...
GoAntiFraud

Smart VoIP-phones

06 May 2016 09:18
VoIP-telephony has entered the world of modern communication and taken its rightful place. Today, it is impossible to imagine a means of communication without connecting the computer-Internet-phone together. VoIP can transform an ordinary audio format into a digital code transmitted via the Inter...
GoAntiFraud

5 Advantages of GoIP Termination Equipment

11 May 2017 15:17
Traffic termination is one of the most profitable businesses in the telecommunications sector. Your monthly income can be over 2 thousand dollars, with you making minimal monetary investments and time costs. Primarily, the number of channels in the work affects the level of earnings of those who are...
GoAntiFraud

The Purchase of Equipment to Start a GSM Termination Business

05 May 2017 15:16
How to start your business in the VoIP termination market? The first thing you need is to purchase specialized equipment - GSM gateways, which convert a VoIP signal to GSM format. The more gateways you install, the more channels will be in operation. The number of channels, in turn, depends on the e...

GoAntiFraud

GoAntiFraud
GoAntiFraud
GoAntiFraud
GoAntiFraud
GoAntiFraud
GoAntiFraud
GoAntiFraud
GoAntiFraud
GoAntiFraud
GoAntiFraud GoAntiFraud GoAntiFraud GoAntiFraud GoAntiFraud GoAntiFraud GoAntiFraud GoAntiFraud GoAntiFraud