Office telephone systems IP telephony IP PBX for special networks

Digital PBX

Modern office and institutional automatic telephone exchanges (PBX) allow, if there is a limited number of city telephone lines provide access to them to all employees of the organization. PBXs can be analog or digital, and also differ from each other in number capacity. For large enterprises, it is more appropriate to use digital PBXs, as they have large capacities, advanced functionality and a wide range of capabilities in the field of IT technologies.
When building a communication system for a large enterprise, the choice of a digital PBX will be determined by the fact that such a PBX can become a universal solution for building a unified telephone and information communication network for the enterprise, will provide developed means for processing a massive flow of calls and the ability to create automated information systems. The advantages of digital PBXs over analog ones are significantly better signal quality, new capabilities and service functions, and a completely different level of versatility. Along with speech signals converted into a digital stream, a digital PBX can process and transmit digital information from other sources (data, video signals, telemetry signals, etc.).

Digital PBX allows you to implement any solutions necessary to ensure the efficiency of business processes and increase the productivity of a large enterprise.

Among such solutions are:

• connection to a telecom operator via digital stream E1 ISDN PRI and organization of multi-channel numbers;
• organization of conference call functions for conferences and conference calls;
• connection of video conferencing equipment via ISDN BRI;
• connecting conference room equipment to the PBX;
• creation on the basis of a digital PBX of a multi-level autoinformer (DISA) with a large number of voice mailboxes, answering machine and voice greeting;
• connecting tarifficators and billing systems to the PBX to control and record telephone calls;
• creating an internal corporate network with remote departments using Voice over IP (VoIP) or ISDN PRI protocols, as well as providing remote subscribers with IP phones;
• connecting to the PBX a telephone conversation recording system;
• connecting to a GSM gateway to the PBX and organizing outgoing communications to cell phones;
• organization of a DECT microcellular communication system to ensure personnel mobility on the territory of the enterprise;
• organization of computer-telephone integration CTI in order to optimize the work of the secretary or operators when processing incoming calls, or to build a full-fledged call processing center.

The most complete functionality of digital PBXs is revealed when using digital subscriber devices, which, in addition to analog and digital telephones (including system telephones), include ISDN terminals, cordless phones, administrator computers, voice mail or charging, call registration printers, built-in modems for remote administration, intercoms, external speakers and calls.

Digital PBX Panasonic KX-TDA series are today the most popular products on the telephone equipment market due to their high reliability and the best price-quality ratio. The KX-TDA series includes 4 PBX models: KX-TDA30, KX-TDA100, KX-TDA200 and KX-TDA600. These stations are made using the same technology and differ from each other in the maximum possible capacity.

The Panasonic KX-TDA PBX series is digital PBX with support for IP telephony and supports such city line interfaces as analog two-wire city lines CO, ISDN PRI (for KX-TDA100/200/600 PBX) and ISDN BRI (for KX-TDA30 PBX), SIP and H.323 trunks, and QSig interstation protocol (via ISDN or VoIP). The initial configuration of the PBX does not provide for either external or internal ports. The required capacity is obtained using expansion cards for external and internal lines. As the company grows, the capacity of the PBX increases and the functionality increases. Thanks to the modular architecture of the Panasonic KX-TDA PBX, it can be configured for a wide range of tasks and for the specific needs of the end user.

IP-PBX IP-PBX is a solution designed to meet the requirements of modern business and is suitable for most organizations that need to deploy or upgrade an office PBX. IP PBX is a full-featured telephone system that services telephone calls over IP data networks. The entire conversation is represented as data packets that are sent over an IP network using Voice over IP (VoIP) protocols. Unlike traditional telephone equipment, IP-PBX does not require the creation of separate data networks; the connection is made through an existing computer Ethernet network. In this case, the same communication channel is used to transmit voice traffic, data, fax messages and video information.
IP-PBXs demonstrate their advantages when implemented in large enterprises consisting of several distributed offices. At the same time, the task is to unite offices into a single telephone system and transfer conversational traffic from one office to another via IP telephony channels. Connecting enterprise offices via IP channels is usually cheaper than via other protocols (ISDN BRI, PRI) due to the lower cost of traffic.

The main advantage of implementing an IP PBX at an enterprise is to reduce the cost of employee communications between offices. As a rule, companies that want to create an interoffice telephone network already have communication channels between local computer networks their offices, and using them for telephone communications can be cost-effective. In addition, telephone communication between offices can be organized via the Internet through closed information exchange channels using VPN (Virtual Private Network) technology.

A properly designed and programmed IP-PBX network between company offices allows you to broadcast standard room, dialed by an employee of one of the offices in a request to connect to a PBX in another office via an IP channel. In addition, the benefits of using IP telephony are especially obvious when it is necessary to frequently use long-distance communications, since the use of the SIP protocol allows you to use the services of a large number of IP telephony providers, receiving significant savings on the costs of long-distance and international calls.

The most effective and popular solutions in the field of IP telephony equipment today are systems Panasonic KX-TDE series and solutions based on Avaya S8300, S8400, S8500, S8700, S8800 communication servers.

IP-PBX Panasonic KX-TDE series is a “pure” IP-PBX, which uses the principle of IP packet switching for connections. The KX-TDE supports all three major IP telephony protocols: H.323, MGCP and SIP, providing compatibility with third-party SIP phones and the entire infrastructure of the previous Panasonic KX-TDA series of digital PBXs. IP-PBX is designed for installation in the offices of large companies with up to 1000 people, as well as in industrial enterprises, large government agencies, educational institutions, etc.

The greatest effect is achieved by using KX-TDE for building corporate networks, for communication with branches and for organizing remote workplaces for company employees or ensuring employee mobility. This PBX can be used both as the head PBX of a single corporate network, and as a component of a corporate network consisting of several PBXs united into a single network via IP channels.

In the initial capacity, the Panasonic KX-TDE series PBX does not have external or internal ports. The required capacity is gained using expansion cards for external and internal lines, thereby providing a simple and convenient increase in the capacity of the PBX as needed. Capacity expansion can also occur by connecting IP phones and IP trunks.

Solutions built on the basis of communication servers Avaya S8300, S8400, S8500, S8700, S8800 enable large and medium-sized businesses to take full advantage of the communications capabilities required by their business today, and expand them in the future as needed. When paired with the Avaya G700, G450, G350 or G250 media gateways, the media servers provide a versatile solution for a growing business, enabling efficient voice and data processing on a single network infrastructure. These systems are a multi-purpose platform for solving various business problems, ranging from classical telephony problems to the construction of contact centers of various subscriber capacities.

Avaya communications servers offer a wide range of benefits. The scalable architecture of Avaya servers allows for virtually unlimited expansion. For example, the Avaya S8800 server, when used in a single-server version, can support up to 2.5 thousand users, and when modified into several servers, the number of users can be increased to 40 thousand.

Avaya systems are highly reliable- due to numerous options for redundancy of various components, uninterrupted operation of the business telecommunication system is ensured. Thanks to the use of equipment and software Based on open standards, Avaya servers are easy to integrate into an existing network, upgrade and provide specialized programs.

Mini PBX is a well-established, high-quality and reliable telephone connection that provides effective work any modern office or enterprise. Modern office IP PBXs will provide the opportunity to organize excellent internal communications, increasing the efficiency of using existing city telephone lines.

Installing a PBX in an office will greatly simplify receiving calls from city lines and redirecting calls between the work team.

Advantages of IP PBX

The undeniable advantages of such devices include the following factors:

• The number of internal subscribers is increasing, which allows increasing the quality and quantity of services.
• Possibility of introducing restrictions on access to long-distance lines.
• Ease and flexibility of adjusting mechanisms for redistributing incoming calls.
• A convenient function for holding conferences, in which external subscribers from the city network can also participate.
• Setting different modes for receiving calls by time. This allows you to route all calls in the evening and at night to the person on duty or in charge. You can also transfer incoming calls to a machine with an answering machine or fax. At the beginning of a new working day, the mini IP PBX will automatically switch to normal operation mode.
• The forwarding function allows you to transfer calls to any office in the office.

Such convenient functions can significantly improve the quality and efficiency of business processes, and increase the productivity of all company employees.

Why us?

Our online store offers IP PBX and mini PBX of excellent quality. Their reliability and advanced functionality allow you to optimally adjust the configuration to solve any business problems. Current and popular models of such IP ATC will be an excellent help in increasing the pace of development of your company.

Some time ago, a friend from Open Systems approached me and asked me to review open IP PBXs. Since he turned not only to me, but also to other IP telephony experts, the result was a compilation in the magazine, in which little remained of my original review. I publish it in its entirety on Habré.

First there will be a theoretical insert, for those who are not quite in the subject. If you're bored, just skip it! Happy reading! And to make it easier to read, the chapters are accompanied by musical gifts (I highly recommend headphones :-)

Let's go!

Theory

PBX (Private Branch Exhange) is an English term denoting an office telephone exchange that provides establishment, maintenance and termination of connections between devices, that is, switching. PBX allows you to share limited resources(landline lines and numbers) between an unlimited number of internal users, using telephone features such as internal dialing plan, call transfer, putting on hold, and others.

This is why a PBX system is necessary for any organization - it allows you to effectively organize telephone communications in an enterprise (well, it’s still needed;-)

Traditional PBX systems switch channels (communication lines) by switching circuits electric current. New PBX systems switch packets over a TCP/IP network and are called IP PBX. IP PBX operates based on IP telephony protocols. IP PBX can also support traditional communication lines - such IP PBX are called hybrid. During the transition period of migration from traditional telephony to an IP environment, it is hybrid IP PBXs that are most in demand, although the function of converting traditional telephone channels into IP packets can also be transferred to a separate device - a VoIP adapter or VoIP gateway, which is then connected via the IP telephony protocol to the IP PBX.

Currently, only two IP telephony protocols are widely used - H.323 and SIP.

Protocol, or more correctly, H.323 protocol stack, was developed by the International Telecommunication Union (ITU), an international organization that sets recommendations in the field of telecommunications and radio. The purpose of creating the protocol was the need to conduct audio and video conferences over modern telecommunications networks, including digital and IP networks.

SIP (Session Initiation Protocol)- a standard for a method for establishing and terminating a user Internet session, including the exchange of multimedia content (video and audio conferences, instant messages, online games, etc.). The development of the protocol was carried out by the Internet Engineering Task Force (IETF), an open international community of designers, scientists, network operators and providers, which is engaged in the development of Internet protocols and architecture.

The H.323 protocol has great standard set capabilities for working with video conferences (it was created by telephone operators, and the Internet is one of its working environments), and the SIP protocol is more suitable for working in TCP/IP networks, and is more universal (it was created by “Internet workers”, and voice and video - only just one of the types of media content).

The Internet has won, and currently SIP is considered the de facto standard for IP telephony, and the H.323 protocol is used mainly in multi-user video conferencing systems and for the exchange of voice traffic over IP between telecom operators, although in these areas there is a tendency to switch to SIP.

Thus, we can confidently conclude that modern IP PBX systems operate based on the IP protocol SIP telephony.

Let's take a closer look at the SIP architecture.

The SIP protocol specification defines a client-server architecture. The client issues requests indicating what it wants to receive from the server. The server receives and processes requests, and issues responses containing notification of the success of the request, notification of an error, or information requested by the client. Call service is distributed between various elements SIP networks.

The main functional element that implements connection management functions is the subscriber terminal. The remaining network elements may be responsible for call routing and also provide additional services. Let's list the main elements:

• Terminal. When the client and server are implemented in the terminal equipment and interact directly with the user, they are called user agent client - User Agent Client (UAC), and a user agent server - User Agent Server (UAS). If a device contains both UAC and UAS, then it is called a User Agent (UA), and is essentially SIP terminal equipment. Examples of UA - hardware or software SIP phone, SIP adapter.
• A proxy server (from the English proxy - “representative”) represents the interests of the user on the network. It receives requests, processes them and performs appropriate actions. A proxy server also consists of a client and server parts, so it can accept calls, initiate requests, and return responses. There are two types of proxy servers:
  stateful. Such a server stores in its memory all received requests and new generated requests associated with it until the end of the transaction.
  stateless. Such a server simply processes the requests it receives and it is impossible to implement complex, intelligent services on its basis.
• Forwarding server - used to determine the user's current location. The forwarding server does not terminate calls or initiate its own requests, but only reports the address of the required terminal or proxy server. For these purposes, it interacts with the location server. To make a connection, the user does not have to use a redirection server if he himself knows the current address of the required user.
  User location server. The user can move within the SIP network, so there is a mechanism for determining his location at the current time. The user location server is used to store the user's current address and is a database of address information.

Thus, the SIP protocol specification does not define anything other than the mechanism for establishing and tearing down a session between client and server, as well as searching for network elements. Therefore, the SIP protocol is used simultaneously with other protocols that implement user services.

One such supporting protocol is SDP - Session Description Protocol, designed to describe a streaming data session, including telephony, Internet radio, multimedia applications, and streaming applications. The SDP protocol describes the format of headers and fields in which SIP clients and servers list their session capabilities (for example, supported compression algorithms - codecs).

The second necessary protocol is RTP (Real-time Transport Protocol), which is used for direct transmission of real-time traffic. The RTP protocol carries in its header the data necessary to restore voice or video at the receiving node, as well as data about the type of information encoding (JPEG, MPEG, etc.). In particular, the header of this protocol contains the timestamp and packet number. These parameters make it possible, with minimal delays, to determine the order and moment of decoding of each packet, as well as to interpolate lost packets. The underlying transport layer protocol is usually UDP. Connection establishment and termination are not included in the list of RTP capabilities; such actions are performed by the SIP signaling protocol.

Thus, the operation of SIP PBX is based on three main protocols: SIP, SDP, RTP.

There are also protocols that implement additional functionality, for example, SIP TLS and Secure RTP, which add encryption of signaling and media streams, and others, but the main ones are still SIP, SDP and RTP.

However, if the SIP protocol does not define any high-level functions and services, then what is an IP PBX based on the SIP protocol?

What is considered a SIP PBX?

Currently, there are quite a large number of telecommunications software products, which differ from each other in architecture, target functions, supported protocols, popularity, and other parameters. To determine whether they are an IP PBX system, you need to consider them according to the following criteria:

• Does the system support SIP registrar functions? IP PBX must know the location of its users, therefore it must implement SIP registrar functions.
• Does the system support SIP proxy functions? An IP PBX must handle establishing connections between its users and also maintain information about the state of those connections.
• Does the system support mechanisms for controlling an established SIP session? IP PBX must be able to interrupt the current session due to an incoming more important call, or to free up a busy line needed by the manager. In the SIP architecture, such functions are performed by the so-called Back-to-back User Agent (B2BUA). When using B2BU, communication is established not directly between two users, but between each user and B2BUA, and one call turns into two completely independent SIP sessions.
• Does the system support RTP traffic proxying functions? IP PBX must pass media streams through itself, for example, for the purpose of recording conversations.
• What additional applications are available to users? Traditionally, PBX systems support features such as voicemail, conference calls, music on hold, call statistics and others.

The purpose of my article is to review free IP PBX systems distributed in source code, which I will compare in accordance with the above criteria. The most popular and mature open source IP PBX systems today are the following:

• Asterisk
• FreeSWITCH
• SipXecs

Let's take a closer look at them. But first, let's launch the next mega-track!

Asterisk

The Asterisk project was initiated in 1999 by Mark Spencer, owner and sole employee of the American company Linux Support Services.

Mark did systems administration and commercial support for Linux, and also programmed in C.

One of Mark's clients approached him about providing office telephony, and Mark discovered that office PBXs cost a lot of money. And I decided to write my own PBX based on Linux. This is how a project called Asterisk was born.

After some time, Mark founded the company Digium, which began producing Asterisk interface cards with traditional telephone networks (via analog and digital ports).

A large community of users and developers has formed around Asterisk, and the project began to actively develop.

Currently, Asterisk is the most popular open IP PBX in the world, occupying almost 85% of the open source PBX “market” (and in general, open PBXs occupy about 18% of the PBX market in the USA - Open PBXs occupy 18% of the North American telephony market).

The name for Asterisk (from the English “asterisk”, symbolized by *) was chosen very well. In IT, an asterisk denotes the replacement of any character, or an unlimited number of characters. Even the standard capabilities of Asterisk are surprising. The modular architecture of Asterisk allows you to easily connect any business logic, written in almost any programming language, or implemented in Asterisk’s own dialplan language, into the switching field.
Here is a shortened list functionality Asterisk:

• Both IP telephony protocols and traditional communication lines are supported. You can insert Digium PCI cards with analog and/or digital ports in the required number and combination into a server running Asterisk.
• All basic and advanced PBX functions are supported: voice menu, call recording, call statistics, music on hold, voice mail, call queuing and distribution to operators (call center functions), and many others.
• Skype is directly supported (channel driver chan_skype from Digium), there is also a small WEB application that allows you to call Skype users with push-button phones via short numbers from your address book
• Video communication is supported.
• There are applications for voice recognition and speech generation.
• IN latest versions Asterisk supports conversation encryption.
• Asterisk has simple and well-documented interfaces for integration with other systems (AGI and AMI), which makes it easy to integrate communications into business processes and business applications.
• There are a large number of various graphical Asterisk administration tools, both paid and free, among which the most popular is the free WEB interface FreePBX. There are also ready-made distribution kits that allow you to deploy an IP PBX server on a regular PC in a matter of minutes. The most popular free Asterisk distributions are TrixBox and Elastix. It should be said that Digium, the author of Asterisk, also offers a commercial solution based on Asterisk - SwitchVox, which is a comprehensive unified communications solution. In addition to SwitchVox, there are several dozen more, both commercial and open systems based on Asterisk.
• Finally, a very large community of users, developers and integrators has gathered around Asterisk, who help each other learn and use the full variety of Asterisk capabilities. In RuNet, the largest community can be found on the website asterisk-support.ru, which was created in 2004 in order to support the community by the community itself. Also, quite recently, in January of this year, the asteriskpeople.ru project was launched, which presents a map of the Asteriskers community.

Currently, Asterisk continues to develop, even more rapidly than before. In 2010 alone, the number of Asterisk users doubled.

If a few years ago commercial support or individual development for Asterisk could only be obtained from a few companies, today dozens of companies from all regions of Russia provide services technical support and system integration of Asterisk-based solutions, which completely eliminated the risk of using free software in business - any company for a reasonable price can quickly receive guaranteed assistance from top-class Asterisk specialists, some of whom are among the world's top ten Asterisk developers.

The abundance of Asterisk capabilities and active development is also a disadvantage of this product - it is difficult for beginners to quickly master a large amount of information. Also, the most recent versions of Asterisk may experience stability problems due to the large number of additions and changes.

To conclude the Asterisk review, it should be said that Asterisk is an IP PBX solution for the office, although many telecom operators are trying to use the system to provide various services to their clients. But Asterisk is not very suitable for this, since it does not scale very well.

FreeSWITCH

FreeSWITCH is a soft switch, the creation of which was initiated by one of the former Asterisk developers, Anthony Minessale, in 2006. After numerous attempts to use Asterisk under high load, Anthony made a number of comments about the basic architecture of the system, and suggested changing it. However, the author of Asterisk, Mark Spencer, refused to change the kernel. Therefore, Anthony left the Asterisk developers and created his own product from scratch, which he called FreeSWITCH.

Therefore, one of the main advantages of the new product is stability and scalability, as well as cross-platform - FreeSWITCH runs both Linux and Windows.

Another feature of FreeSWITCH is the use of the sofia-sip SIP stack from Nokia, which is considered the best open source implementation of the SIP protocol. In Asterisk, chan_sip is implemented with incomplete compliance with standards. SIP is the main protocol for FreeSWITCH, although others are also supported. PCI drivers boards for integration with traditional telephony, as well as other IP telephony protocols.

FreeSWITCH can be used as a SIP proxy and SIP registrar, as a Session Border Controller (SBC), transcoding Back-to-back User Agent (B2BUA), as a conference or voicemail server.

FreeSWITCH also supports many IP PBX functions, such as call transfer, interception, call parking, call recording, listening and others.

However, today the list of IP PBX applications available for FreeSWITCH is inferior to that of Asterisk.

FreeSWITCH's main configuration interface is text files in XML format, which makes the administration of this system difficult, while Asterisk uses highly readable and convenient .ini files in the section / option format.

There are no ready-to-use ones for FreeSWITCH graphical interfaces management, which also makes it difficult to use. And the existing GUIs for FreeSWITCH (WikiPBX, FusionPBX, blue.box) are far from the functionality of the same FreePBX for Asterisk.

However, FreeSWITCH is actively developing. Some open source telecommunications software experts call FreeSWITCH an “Asterisk killer app,” while others argue (myself included!) that both products have a place in the market, as each has its own unique features.

SipXecs

The SipXecs product is based on the source code of the SipXpbx software, published for free access in 2004 by PingTel.

It should be said that PingTel specialists created one of the very first products with the help of which SIP devices from different manufacturers successfully interacted, and they can rightfully be considered the pioneers of “SIP building”. Since then, SipXecs has been considered the most complete and correct implementation of SIP RFC.

After the launch of SipXpbx, PingTel continued to develop its commercial product, SIPxchange, periodically publishing open access various pieces of code and adding them to SipXpbx.

As active developers joined the open source project, it became difficult to maintain two different products, since the current licensing policy did not allow open source code written by enthusiasts to be included in a commercial product. To solve this problem, in 2007, PingTel changed the structure of the projects, and released the rest of the closed code into general access, combining it with SipXpbx. The new project is called SipXecs.

In 2008, PingTel was acquired by Nortel. Nortel was already shipping its SCS (Software Communications System) product, based on the SipXecs source code, to its customers. Nortel has made significant contributions to both its commercial SCS product and the open source SipXecs project.

In 2009, Nortel declared bankruptcy and the rights to the commercial SCS product were transferred to Avaya. In March 2010, Avaya stopped adding its work to the SipXecs source code. Then the SipXecs user community, including some former PingTel employees, united under the roof of the newly created eZuce company, which is currently supporting and developing the project.

SipXecs software is written in the C++ and Java programming languages ​​(in Java, in particular its SIP stack is written using the Jain SIP library) and runs on Linux OS.

This is the only open IP PBX system, the core of which included a WEB management interface from the very beginning. If Asterisk is positioned as a voice platform, then the SipXecs developers consider their product a “boxed” unified communications solution!

The rich arsenal of Asterisk is located in large number configuration files of various modules, as well as in the built-in command line on management (CLI). SipXecs is controlled via a WEB interface, and it is possible to do only what is provided by the developers.

Asterisk supports many different telephone interfaces - analog, digital, and several IP telephony protocols. SipXecs supports only SIP, being a pure SIP solution. All telephone functionality is implemented within the framework of the SIP protocol specification, and is also divided into completely independent components that interact via SIP protocols/ HTTP / XML-RPC, and which can work both on one and on different servers, which, by the way, ensures reliability and scalability at a new level.

If Asterisk is a "multi-protocol" system that receives calls from different types channels, and converting them into its internal format for the purposes of processing and switching (replacing the old PBX), then SipXecs is a SIP proxy that routs SIP transactions without passing media streams through itself, but closing them directly between agent devices (IP telephones).

However, from the strengths of the SipXecs package also come all its weaknesses. Since media streams are not proxied, it is impossible to implement some important PBX functions, such as call recording. Also, a problem arises when the user is inside a network with private IP addresses - the NAT problem. It is also impossible to implement transcoding where necessary. However, these problems are solved in the latest versions of SipXecs using the FreeSWITCH package, which organically fits into the SipXecs architecture, performing such functions as a conference call server and an IVR server.

Yate

The Yet Another Telephone Engine (Yate) project was started in 2004. Supported operating systems: Linux, BSD, Windows. Written by Yate in C++. Yate does not use external SIP libraries, but implements the SIP stack independently.

Yate is a softswitch that also contains many PBX functions, in particular:

• transfer, hold and call parking;
• waiting music;
• conference call;
• queues
• call statistics

However, Yate is first and foremost a multi-protocol switch with very flexible routing rules. Yate supports IP telephony protocols such as H323, IAX2, MGCP, various SS7 levels (MTP2, SIGTRAN), drivers for streaming digital cards from different manufacturers.

Yate also includes a clustering engine that allows for highly scalable solutions.

Architecturally, Yate uses a microkernel model and a message bus, and uses regular expressions with the ability to place any messages on the bus. This architecture makes it easy to add new modules without affecting existing code. Yate is a real low-level telephone engine.

There is a special free distribution with Yate and a WEB management interface - FreeSentral, which includes a user interface where it manages its settings, such as forwarding, voicemail, address book, and can also view the statistics of its calls.

Among all the products reviewed, Yate has the least functionality, but what Yate can do, it does very well and stably. Another disadvantage is insufficient documentation.

The most common application of Yate is the H323-SIP signaling converter.

Conclusion

Choosing an IP PBX system for your organization among open products is not easy.

What makes the situation worse is that all of them can, in principle, work simultaneously.

Or maybe use them all. Nothing prevents you from using SipXecs as a backbone IP-PBX on which users register, FreeSWITCH as an audio conference server, Yate as a SIP-H323 translator for connecting communication providers via the H323 protocol, since the only implementation of H323 in Yate is much better than any of them 3 implementations of H323 in Asterisk, and Asterisk as a media gateway with Digium or Sangoma streaming boards, as well as a server additional applications, for example, video intercom.

Welcome to the world of open and free telephony solutions!

And for those who read to the end - a musical bonus! Enjoy!

Digital telephone exchanges (IP-PBX)

The Agat UX IP-PBX telephone station is a hybrid solution from the Agat-RT company; it is universal and is intended for building an IP telephony system at an enterprise.

The equipment is successfully used in small and medium-sized firms, government agencies and industrial companies. With the help of the proposed IP telephone stations "Agat UX" you can use cable and wireless technologies, traditional and modern (in particular VOIP) solutions. This guarantees subscribers the most comfortable conditions for communication, created on the basis of a single autonomous device.

Lines served:
Up to 80 FXS, up to 80 FXO.

Up to 1 E1 streams.
SIP Proxy up to 256 subscribers.
SIP, H.323 trunks – up to 30.

Power: built-in 220V.

Lines served:
Up to 80 FXS, up to 80 FXO.
Up to 40 digital subscriber lines.
Up to 2 E1 streams.

SIP, H.323 trunks – up to 30.

Power: built-in 220V.

Case: metal 1U, 482 x 250 x 44 mm.

Lines served:
Up to 32 FXS, up to 48 FXO.

Up to the 1st flow E1.
SIP Proxy up to 256 subscribers.
SIP, H.323 trunks – up to 30.

Lines served:
Up to 32 FXS, up to 48 FXO.
Up to 16 digital subscriber lines.
Up to 2 E1 streams.
SIP Proxy for 128 IP subscribers with the ability to expand to 256 subscribers.
SIP, H.323 trunks – up to 30.
SPRUT-7UX-TDM for 10 channels.

Power: External power supply 24V, 2.75A.

Case: metal compact, 190 x 230 x 60 mm.

Lines served:
Up to 4 FXO
SIP Proxy for 16 IP subscribers with the ability to expand up to 256 subscribers.
SIP, H.323 trunks – up to 30.

Power: Power supply with output voltage 5V, 2A, power 15W or PoE (Power over Ethernet) (optional).

Case: metal 146 x 104 x 28 mm

Lines served:
Up to 1st E1
SIP Proxy for 8 IP subscribers with the ability to expand to 256 subscribers.
SIP, H.323 trunks – up to 30.

Power: Power supply 220V or PoE (Power over Ethernet) (optional).

Housing: plastic, 1U, 432 x 175.6 x 43.6 mm

Features of digital IP-PBX "Agat UX"

An important advantage of the proposed solutions is the absence of restrictions on the type of devices used. The capabilities of the Agat UX IP-PBX provide for the installation of regular or IP telephones, SIP clients, DECT systems, etc. - depending on the requirements for the functionality and price of the devices.

The Agat-RT company has more than 20 years of experience in developing telecommunication systems and offers the most efficient, reliable and convenient automatic telephone exchanges.

The use of IP-PBX significantly (on average by 30-40%) reduces the cost of long-distance communication. Incoming calls are quickly addressed to a specific subscriber. All structural divisions of the company, including those located remotely, are united into a single telephone exchange IP-PBX “Agat UX”. An additional advantage is the ease of configuration and setup, which allows administration and maintenance in-house.

Purpose

Installing a telephone exchange provides not only complete telephone installation for the office, but also a whole range of modern additional services. For example, you can choose the least expensive communication channel. This feature of the proposed PBX allows you to optimize the cost of calls.

The ability to track the intended use of corporate telephony is also implemented. Using the Agat UX digital PBX, you can build a distributed network, as well as create a unified subscriber directory for employees of remote departments. This will allow them to call colleagues using internal numbers, eliminating the need to use external lines and, accordingly, the use paid services providers.

The capabilities of the IP-PBX "Agat UX" provide for the organization of conferences using the built-in server, which makes it possible to hold meetings online at the lowest cost of operating communication channels. The presence of an automatic dialing system ensures timely notification of personnel about emergencies and other abnormal situations.

An important feature of automatic telephone exchanges is the option to record conversations. For this purpose it is used professional program"Octopus-7". When an active connection is detected, it forwards a copy of the conversation to remote computer, and also opens up access to real-time listening. This allows additional control over employees, as well as providing a higher level of corporate security.

Digital telephone exchanges IP-PBX "Agat UX" are distinguished by good fault tolerance. This is due to the fact that system settings are stored in the memory of each device. Accordingly, there is no central link on whose work the functioning of the entire network would depend.

The ability to build decentralized systems that unite up to 10,000 users has been implemented. This allows the company to cover an unlimited territory with a single corporate network. In this case, the internal network represents for all users one large distributed PBX with a common number plan and short numbers. Calls within such a network are free for all subscribers, regardless of their distance.

General principles of telephone PBX operation

The Agat IP-PBX architecture is designed to process incoming calls via analogue PSTN lines, E1 streams (using E1 or FXO mezzanines) or through IP telephony channels via the built-in Ethernet port. Once received via any communication channel, the call is processed by internal software and redirected:

• to a specific subscriber;
• group of subscribers;
• to a robotic service that helps the subscriber determine the further route of the call.

When making outgoing phone calls, the PBX station determines the best available route. This could be an analog line, IP telephony or mobile communications via a connected GSM gateway.

Installing an IP PBX optimizes work and increases the efficiency of each employee and the company as a whole. Thanks to its low price and simple interface, the Agat system is available to any organization, regardless of its field of activity and number of employees.

Main advantages of IP-PBX “Agat UX”

• The offered PBXs are compatible with all existing communication channels. The system works with FXS, FXO, E1, GSM (via GSM gateways), H.323, SIP, ADASE. At the same time, the functionality of a digital PBX is fully available when working through any channel.
• Many additional services are supported: voice menu, fax, conference server, etc. Therefore, the Agat UX IP-PBX is a multifunctional solution that allows you to solve a range of related tasks: buy it, and you will not have to purchase peripheral equipment.
• Various options for automating incoming calls, which allows you to minimize the number of unprocessed calls. For convenient operation of the PBX station, employees create optimal algorithms for working with incoming calls, coordinating them with the needs of the company.
• Many related functions for each individual subscriber. These include voice mail, forwarding, receiving fax messages, personal telephone directory, missed call logging, ordering call back etc.
• Compatible with modern software. In particular, it is possible to integrate the Agat UX automatic telephone exchange with corporate information systems, CRM products, ERP-class complexes, etc.
• Variety of solutions offered. This allows you to order the “Agat UX” version, which is maximally adapted for use in a specific office. The number of subscribers varies from 2 to 10,000, there are budget offers for small companies and multifunctional branched solutions for large enterprises, corporations and holdings.
• High level of protection against unauthorized entry and unauthorized access. In particular, closed operating system and protocols for interaction with services.

Functionality

• Perform all modern operations for receiving and servicing incoming calls.
• Availability of a flexible mechanism for transporting and distributing calls (routing tables, dial plan). Broadcast and convert numbers automatically.
• Call distribution mode along alternate routes. This opportunity PBX (telephone exchange "Agat UX") is especially in demand in case of unstable operation and other problems in the operation of the main line.
• Monitoring the status of all network components using the SNMP protocol, as well as monitoring an unlimited number of working devices from a single interface.
• Support for fax communication using T.38 and T.30 protocols.
• Automatic license plate recognition using the built-in identifier. Not only the Russian Caller ID format is supported, but also CallerID (FSK/DTMF).

Head of the IT department of Topol-EKO Antonov Evgeniy: “I would like to note another important advantage of the IP PBX Agat UX - the ability to remotely configure and administer the station. Together with Agat RT engineers, we compiled a detailed terms of reference, according to which the same engineers pre-configured all our 11 automatic telephone exchanges in the Moscow office of the company. The automatic telephone exchanges were ready for work, our corporate network was deployed in full, as they say “on my desk”, we were able to test it, make sure that everything worked, prepared instructions for connecting lines for local specialists and sent the stations to branches. On the given day and hour, all 11 PBXs were connected to the corporate IP network, and we began using the new telephony for the entire company at once. Further adjustments to the settings were made by us from Moscow without any difficulties.”




Do you need a PBX for your organization? In this case, you have come to the right address. We will find the optimal solution (digital or IP-PBX, system and regular terminals for it, headsets for these terminals and more).

Today, most digital or IP PBXs on the market have very rich and diverse functionality, supporting both traditional TDM telephony (uses digital and analog signaling) and IP telephony (uses IP protocols SIP, H.323, MGCP ). At the same time, both digital and IP-telephony PBXs differ greatly in price and capabilities - so there is always a solution for every budget and taste.

Common types of PBX

Analog PBXs- this is not even the “past” century, but the “before last” century. Although they can still be found on sale (and even here!), there is hardly any point in using them - except for the purpose of absolutely minimizing the budget with extremely limited functionality.

Digital PBX- this is also a completely out-of-date solution, although it is quite widely used. The main advantage of such PBXs is a relatively low budget with very good communication quality and fairly rich functionality.

IP PBX- this is the most modern solution. They use the most current VoIP protocols - (SIP, H.323, MGCP). One (and quite significant) advantage of IP-PBX is that they do not require laying separate telephone wires, installing telephone sockets, etc., but are easily integrated into the company’s existing IT infrastructure. In addition, IP-PBX integrates well with various applications and programs, in particular, with CRM systems. For IP phones, such a PBX acts as a server on which they register.

Hybrid PBXs- this is the most popular option. Such PBXs allow you to connect analogue, digital and IP external lines, and both analogue (the most cost-effective solution) and digital and IP telephones as internal terminals. This makes it possible not for all employees to purchase new phones, but to use existing terminals

Why is it better to buy a digital or IP PBX from Viktel?

There are several main reasons why, after comparing many suppliers, they return to us. Namely:

• installation and configuration of PBX is our main specialization;
• our engineers of the highest category have been certified by equipment manufacturers and have all the necessary knowledge of automatic telephone exchange;
• everything you need to organize a corporate telephone network can be purchased at Viktel;
• We provide training for specialists from partner companies and clients in administering and setting up PBX.

In addition, we are ready to provide free and detailed consultation to each client on the selection of a telephone exchange and the purchase of the necessary associated equipment! Call and place an order!