SIP server is a set of software for running IP telephony within an office or production. Traditional telephony is characterized by high call costs and does not offer any particular business benefits. Deploying your own production or office PBX makes it possible to customize the distribution of calls, reduce the cost of communication within the company and establish voice communication with customers.
It is not difficult to choose an IP telephony server - in our review you will find solutions for Windows and Linux. But they are increasingly being replaced by ready-made solutions from providers. In addition, the prices for launching office telephony are cheap. The client only has to choose a tariff, pay for communication services, connect the equipment to the network and make all the necessary settings.
Before us is one of the most popular SIP servers in the world for organizing office telephony. The project appeared in 1999 and was intended to replace expensive mini-automatic telephone exchanges. The server runs under the Linux operating system, has all the necessary functionality:
- Supports work with traditional telephony.
- Knows how to manage the distribution, processing of telephone calls.
- Supports video sessions.
- Can be integrated into CRM systems.
- Supports encrypted calls to prevent eavesdropping.
The functionality of the Asterisk SIP server can be expanded with additional software. It works with almost any IP telephony protocols and can solve even the most complex problems. Its main drawback is complexity. Convenient Web-based interfaces have been developed to manage the server, but they do not solve the problem of the complexity of this software product.
Server from 3CX
The 3CX Phone System SIP server is designed to provide telephony services to businesses of any size. These can be small firms or large corporations with dozens of branches, divisions and divisions. It supports the full functionality of office PBXs - work with calls, integration into CRM, conference calls, call center functions and much more. The product is notable for its comprehensive developer support. Working environment - Windows operating system. Implementing your own developments, as in Asterisk, will not work due to the closed source code of the server.
SipXecs Server
Another software PBX for solving business problems. It lacks support for many protocols, it only works with SIP. A web interface is used to control telephony. There is support for most of the standard functions - transfer / processing of calls, fast dialing, conferences, hold and wait, multi-channel communication and much more. The server runs under Linux operating system.
OfficeSIP Server
Free application for organizing office telephone communications. Suitable for small and medium-sized offices that do not require additional functions. For large enterprises with divisions and branches around the world, this SIP server is not suitable. But to connect the accounting department, directors, personnel department, several offices with intercity and international communication is always welcome.
The server runs under the Windows operating system and does not create any difficulties. It is free even for business customers, which determines some demand for this product. Installation takes place quickly and without delays, registration of new subscribers is done in a couple of mouse clicks. If the task is to set up a connection with your own hands, but you do not have much experience, use this simple and free solution.
Ready-made solutions from providers
Recently, the business has switched to ready-made solutions. There are several reasons for this:
- Reduced costs - connection is often free, only long distance costs, jobs and some additional functions are paid.
- Security - self-configuring VoIP in the office will not give you confidence in the security of the system from hacks and attacks. Providers have certified personnel.
- Convenience - only computers and telephones are needed from additional equipment. No separate hardware for IP servers.
Let's consider several solutions for organizing IP telephony for business.
Cloud PBX from Zadarma
This provider connects office telephony at prices from 10 kop / min, with premium voice quality. The system administrator of your office does not have to fiddle with the equipment - it is enough to add subscribers to the system and set up the distribution of calls. Benefits of Zadarma:
- Free connection to IP telephony.
- The provider offers multichannel numbers in 90 countries of the world and in many Russian cities.
- The ability to integrate with the used CRM.
- Full cloud PBX functionality.
- Free calls within the company and its branches, regardless of the geographic location of the workplace.
- Access to 8-800 numbers with the functionality of a full-fledged call-center.
- API interface for implementing your own business tasks.
The provider guarantees high quality of voice transmission, supports clients by phone or through internal chat, offers inexpensive calls within Russia and around the world. And all this without expensive equipment and settings. Order the service and get a ready-made cloud PBX in 5 minutes. The configuration is carried out through a convenient web interface.
As customer reviews show, the Zadarma provider provides high-quality voice transmission and full-fledged office PBX functionality for large enterprises and small businesses.
Cloud PBX from SIPNET
One of the oldest IP telephony providers. He works not only with individuals, but also with corporate clients. The starting rate will cost only 1000 rubles. It will include three telephonic workplaces, a package of minutes to choose from (from 600 to 1500 minutes to numbers in Moscow and St. Petersburg, throughout Russia or to mobiles). There is no connection fee. Also, customers have access to options that expand the functionality, the number of seats and provide the services of a personal manager. SIPNET is a full-fledged PBX for business, including with call-center functions.
SIP telephony can significantly reduce the cost of telephony. By using the services of IP-providers, we save money and get the opportunity to call on reduced tariff plans from anywhere in the world. This type of communication is also used to organize intra-office telephony - for this you need to install a SIP server on one of the computers and connect software and hardware phones to it. In this review, we will compare the most popular SIP servers, including free ones:
- Asterisk;
- Kamailio;
- OfficeSIP Server;
- sipX.
Let's take a closer look at these servers and find out how to start a SIP server with your own hands.
We will begin this review by considering one of the most famous servers for IP telephony - the Asterisk SIP server. It is focused on organizing office telephony and is very popular.
SIP Server Asterisk
Asterisk can be called a free solution, but it does have licensed modules. The program works in Linux operating systems and is released in the form of several distributions, differing in functionality, web interfaces and sets of additional modules. This is not to say that this is a solution for novice users. - rather, it is a more professional solution. SIP server Asterisk is endowed with the following features:
- Call forwarding and transfer;
- Call holding and waiting (with background music);
- Call pickup and parking (functions allow you to answer calls from other devices or continue conversations on them, started on other devices);
- Conference calls;
- Video communication;
- Call center functions;
- Integration of traditional telephone lines;
- Administration via the web interface;
- Billing functions.
We can say that using the Asterisk SIP server will allow you to solve a problem of any complexity. Scalability, availability of additional modules, a huge number of supported protocols - all this can be called the advantages of the program. As for the disadvantages, it is the complexity in the settings for novice users and the presence of a dual license.
Despite the fact that this server is free, it may contain modules distributed based on the licensed code - sometimes this causes some problems.
SIP server Kamailio
This project was once referred to as the OpenSER SIP server, but in 2008 it was renamed Kamailio. But he is not the most famous when compared to such monsters as 3CX or Asterisk. The server has decent functionality and is most often used in a professional environment. therefore it is not suitable for solving simple problems.
In the list of its advantages, we can include support for a large number of various modules that extend its functionality. The disadvantages included the complexity of the setup.
SIP server sipX
This is another free product that runs on Linux systems. The sipX server is simple and office-oriented. The developers have endowed it with decent functionality, providing a large number of functions for managing voice calls. When using the right equipment, the sipX SIP server allows you to solve even the most complex tasks.
Its strengths include stability, simplicity and minimal size. SipX allows you to deploy local SIP networks in a matter of hours, which is used for fast telephony in offices. Also, this server is free. As for the disadvantages, the most negative point is that for all functions to work, you need advanced phones and VoIP gateways.
SIP servers for Windows
Linux systems are extremely robust and performant. But they require a certain amount of knowledge, and they cannot be called friendly to ordinary users. Therefore, in the world of software, there are more understandable SIP servers for Windows. Of course, here too, users and system administrators can face various difficulties, but it is much easier to get around them.
SIP server 3CX
Among the most advanced SIP servers, we can highlight the Voip PBX 3CX Phone System for Windows. This solution is designed for organizing corporate communications of any scale, even if separate offices are located on different parts of the planet. Server advantages:
- Full voice functionality;
- Support for a large number of clients (including our own software for various platforms);
- Support for web conferencing;
- Integration of services of third-party SIP providers and traditional telephony operators.
Using the 3CX Phone System Server allows you to minimize communication costs and make office telephony more convenient... The developer provides users with a lot of training materials, conducts training events, provides comprehensive user support. Customers can choose from a standard free version, as well as a commercial version that supports additional functions.
The free trial version is quite functional and can be used as a basic option for organizing IP telephony.
This product has many advantages. First of all, you need to highlight the fact that the 3CX Phone System server runs under the Windows operating system. It is extremely flexible in settings and has great functionality. If you need regular telephony, and not a whole call center, then the free version will be enough for you. Disadvantages - it is impossible to supplement the system with something of your own, since the source code is closed. However, this cannot be considered a significant drawback.
SIP Server OfficeSIP Server
Free SIP Server OfficeSIP Server is free software for Windows. This server is so simple that even the most inexperienced user can handle its installation and configuration. Installation and launch of the program takes a couple of minutes, after which you can start creating local user accounts.
Also it is possible to connect to a third party IP provider for calls around the world... An excellent program for small offices that need office telephony. Benefits of the program:
- Ease of settings;
- Work in Windows environment;
- Ease of connecting new subscribers;
- Communication with the outside world.
Disadvantages of the program:
- Lack of many convenient office and voice functions;
- Impossibility of scaling;
- There is no possibility of connecting to "your" PBX from anywhere in the world (only local connections).
However, it is an extremely affordable and free SIP server for small offices.
Corporate use of SIP-numbers often takes place under the Windows OS installed on most office PCs. Consider the existing VoIP solutions for this system.
Web calls: what, how, where
Approved by top management
Senior management software is usually designed to minimize the difference between online conferencing and physical proximity at the meeting table as much as possible. It is over the elimination of the boundaries of the virtual and real worlds that the developers of Silicon Valley are fighting, hoping to functionally "surpass" the Asterisk server.
- B-Force. Developed by the company of the same name in 2010, and has been improving daily since then. Users of the Russian-language Wikipedia position the program as one of the few that are suitable for security requirements even for use in government agencies.
- 3CX Phone is multiplatform, can be used "in conjunction" not only with Windows, but also with linkusoids, as well as under mobile OS - Android, i- / Mac-OS, etc. All possibilities are available to subscribers free of charge, which is surprising given the quality of services. the work of technical support and the convenience of the interface. The latter, by the way, is recognized (according to the results of Software Advice research) as the leader of the TOP-5 most comfortable sipphones to use.
- Brosix. One of the most secure programs, working according to the US federal standard under 256-AES symmetric encryption. Corporations preferring to use Brosix Business will have to pay for a license in exchange for the ability to create private crypto-resistant networks with the click of a few buttons. Individuals can legally use the program for free, but in the light version, which does not have the functions of a "whiteboard", exchange of desktops and conference calls.
Comfort of clerks is the key to stable work of the company
But quality communication is needed not only by the management, but also by ordinary “white collars”. No matter how routine the work of clerks is, it is on it that the company's activities are based, and therefore it is in the interests of management to simplify their actions as much as possible. Many offices make do with the functionality of such programs as Skype, Yahoo! Messenger and the like for internal communication, but in some cases, the best solution would be to use special software.
- Call Office. "Sharpened" for work with large client bases. Simplifies calling, sending messages to (e-mail / SMS) and other mass notifications as much as possible.
- Ventrilo. Softphone walkie-talkie associated with gamer voice chats. Despite the stereotypes, it is popular in companies where profit depends on the speed of reaction and dynamics - for example, in delivery services or closed offline exchanges.
- Sippoint. A utility that supports a multi-user interface and allows you to configure multi-level contact databases. In addition, users can exchange files on a closed intra-office network. It is notable for the fact that it easily ports data from / to other systems - Google Talk, QIP and other popular instant messengers.
- Jabbin. The main advantage of the softphone is the ability to make calls even without a SIP provider connection, only with a custom web connection, including local intra-connections. But at the same time, alas, there is no way to call a landline or mobile number.
The best softphone for the best subscribers
Subscribers of the site will not have to be tormented by a choice dilemma: there is a universal and at the same time simple program available to all users - a telephone IP server - YouMagic Softphone. In addition to the obvious advantages of working with the provider itself, the subscriber will receive the following "bonuses":
- virtual PBX with protection against flooding, spam and DDoS attacks on central nodes, which guarantees comfortable communication without interruptions;
- the technical support service will answer any question in detail, and in case of problems - will promptly solve them;
- each softphone user will be able to use several accounts and financial calculators for each of them, thereby automating the accounting of traffic costs.
These and many other features make the use as comfortable as possible on any platform, including Windows, Android and other operating systems.
SIP telephony server service
Including this service, you get the opportunity to use sIP telephony server (PBX) based on Asterisk inside your home network.
You can register your smartphone or computer with a SIP client in this telephone exchange and call your relatives and friends who are also registered in this server.
Hint! In addition, you can configure your SIP telephony server using this instruction
Example of use and settings
Everything is very simple.
1. On the applications page, you need activate the SIP telephony server service, which will act as a single point of registration for your smartphones, computers and other devices using the SIP protocol. This server will switch your phone calls within your WAN.
The server address in your network is 172.16.255.14
After starting the server, check its availability by running the command ping 172.16.255.14
2. If the ping was successful, then register your devices. To do this, on our website, enter the desired phone number and password for this device, and then configure your device, as shown in the example below.
2.1. On the SIP telephony server page, specify the desired phone numbers and passwords for connection.
In this example, two phone numbers are specified - 10 and 11 with a password of 1111 each.
2.2. Customize your device. This example shows two connection implementations - a standard SIP client of Android OS and using the Zoiper application located on a PC with Windows 8
So Android. It has a built-in SIP telephony client.
First, create a new SIP server account
We indicate the previously selected phone number with which we will register on the SIP telephony server (in our example 10), password and server address
After you save the account, the phone will try to register with the SIP server.
There are also various settings, and most likely you will need to select "Accept incoming" to keep the phone connected to the SIP server and wait for an incoming call. Actually, that's all.
Now let's get in touch with the person we are going to call through our SIP telephony server. To do this, go to the notebook and add a new Contact , which we will call "Dacha". But there is a nuance ... we need to indicate the number of the "Dacha" and this must be done in the field called "Call via the Internet".
This field is absent on the main contact screen, so you need to scroll down to the item "Add another field" and then a new window will open with a choice of fields, among which there will be "Call via the Internet"
Now the last thing is left - to indicate the number in this field. It is indicated as shown in the figure below -
At this point, our client on Android is ready. Let's add the settings on the second side of our future telephone connection.
2.2 As the second party, we will have Windows 8 PC with Zoiper SIP telephony client installed.
After installation, enter the settings and add a new account with the SIP protocol.
In your account settings, enter your username and server address in this format: This email address is being protected from spambots. You need JavaScript enabled to view it. .4 and password. Then check the box " Skip auto detection"
After saving the settings, go to the settings again and click the Register button. A status entry should be displayed in the right corner - Registred.
If everything is registered successfully, then you can try and call. Close this window. From the home screen select Dialpad and dial the Android number - 10.
We hope your Android has rang and you can check the quality of the connection.
That's all, actually.
Technical features
Your SIP telephony server located at 172.16.255.14 is only a SIP server and no longer contains any other data other than the numbers you entered.
Service testing period
We plan that the testing period for the SIP telephony server service will take about a month.
Service cancellation
You can cancel the service at any time. In this case, the registrations of your devices will be deleted, and the SIP telephony server is stopped.
Despite the development of various information exchange systems, such as e-mail and instant messaging services, the ordinary telephone will remain the most popular means of communication for a long time. A key event in the history of telecommunications and the Internet was the emergence of voice over IP networks, which is why the very concept of a telephone has changed in recent years. The use of VoIP is modern, convenient, cheap, since it is possible to combine remote offices without even resorting to the services of telephone operators. What other reasons do you need to raise your IP telephony server?
Project Asterisk
Asterisk is present in the package repositories of most distributions. So, in Ubuntu the command sudo apt-cache search Asterisk produces a decent list of packages, after installing which you can immediately start configuring. But installing from the repository has one drawback - as a rule, it contains the version Asterisk well behind the current one, which can be downloaded from the official site. Therefore, we will consider a universal installation method using the example of the same Ubuntu, although everything that has been said (with rare exceptions) applies to other distributions.
Install the packages required for compilation:
$ sudo apt-get install build-essential automake
autoconf bison flex libtool libncurses5-dev libssl-dev
In addition, it is highly recommended to install the libpri library even if you do not need support for Primary Rate ISDN (Primary Integrated Services Digital Network). This can be done either through the repository: sudo apt-get install libpri1.2, or using the sources:
$ wget -c downloads.digium.com/pub/libpri/libpri-1.4-current.tar.gz
Compilation of the library is standard, so we will not dwell on this.
Now download the source texts from the site Asterisk and configure:
$ wget -c downloads.digium.com/pub/Asterisk/Asterisk-1.4.11.tar.gz
$ tar xzvf Asterisk-1.4.11.tar.gz
$ cd Asterisk-1.4.11
$ ./configure --prefix \u003d / usr
When the script finishes, we will see the project logo and some information about the settings in the console.
$ make
$ sudo make install
Note: if you are installing version 1.2, then to support the mp3 format, you should enter "make mpg123" before the make command, version 1.4 does not react to this command in any way.
After compilation, the following binaries will be installed, among other things:
- / usr / sbin / Asterisk - server daemon Asterisk, which provides all the work;
- / usr / sbin / safe_Asterisk - script for starting, restarting and checking server operation Asterisk;
- / usr / sbin / astgenkey - script for creating private and public RSA keys in PEM format, which are required for work Asterisk.
To install configuration file templates and documentation, type:
$ sudo make samples
Sample config files will be copied to / etc / Asterisk... If there are already configuration files in this directory, they will be renamed with the ".old" prefix. To build the documentation, you need the doxygen package, if it is not there, install:
$ sudo apt-get install doxygen
$ sudo make progdocs
Install the extension package in the same way. Asterisk-addons (this step is optional, you can safely skip it). Many of the modules included in this kit are experimental. They should be installed only if you need to record information in the database, support for mp3 files and the ooh323c protocol (Objective Systems Open H.323 for C):
$ wget -c downloads.digium.com/pub/Asterisk/Asterisk-addons-1.4.2.tar.gz
$ tar xzvf Asterisk-addons-1.4.2.tar.gz
$ cd Asterisk-addons-1.4.2
$ ./configure; make; sudo make install; sudo make samples
Installation Asterisk finished. First, it is recommended to run the server in debug mode and view the output for errors:
$ sudo / usr / sbin / Asterisk -vvvgc
If we receive the message “ Asterisk Ready ”and the prompt of the management console, then everything is in order. We leave:
* CLI\u003e stop now
Now you can proceed to further configuration.
Configuring Interface Card Support
If you plan to connect the server Asterisk using special interface cards to ordinary telephone networks, you should take care of the availability of appropriate drivers, implemented as a kernel module. But even if there are no such devices in the computer, it is still recommended to install these drivers. The fact is that all Zaptel devices have a timer, and for the full operation of the IP telephony server it is necessary. But if a Zaptel device is not at hand, you can use a special driver - ztdummy to emulate it.
From the repository, install the zaptel and zaptel-source packages and build the modules for our system:
$ sudo apt-get install zaptel zaptel-source
$ sudo module-assistant prepare
$ sudo m-a -t build zaptel
The package zaptel-modules - * _ i386.deb will appear in / usr / src, install it using dpkg. After that, we check the operation of the kernel modules:
$ sudo depmod -a
$ sudo modprobe ztdummy
And if device support is needed:
$ sudo modprobe zaptel
$ sudo modprobe wcfxo
To ensure that they are automatically loaded, run the following command:
$ echo "ztdummy \\ nzaptel \\ nwcfxo" \u003e\u003e / etc / modules
Create rules for UDEV:
$ sudo mcedit /etc/udev/rules.d/51-zaptel.rules
KERNEL \u003d "zapctl", NAME \u003d "zap / ctl"
KERNEL \u003d "zaptimer", NAME \u003d "zap / timer"
KERNEL \u003d "zapchannel", NAME \u003d "zap / channel"
KERNEL \u003d "zappseudo", NAME \u003d "zap / pseudo"
KERNEL \u003d "zap0-9 *", NAME \u003d "zap /% n"
You can also use the source code or the CVS version of the driver. If you compile yourself, you will need the kernel header files (or sources):
$ sudo apt-get install linux-headers-`uname -r`
$ sudo ln -s /usr/src/linux-headers-2.6.20-15-generic /usr/src/linux-2.6
Now we get the latest drivers:
$ cd / usr / src
$ wget -c downloads.digium.com/pub/zaptel/zaptel-1.4-current.tar.gz
We compile and install:
$ tar xzvf zaptel-1.4-current.tar.gz
$ cd /usr/src/zaptel-1.2.17.1
$ ./configure
$ make
$ sudo make install
And so as not to manually create configuration files:
$ sudo make config
After this command, a script will be created to automatically start the modules that are part of Zaptel, and the config / etc / default / zaptel (or / etc / sysconfig / zaptel), which will indicate which modules should be loaded. I recommend leaving only what you need in this file. Trying to load the module:
$ sudo modprobe ztdummy
$ lsmod | grep ztdummy
ztdummy 6184 0
zaptel 189860 1 ztdummy
It's okay. After installation, two more files will appear on the system:
- /etc/zaptel.conf - describes the hardware configuration;
- /etc/Asterisk/zapata.conf - server settings Asterisk for the Zap channel driver to work.
Detailed instructions for various devices are given in the documentation. In Russian, you can read about this in the document "Asterisk% 0A+ config + zaptel.conf "\u003e Configuring the Zaptel Kernel Driver." But we don't stop there, we still have a lot of work ahead of us. After configuring, we check it with ztcfg -vv.
User registration
Now if you look in the directory / etc / Asterisk, a large number of files can be found. But the size of the journal article will allow us to get acquainted with only some of them. So, in Asterisk.conf specifies the directories that will be used Asterisk during operation, the location and owner of the socket used to connect the remote control console, as well as the default server startup parameters. Some directories are not created during installation, you have to do this manually:
$ sudo mkdir -p / var / (run, log, spool) / Asterisk
$ sudo adduser --system –-no-create-home Asterisk
$ sudo addgroup --system Asterisk
Add user Asterisk to the audio group:
$ sudo adduser Asterisk audio
$ sudo chown Asterisk: Asterisk / var / run / Asterisk
$ sudo chown -R Asterisk: Asterisk / var / (log, spool) / Asterisk
Next, we are interested in the sip.conf file, which defines the SIP servers and clients with which our Asterisk... Each of them is represented in the file as a separate block, which begins with a table of contents, enclosed in square brackets. There are a lot of parameters in sip.conf, we will limit ourselves to adding a SIP account:
$ sudo mcedit /etc/Asterisk/sip.conf
type \u003d friend
host \u003d dynamic
; defaultip \u003d 192.168.1.200
username \u003d grinder
secret \u003d password
language \u003d ru
nat \u003d no
canreinvite \u003d no
context \u003d office
callerid \u003d grinder<1234>
[email protected]
; all codecs should be disabled before using the allow parameter
disallow \u003d all
; the order of the codecs does not matter
allow \u003d ulaw
allow \u003d alaw
The type field indicates what this client can do. If the value is user, it will only be allowed to receive incoming calls, with peer, it will only be able to call, and friend means all actions at once, that is, user + peer. The host field specifies the IP address from which this client is allowed to connect. If it can connect from any address, specify host \u003d dynamic. And in order to call the client in this case, when it is not yet registered, you should write down the IP address in defaultip, where you can always find it. In username and secret, we indicate the login and password used by the client when connecting. The Language parameter specifies the greetings language code and specific telephone signal settings, which are defined in the indications.conf file. If the client is behind NAT, set the corresponding field to yes. Disabling canreinvite forces all RTP voice traffic to pass through Asterisk... If clients support SIP re-invites, they can be allowed to connect directly by specifying canreinvite \u003d yes. The сontext field defines the dialing plan, in which calls from this client fall, and the callerid - the string that will be displayed when calling from the client. By default, the default context is used, which takes all settings from the demo context. The latter is intended solely for demonstration purposes, in the working system you need to create your own context. The mailbox field points to voicebox 1234 in the office context. The rest of the clients are configured in the same way.
After defining SIP accounts, our clients can register on the server Asterisk and make outgoing calls. To enable them to receive calls, refer to the extensions.conf file, which describes the Dialplan that distributes calls in the system. All allowed extensions are also listed here.
$ sudo mcedit /etc/Asterisk/extensions.conf
include \u003d\u003e default
exten \u003d\u003e 1234,1, Dial (SIP / grinder, 20)
exten \u003d\u003e 1234,2, Voicemail (grinder)
Everything is simple here. We assign the number 1234 to the grinder user, and if he does not answer the call, he can leave a message in his voice mail. The number after the number indicates the priority, which determines the sequence of tasks. Now if Asterisk is running, you should connect to its console by running on the same machine Asterisk -r, and use the reload command to force it to re-read the config files. There are also commands for reloading a specific file. For example, the dialplan is re-read with the extensions reload command.
The server is ready to receive clients. At Asterisk address% 0AAsterisk% 0A_softphone.html "\u003e www. Asteriskguru.com/tutorials/configuration_ Asterisk_softphone.html choose a soft client and try to connect. For example, I like the free version of the simple and easy-to-use program ZoIPer (formerly Idefisk). There are versions for Linux, Windows and Mac OS X. Another good and also multiplatform client is X-Lite.
If everything is fine, a message like “Registered SIP" grinder "at 192.168.0.1 port 5060” should appear in the console, dial the number and call.
We have configured Asterisk in a minimal configuration, but that's not all it can. The connection to another IP-telephony server, call parking, music while waiting, billing, using the GUI for server administration, etc., remain behind the scenes, but we will try to fill these gaps in the following articles.
