Book Image

Building Enterprise Ready Telephony Systems with sipXecs 4.0

Book Image

Building Enterprise Ready Telephony Systems with sipXecs 4.0

Overview of this book

Open source telephony systems are making big waves in the communications industry. Moving your organization from a lab environment to production system can seem like a daunting and inherently risky proposition. Building Enterprise Ready Telephony Systems with sipXecs delivers proven techniques for deploying reliable and robust communications systems. Building Enterprise Ready Telephony Systems with sipXecs provides a guiding hand in planning, building and migrating a corporate communications system to the open source sipXecs SIP PBX platform. Following this step-by-step guide makes normally complex tasks, such as migrating your existing communication system to VOIP and deploying phones, easy. Imagine how good you'll feel when you have a complete, enterprise ready telephony system at work in your business. Planning a communications system for any size of network can seem an overwhelmingly complicated task. Deploying a robust and reliable communications system may seem even harder. This book will start by helping you understand the nuts and bolts of a Voice over IP Telephony system. The base knowledge gained is then built upon with system design and product selection. Soon you will be able to implement, utilize and maintain a communications system with sipXecs. Many screen-shots and diagrams help to illustrate and make simple what can otherwise be a complex undertaking. It's easy to build an enterprise ready telephony system when you follow this helpful, straightforward guide.
Table of Contents (17 chapters)
Building Enterprise-Ready Telephony Systems with sipXecs 4.0
Credits
About the Author
About the Reviewer
Preface
Glossary

Appendix A. Glossary

Analog Phone: Analog phones are usually the same sort of phone you might find in a residence. They can provide signaling to the PBX for special functionality by flashing the hook switch and utilizing different DTMF codes.

Automatic Call Distribution: This can be thought of as intelligent hunt groups. They allow phone system users (agents) to sign in and out of calling queues. Calls then ring agents based on different factors such as who is the first person in the ACD list or which agent has been idle the longest.

Basic Rate ISDN (BRI): A cheaper option than PRI, BRI may be offered in some areas. A BRI has two 64 Kbps B channels and a single 16 Kbps D channel for signaling. In the UK a BRI may also be called ISDN2e.

Blind Call Transfer: In a blind (also referred to as unattended) transfer, the call is simply transferred to the selected extension.

Bootrom (Boot ROM): Software that controls the booting of a device such as a phone or gateway.

Busy Lamp Field (BLF): This field show the status (or presence) of a call on a remote phone.

Call Detail Records (CDR): Information about calls that route through a phone system.

Call Forwarding: This service allows a user (or the phone system) to have a call redirected to another extension or number.

Call Hold: When a user presses a button for call hold on his or her phone, the caller is set into a mode such that neither party can hear the other.

Call Park Orbit: This was designed for PBX systems where the concepts of phone lines to users don't exist. Putting a call into a park orbit is accomplished by transferring a call to a holding queue (orbit). That call can be retrieved on any phone by dialing a pickup code and the park orbit number.

Call Pickup: This is the ability of one user to pick up another user's ringing phone.

Call Routing: How phone calls are handled and transferred through a phone system. Call routing logic evaluates calls and directs them (referred to as switching) to where they need to go based on many different factors. These factors include, but are not limited to, what number was dialed, who dialed it, and what time of day it is.

Call Transfer: This is the ability of a user to send a phone call to another extension on the phone system.

Caller-ID: Telephone number or name of caller that is passed in to a phone system.

Circular Hunt Group: This is a hunt group that "remembers" the last number that answered ringing and begins ringing on the next number in the list. When the end of the list is reached, it wraps around to the first number in the list again.

Clustering: A cluster is a collection of servers working together to act like a single system to provide high availability and load balancing.

Comma Separated Values (CSV): This is a data format typically used to exchange tabular data.

Compression Decompression Algorithm (Codec): This is a method for compressing and uncompressing media (voice / video).

Conferencing: A conference is a call between three or more parties. A conference may be a simple phone-based multi-party conversation or may be hosted by a full-featured conference server.

Consultative Call Transfer: In a consultative (also referred to as attended or supervised) transfer, the called party confers with the party that they will transfer the call to before the call is transferred.

Demarc: The demarcation point is the location in a facility at which communications facilities owned by the telecommunications provider interface with your organization's communications systems.

Dial Plans: These provide the routing logic for inbound and outbound calls from the system. The dial plans evaluate the dialed numbers, looking for patterns of digits, and direct calls to different destinations.

Dial Tone Multi Frequency (DTMF): This is the sound you hear when you push the dial pad buttons of a phone.

Digital Enhanced Cordless Telecommunications (DECT): This is a wireless standard for connecting communications equipment (headsets/phones).

Digital Phone: Digital phone sets provide higher functionality and programmability for phone systems than analog phones. They are proprietary to each vendor and type of phone system. Digital sets can be programmed centrally, provide excellent call quality, and usually have many buttons that can be programmed to provide different functionality to the user.

Direct Inward Dial (DID): Phone numbers assigned by a telecommunications provider that can be passed in to a phone system before the call arrives at the phone system.

Direct Station Selection (DSS): This is a one-touch speed dial assigned to a key on a user's telephone.

Domain Information Groper (DIG): This is a program, like NSLookup, that queries DNS servers.

Domain Naming System (DNS): This is a naming system for computers, services, or any resource participating in an IP network.

Dynamic Host Configuration Protocol (DHCP): This is a network protocol that allows network devices to obtain information about a network and network addressing.

E1: A physical layer protocol, much like Ethernet, that defines a 2 Mbps pipe for data or communications connectivity. This pipe can be used for data, split into 32 64 Kbps communications channels, or a mixture. If the pipe is used for communications channels, 30 of the channels can carry telephone conversations and the remaining 2 carry signaling and timing information. E1 circuits are typically found in Europe.

Firmware: Software that controls the operation of a device such as a phone or gateway.

Foreign Exchange Office (FXO): FXO lines are analog interfaces that connect to plain old telephone service (POTS) lines provided by the telecommunications provider.

Foreign Exchange Station (FXS): FXS lines are analog interfaces that connect to analog stations (phones or fax machines).

Fully Qualified Domain Name (FQDN): This is the combination of a DNS host name and a DNS domain name. For example with the FQDN of www.sipfoundry.org, www is the host name and sipfoundry.org is the domain name.

Ground Start Analog Circuit: Ground start circuits provide disconnect notification by actually grounding the circuit (when a caller hangs up the phone, also called answer and disconnect supervision).

Hard Phone: This is a physical telephone set.

Hunt Group: This is a collection of extensions that ring in a particular order when the hunt group number is dialed. The hunt group number is often referred to as the pilot number of the hunt group.

Intercom: The intercom function in a phone system allows a single user to dial another user's extension, have the receiving user's phone automatically go "off-hook" in speaker phone mode and have the two parties converse.

Internet Telephony Service Provider (ITSP): This is a company that provides telecommunications services over the Internet.

iPBX: This is a modern PBX (also referred to as a Softswitch). This name is derived from the fact that the PBX functionality is all accomplished in software running on a standard server.

ISDN: A more modern circuit signaling protocol that was designed to overcome problems that existed with T1 and E1 circuits. On E1s, EuroISDN signaling is standard. On T1s different providers utilize different standards. NI1, NI2, DMS100, and DMS250 are all examples of ISDN signaling protocols, each delivering different levels of functionality.

ISO Image: This is an archive file (or disk image) of an optical disk (CD / DVD / and so on).

Key Telephone System: This is a type of phone system that allows users to directly select outside lines through keys on the handsets. These systems were designed with smaller organizations in mind.

Linear Hunt Group: A hunt group that always starts ringing the first extension in the list and ends ringing the last in the list of extensions.

Localization: The ability of a system to localize its text and voice prompts for different regions of the world.

Loop Start Analog Circuit: Loop start analog circuits are the more typical home and key system phone lines. Loop start lines use either a polarity reversal (called battery reversal) or removal of the line voltage (battery drop) for answer and disconnect supervision.

Managed Gateway: A SIP gateway for which sipXecs can create configuration files.

Managed Phone: This is a SIP phone for which sipXecs can create configuration files.

Message Waiting Indicator (MWI): This is an important but seemingly simple responsibility of the voicemail system is to signify to users that they have messages waiting. This notification usually takes the form of a message waiting indicator (MWI) light that is lit on handsets.

Music on Hold (MoH): This is the music that is played to callers when their line is placed on hold.

Network Address Translation (NAT): This is a function of firewalls and routers that allows private internal IP address to be translated to one or more valid Internet IP addresses.

Network Interface Device (NID): This is a box that the telecommunications provider utilizes to break-out their lines to interfaces the customer can utilize.

NSLookup: This is a program that can query DNS servers to determine address and DNS information.

Paging: Similar to intercom functionality, paging differs in that it is designed to allow a single user to broadcast a message to one or more phones without the ability for the receiving phones to talk back to the caller.

Phantom User: This is a user account on the system that will never have a User Agent (phone) registered to it. Phantoms can be used for voicemail-only mailboxes or for call routing purposes.

Ping: This is a program that allows systems to test if other systems are available across a network.

Plain Old Telephone Service (POTS): This is another term for analog telephone lines.

Power over Ethernet (PoE): This is a method to provide DC power to various devices (such as phones and wireless access points) across network cabling.

Primary Rate ISDN (PRI): A PRI is an E1 or T1 with ISDN signaling running on top of it. ISDN signaling provides reliable call setup and tear-down detection as well as detailed information about each call. In the UK a PRI is also referred to as ISDN30. Voice channels on a PRI are referred to as B channels and the signaling channels are referred to as the D channel. On an E1, a PRI will provide 30 B channels of voice and utilize one of the signaling channels as the D channel. Since T1s have no signaling channels, a PRI on a T1 will utilize one of the channels as a D channel and have 23 B channels for voice.

Private Branch Exchange (PBX): A PBX is a type of phone system that is typically found in larger organizations.

Public Switched Telephone Network (PSTN): This is a telecommunications connectivity provided by a phone company.

Quality of Service (QoS): Information contained within Ethernet packet headers that signals network equipment should give packets priority over other packets.

Session Border Controller (SBC): This allows SIP calls to traverse network boundaries (firewalls). Because SIP traffic contains IP-specific information, SIP messages need to be modified when they traverse a NAT'd (Network Address Translation) network connection.

Session Description Protocol (SDP): This is a messaging format used to describe streaming media formats. SIP utilizes SDP for setting up sessions.

SIP: This is an Internet Engineering Task Force (IETF) standard protocol user for conducting interactive communications. SIP can be utilized for many forms of communications sessions including voice, video, and chat. The SIP call signaling is independent from the media sessions it controls.

SIP Gateway: The SIP gateway provides communications system connectivity to the telecommunications providers. A gateway may be a physical device connecting a traditional type phone circuit, discussed earlier, or a software-based gateway providing connectivity to Internet Telephony Service Providers (ITSP).

SIP Proxy: This can be thought of as a call router. Its job is to direct SIP calls through the system. The proxy itself does not handle any voice traffic (media).

SIP Trunk: This is a term used to describe the method for providing telephone service over the Internet.

Speed Dial: Speed dials are short phone numbers that can be dialed in order to dial a more complicated number. For example a user would dial 752 and the phone system would actually dial 18005555555.

SRV Record or Service Record: This is a type of record found in DNS domains that specifies information about available services.

T1: Similar to an E1 but it is common in North America. T1s are 1.544 Mbps pipes that can carry data or 24 telephone channels. There are no signaling channels on a T1. Also, like an E1, T1s can be channelized and utilized to deliver voice and data.

Unified Messaging System: Unified messaging systems are an extension of voicemail systems that allows users to have a single inbox combining voicemail, email, and faxes. A true unified system will integrate these systems at the server level such that when you open or delete voicemail on a computer it is marked as read or deleted in the voicemail system. A simple version of unified communications involves SMTP forwarding of voicemail to an email or requires setup of client software that handles email integration on the user's computer.

Uninterruptable Power Supply (UPS): This is a system that typically employs batteries to maintain power to devices.

Unmanaged Gateway: A SIP gateway that must be manually configured for use with sipXecs.

Unmanaged Phone: A SIP phone that must be manually configured for use with sipXecs.

Virtual Local Area Network (VLAN): This is an Ethernet standard that allows multiple virtual networks to be defined within network equipment.

Voicemail System: These systems provide auto attendant functions and the playing and recording of messages. The voicemail system can be thought of as the voice of the phone system.

Wireless Phone: This is a telephone that does not need to be corded to a phone system. These phones connect either via WiFi or by DECT.