SIP telephony server service

Including this service, you get the opportunity to use sIP telephony server (PBX) based on Asterisk inside home network.

You can register your smartphone or computer with SIP client in this telephone station and call your relatives and acquaintances, which are also registered in this server.

Hint! In addition, you can configure your SIP telephony server on this instruction.

Example of use and configuration

Everything is very simple.

1. On the Application page you need activate SIP telephony server servicewhich will perform a single point of registration of your smartphones, computers and other devices using the SIP protocol. This server will switch your telephone calls within your distributed network.

Server address on your network - 172.16.255.14

After starting the server, check its availability by running the command. ping 172.16.255.14

2. If Ping was successful, then register your devices. To do this, you will specify 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 connecting.

In this example, two phone numbers - 10 and 11 with password 1111 each.

2.2. Adjust your device. This example shows two implementations of the connection - a regular SIP client of the Android OS and using the Zoiper application posted on a PC with Windows 8

So android. He has a built-in SIP telephony client.

First create a new SIP server account.

We specify the previously selected phone number with which we will register on the SIP telephony server (in our example 10), password and server address

After conservation account The phone will try to register on the SIP server.

There are still various settings, and most likely, you will need to choose "Take Inbox" so that the phone is in connection with the SIP server and expected receipt incoming challenge. Actually it's all.

Now let's get the contact of the one who is going to call through our SIP telephony server. To do this, let's go to the notebook and add new contact which is called "Cottage". But there is a nuance ... we need to specify the "cottage" number and do it is needed in the field called "Challenge via the Internet".

On the main screen of contact, this field is missing, so you need to flush down to the point "Add another field" And then the new window will open with the selection of fields, among which will be "challenge via the Internet"

Now it remains the last - to specify the number in this field. It is indicated as it is represented in the figure below -

On this, our client on android is ready. Let's add settings on the second side of our future telephone connection.

2.2 In the role of the second side we will have PC with Windows 8 with the Zoiper SIP telephony client.

After installation, enter the settings and add new account with SIP protocol.

In the account settings, specify the username and server address in such a format: This email address is protected from spam bots. You must have JavaScript enabled to view. .4 and password. Then check the box " Skip auto detection"

After saving the settings, go back to the settings and click the Register button. In the right corner, the status record should be displayed - Registred.

If everything has registered successfully, you can try and call. Close this window. On the main screen, select DialPad and type the Android number - 10.

We hope that your android rang and you can check the quality of communication.

Here, actually, and all

Technical features

Your SIP telephony server, located at 172.16.255.14, is only a SIP server and no longer contains any other data except the numbers you entered.

Period testing services

We plan that the testing period of the SIP telephony server will take about a month.

Refusal of the service

You can refuse the service at any time. In this case, the registration of your devices will be deleted, and the SIP telephony server is stopped.

The SIP server is a set of software to launch IP telephony within an office or production. Traditional telephony is characterized by high call prices and does not provide special advantages for business. Deploying its own production or office PBX makes it possible to adjust the distribution of calls, reduce the cost of communication within the company and establish voice communication with clients.

Select IP-telephony server is easy - in our review you will find solutions for Windows and Linux. But they are increasingly displacing ready-made solutions from providers. In addition, the prices for launching office telephony kopeck. The client remains to choose the tariff, pay communication services, connect equipment to the network and spend everything necessary settings.

We have one of the most popular SIP servers in the world for organizing office telephony. The project appeared in 1999 and was called upon to replace the expensive mini-PBX. The server is running under control operating system Linux, has all the necessary functionality:

• Supports work with traditional telephony.
• Able to manage distribution, handling phone calls.
• Supports video seasters.
• Can be integrated into the CRM system.
• Supports call encryption to prevent listening.

Asterisk SIP Server functionality can expand due to additional software. It works almost with any IP telephony protocols, even the most complex tasks. His main drawback is complexity. For server management, convenient Web interfaces have been developed, 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 telephonization of enterprises of any size. It can be small firms or large corporations with dozens of branches, divisions and divisions. It supports the full functionality of office PBX - work with calls, integration in CRM, conference communication, call-center functions and much more. The product is notable for comprehensive support from the developer. Working environment - Windows operating system. Implement your own developments, as in Asterisk, will not work in connection with the server's closed source code.

SipXECS Server

Another software PBX to solve business tasks. It is deprived of support for multiple protocols, only works with SIP. A web interface is used to control the telephony. Present support for most standard functions - translation / processing of calls, fast dialing, conference, holding and expectation, multichannel communications and much more. The server runs running the Linux operating system.

OfficeSip Server

Free app for organizing office telephone communication. Suitable for small and medium offices that do not need additional features. For large enterprises with divisions and branches worldwide, this SIP-Server will not suit. But to connect the accounting department, the director, the department of personnel, several cabinets with access to Intercity and international Communications - you're welcome.

The server operates under the Windows operating system and does not create difficulties. It is free even for business customers, which determines some demand for this product. Installation passes quickly and without delay, registration of new subscribers is made in a couple of clicks with mouse. If the task is to configure the connection with your own hands, but you do not have much experience, take advantage of this simple and free solution.

Ready solutions from providers

Recently, business has moved to ready-made solutions. There are several reasons for this:

• Cost reduction - Connection is often free, only expenses for intercity, jobs and some additional functions are paid.
• Safety - self-configuration VoIP in the office will not give confidence in the security of the system from hacking and attacks. Providers do this certified personnel.
• Convenience - only computers are needed from additional equipment and telephone devices. No separate "iron" for IP servers.

Consider several solutions for organizing IP telephony for business.

Cloud PBX from Zadarm

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 mess around with the equipment - it is enough to make the subscriber system and configure the distribution of calls. Benefits of Stage:

• Free connection to IP telephony.
• The provider offers multi-channel numbers in 90 countries of the world and in many Russian cities.
• The ability to integrate with the CRM used.
• Full functional cloud PBX.
• Free challenges within the company and its branches, regardless of the geographical position of jobs.
• Access to the number 8-800 with a full-fledged call center functionality.
• API interface to implement your business tasks.

The provider guarantees high quality voice, supports customers by phone or through the inner chat, offers low-cost calls in Russia and the world. And all this without expensive equipment and settings. Order a service and get ready cloud PBX After 5 minutes. Setup is carried out through a convenient web interface.

As customer reviews show, Provider Zadarma provides high-quality voice transmission and full functionality of office PBX for large enterprises and small firms.

Cloud PBX from sipnet

One of the oldest IP telephony providers. It works not only with individualsbut also with corporate clients. The starting rate will cost only 1000 rubles. It will include three television jobs, a package of minutes to choose from (from 600 to 1500 minutes to the rooms of Moscow and St. Petersburg, throughout Russia or mobile). There is no connection fee. Also, clients are available options that expand the functionality, the number of places and providing services of a personal manager. SiPNet is a full-fledged PBX for business, including the features of the call-center.

Despite the development of various information exchange systems, such as email and instant messaging services, the usual phone will still remain the most popular communication tool. The key event in the history of telecommunications and the Internet was the emergence of voice transfer technology over IP networks, so in recent years the concept of the phone has changed. Using VoIP is modern, convenient, cheap, as you can combine remote offices, without even resorting to telephone operators. What other arguments are needed in order to raise your IP telephony server?

Project Asterisk.

Asterisk. Present in the repositories of packets of most distributions. So, B. Ubuntu team sudo apt-cache search Asterisk. It gives a decent list of packages, after installing which you can immediately start setting up. But the installation from the repository has one minus - as a rule, in it version Asterisk. It is decent lagging behind the current, which can be downloaded from the official site. Therefore, we consider universal way Installations on the example of the same Ubuntu, although all said (with rare exception) applies to other distributions.

Install packages required for compilation:

$ sudo APT-Get Install Build-Essential Automake
Autoconf Bison Flex Libtool Libncurses5-DeV Libssl-Dev

In addition, it is strongly recommended to install the libpri library, even if the Primary Rate ISDN is not needed (the primary type of digital network with the integration of services). This can be done either through the repository: sudo APT-Get Install Libpri1.2, or using the source texts:

$ wget -c downloads.digium.com/pub/libpri/libpri-1.4-current.tar.gz

Compiling the library is standard, so we will not stop at it.

Now download the source code 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

At the end of the script work in the console, we will see the project's emblem and some information about the settings.

$ Make
$ sudo make install

Note: If version 1.2 is installed, then to support MP3 format Before the MAKE command, you should enter "make MPG123", version 1.4 does not respond to this command.

After compiling, among other things, the following executable files will be installed:

1. USR / SBIN / ASTERISK - Server Demon Asterisk.which provides all the work;
2. / USR / SBIN / SAFE_ASTERISK - script for running, restarting and verifying the server Asterisk.;
3. / usr / sbin / astgenkey - script to create a closed and public RSA keys in PEM compounds that are necessary for work Asterisk..

To set configuration file templates and documentation, type:

$ sudo make samples

Examples of configuration files will be copied to / etc / Asterisk.. If configuration files are already available in this directory, they will be renamed with the ".old" prefix. To build the documentation, you will need a DOXYGEN package if there is no, install:

$ sudo APT-Get Install Doxygen
$ sudo make progdocs

Similarly, put the package with extensions Asterisk.-Addons (this step is not obligatory, it can be safely skipped). Many modules included in this set are experimental. They should be installed only if information is required in the database, support for MP3 files and 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 start the server in debug mode and view error output:

$ sudo / usr / sbin / asterisk -vvvgc

If you get the message " Asterisk. Ready "and inviting the control console, it means everything is in order. We leave:

* CLI\u003e Stop Now

Now you can go to further configuration.

Configuring Interface Card Support

If you are planned to connect the server Asterisk. Using special interface cards to ordinary telephone networksYou should take care of the presence of appropriate drivers implemented in the form of the kernel module. But even if there are no such devices in the computer, these drivers are still recommended to install. The fact is that in all Zaptel devices there is a timer, and for the full operation of the IP telephony server it is necessary. But if there is no zaptel-devices at hand, it can be used to emulate it special driver - Ztdummy.

From the repository, we install Zaptel packages, zaptel-source and collect modules under your system:

$ sudo APT-Get Install Zaptel Zaptel-Source
$ Sudo Module-Assistant Prepare
$ sudo m-a -t build zaptel

In / usr / src, the ZAPTEL-MODULES package will appear - * _ i386.deb, set it using DPKG. After that, check the operation of the kernel modules:

$ sudo depmod -a
$ sudo modprobe ztdummy

And if you need support for devices:

$ sudo modprobe zaptel
$ sudo modprobe wcfxo

To provide them automatic loading, Perform 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. When self-compilation, the kernel header files (or source texts) will be needed:

$ 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 get last version Drivers:

$ CD / USR / SRC
$ wget -c downloads.digium.com/pub/zaptel/zaptel-1.4-current.tar.gz

Complete and install:

$ tar xzvf zaptel-1.4-current.tar.gz
$ cd /usr/src/zaptel-1.2.17.1.
$ ./configure.
$ Make
$ sudo make install

And to manually do not create configuration files:

$ Sudo Make Config

After this team, a script will be created for automatic launch Modules included in Zaptel, and config / etc / default / zaptel (or / etc / sysconfig / zaptel), in which it will be indicated which modules must be downloaded. I recommend to leave only the necessary one in this file. We try to download the module:

$ sudo modprobe ztdummy
$ lsmod | Grep Ztdummy
Ztdummy 6184 0.
Zaptel 189860 1 Ztdummy

Everything is fine. After installation, two more files will appear in the system:

1. /etc/zaptel.conf - describes the configuration hardware;
2. /etc/asterisk/zapata.conf - Server Settings Asterisk. For the work of the ZAP channel driver.

Detailed instructions for all sorts of devices are given in the documentation. In Russian, on this occasion, you can read in document "Asterisk% 0a+ Config + zaptel.conf "\u003e Configuration of the Zaptel kernel driver. But you do not stop there, we still have a lot of work. After the setup, check the ZTCFG -VV command.

Registration of users

If you now see in the catalog / etc / Asterisk., You can detect a large number of files. But the size of the magazine will allow us to get acquainted only with some of them. So, B. Asterisk..conf shows the catalogs that will use 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, it will have to be done manually:

$ sudo mkdir -p / var / (Run, Log, Spool) / Asterisk
$ sudo adduser --System --No-Create-Home Asterisk
$ sudo addgroup --System Asterisk

Add a user Asterisk. In the Audio group:

$ sudo adduser asterisk audio
$ Sudo Chown Asterisk: Asterisk / Var / Run / Asterisk
$ sudo chown -r Asterisk: Asterisk / Var / (Log, Spool) / Asterisk

Further, we are interested in the file sip.conf, where the servers and the SIP clients are defined, with which our Asterisk.. Each of them is presented in the file with a separate unit, which begins with the table of contents enclosed in square brackets. The parameters in sip.conf are quite a lot, 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]
; Before using the Allow parameter, disable all codecs.
DISALOW \u003d ALL
; The order of the codecs does not matter
Allow \u003d ulaw.
Allow \u003d Alaw.

The Type field indicates that this client can do. With the user value, it will only be allowed to receive incoming calls, with PEER it can only call, and Friend means all actions immediately, that is, User + Peer. The Host field indicates the IP address from which the connection of this client is allowed. If it can connect from any address, specify HOST \u003d DYNAMIC. And in this case, in this case, call the client when it is not yet registered, in DefaultIP you should record the IP address for which it can always be found. In UserName and Secret, specify the login and password used by the client when connected. The Language parameter sets the greeting language code and specific settings of the phone signals that are defined in the Indications.conf file. When working client for NAT "Ohm in the appropriate field, you must set the value of Yes. Disable CANREINVITE causes all voice RTP traffic through Asterisk.. If customers support SIP Re-Invites, they can be resolved to connect directly by specifying CANREINVITE \u003d YES. The context field determines the set plan in which the calls coming from this client come from this client, and the CallerID line will be displayed when calling from the client. The default Default context is used, which takes all the settings from the Demo context. The latter is intended solely for demonstration purposes, in the working system you need to create your context. The Mailbox field indicates a 1234 voice box in the Office context. The remaining customers are configured in the same way.
After defining SIP accounts, our clients can register on the server Asterisk. and make outgoing calls. So that they have the ability to receive calls, you should contact the extensions.conf file, which describes the dialplan plan, distributing calls in the system. All permitted expansions are indicated 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. Behind the user Grinder fix number 1234, and if he does not answer the call, it will be possible to leave a message in voice mail. The number after the number means a priority that determines the sequence of tasks. Now, if Asterisk. Running, you should connect to its console by performing on the same machine Asterisk. -R, and using the Reload command to force it to re-read the configuration files. There are both commands to restart a specific file. For example, a set plan is rereaded by the Extensions Reload command.

The server is ready for receiving customers. At Asterisk% 0aAsterisk% 0a._SoftPhone.html "\u003e www. Asterisk.guru.com/tutorials/configuration_ Asterisk._SoftPhone.html choose a software client and try to connect. For example, I like the free version of the simple and understandable to use the Zoiper program (previously IDEFISK). There are versions for Linux, Windows and Mac OS X. Another good and also multiplatformed client - X-Lite.

If everything is fine, the message should appear in the console like "REGISTERED SIP" GRINDER "AT 192.168.0.1 Port 5060", type the number and call.

We set up Asterisk. In the minimum configuration, but this is not all that he can. The frame remains to connect to another IP telephony server, a call parking, music during waiting, billing, using the GUI to administer the server and so on, but we will try to fill these gaps in the following articles.

Corporate use of SIP numbers often passes under the Windows installed on most office PCs. Consider existing VoIP solutions for this system.

Web calls: What, how, from where

Approved by the supreme leadership

The software intended for the highest level of management is usually designed for the maximum possible leveling of the difference between online conferences and a physical neighborhood at the table at the meeting. It is above the elimination of the borders of the virtual and real worlds, the developers of the silicone valley are fighting, in the hope of functionally "surpass" the ASTERISK server.

• B-Force. Developed by the company of the same name in 2010, and since then is being improved daily. Russian-speaking Wikipedia users position the program as one of the few suitable safety requirements even for use in government agencies.
• 3CX Phone - Multiplatform, can be used "in a bundle" not only with Windows, but also with Lincusoids, as well as under mobile OS - Android, I- / Mac-OS, etc. All possibilities are available to subscribers for free, 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) the leader of the top 5 most comfortable in the use of sypphones.
• Brosix. One of the most secure softwareworking on the US federal standard under 256-AES symmetric encryption. Corporations preferring to use Brosix Business will have to pay a license, in return to get the opportunity to create private crypto-resistant network by pressing multiple buttons. Physicians can legally use the program for free, but in the Light version, where there are no "white board" functions, the exchange of desktops and conference calls.

Clerk Comfort - Pledge of Stable Work Company

But in high-quality communication, not only leadership, but also ordinary "white collars" needs. Some routine, neither the work of clerks, it was on it that the company was founded, and therefore in the interests of the leadership to simplify their actions. Many offices are costing for internal communication with the functionality of such programs as Skype, Yahoo! Messenger and them like, but in some cases optimal decision There will be a special software.

• Call Office. "Sharpened" under work with large client bases. Maximum simplifies call, sending messages to (e-mail / sms) and other mass notifications.
• Ventrilo. Softfon-radio associated with gameimascular chaty. Despite stereotypes, popular in companies, where profit depends on the reaction rate and dynamics - for example, in delivery services or closed offline exchanges.
• Sippoint. A utility that supports the multiplayer interface and allows you to configure multistage contact databases. In addition, users can share files in a closed intuophice network. Related by the fact that it easily ports the data from / to other systems - Google Talk, QIP and other popular messengers.
• Jabbin. The main advantage of the softphone is the possibility of calls, even without provider SIP connection, only with a custom web connector, including local intra-connections. But at the same time, alas, there is no possibility to call an urban or mobile number.

Best Softphone for the best subscribers

Subscribers Website do not have to torment the dilemma of choice: there is universal and at the same time simple programAvailable to all users - telephone IP server - Youumagic Softphone. In addition to the obvious advantages of working with the provider himself, the subscriber will receive such "bonuses":

• virtual PBX with flood protection, spam and DDoS attacks on central nodes, which guarantees a comfortable connection without interruptions;
• the technical support service will respond in detail to any question, and in case of problems - will quickly decide them;
• each softphone user will be able to use multiple accounts and financial calculators for each of them, thereby taking into account spending on traffic.

These and many other features make use as comfortable as possible on any platform, including Windows, Android and other OS.

Tags:

SIP telephony can significantly reduce the costs of telephone communication. 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 Communications is used for organizing intuophisc telephony - for this you need to install on one of the SIP-server computers and connect software and hardware phones to it. In this review, we compare the most popular SIP servers, including free:

• Asterisk;
• Kamailio;
• OfficeSip Server;
• sIPX.

Let's consider these servers in more detail and find out how to run the SIP server with your own hands.

We will start this review with the consideration of one of the most famous servers for IP-telephony - this is an Asterisk SIP server. It is focused on organizing office telephony and is very popular.

Asterisk SIP Server

Asterisk can be called a freely distributed solution, but the licensed modules in it still have. The program works in operating linux systems and produced in the form of several distributions that differ in functionality, web interfaces and sets of additional modules. It cannot be said that this is a solution for novice users - Rather, this is a more professional solution. The Asterisk SIP server is endowed with the following features:

• Forwarding and calling calls;
• Holding and waiting for a call (with a musical background);
• Interception and parking of calls (functions allow you to respond to calls from other devices or continue to talk on them started on other devices);
• Conference call;
• Video communication;
• Call-center functions;
• Integration of traditional telephone lines;
• Administering through the web interface;
• Billling functions.

We can say that the use of the Asterisk SIP server will solve the problem of any complexity. Scalability, the presence of additional modules, a huge number of supported protocols - all this can be called the advantages of the program. As for the shortcomings, this is the complexity in the settings for novice users and the presence of a double license.

Despite the fact that this server is free, there may be modules distributed based on the licensed code - sometimes it causes some problems.

Kamailio SIP Server

Once this project was called as the Openser SIP server, but in 2008 he was renamed Kamailio. But it is impossible to be called the most famous if compared with such monsters as 3cx or Asterisk. The server is characterized by a decent functionality and is most often used in a professional environment. therefore for solutions simple tasks He is not suitable.

In its advantages, we can enable the support of a large number of all kinds of modules that expand its functionality. The shortcomings included the complexity of the setting.

SIPX SIP server

This is another free product running Linux systems. The SIPX server has simplicity and orientation for office use. The developers have endowed it with a decent functionality, providing a large number of voice call management functions. When using suitable equipment, the SIPX SIP server allows you to solve even the most complex tasks.

Its advantages included stability, simplicity and minimum sizes. SIPX allows you to deploy local SIP networks in a matter of hours, which is used to quickly telemine offices. Also, this server is distinguished by free. As for the shortcomings, the most negative Moment It is that all functions are needed to have advanced phones and VoIP gateways.

SIP servers for Windows

Linux systems have the highest stability and excellent performance. But they require certain knowledge, and they cannot be called friendly to simple users. Therefore, more understandable SIP servers for Windows appeared in the world. Of course, here are users and system administrators They may wait for various difficulties, but they can get around them much easier.

SIP server 3CX

Among the most advanced SIP servers, we can highlight VoIP-PBX 3CX PHONE SYSTEM for Windows. This solution is oriented to an organization. corporate communications Any scale, even if individual offices are at different ends of the planet. The advantages of the server:

• Full voice functionality;
• Support large number customers (including their own software for various platforms);
• Web conference support;
• Integration of third-party services of 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 many training materials, conducts training activities, is carried out comprehensive user support. The choice of clients is a standard free version, as well as a commercial version, characterized by supporting additional features.

The trial free 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 what the phone System server 3CX is running operating windows systems. It is extremely flexible in the settings and has enormous functionality. If you need ordinary telephony, not a whole call center, then you will be enough free version. Disadvantages - it is impossible to supplement the system with something, since the source code is closed. However, it cannot be considered a significant disadvantage.

OfficeSip Server SIP Server

Free OfficeSip Server SIP server is free distributed software For Windows. This server is so simple that even the most serious user can cope with its installation and configuration. Installation and starting program take a couple of minutes, after which you can start creating local user accounts.

Also it is possible to connect to third-party IP provider for calls around the world.. Excellent program for small offices in need of office telephony. Advantages of the program:

• Ease in the settings;
• Work in the Windows environment;
• Easy to connect new subscribers;
• The presence of communication with the outside world.

Disadvantages of the program:

• Lack of many convenient office and voice functions;
• The impossibility of scaling;
• There is no possibility of connecting to "your" PBX from anywhere in the world (only local connections).

However, this is an extremely affordable and free SIP server for small offices.