Phone Service Frequently Asked Questions

Documents necessary to keep your existing phone number

To Keep Your existing phone number with Aannex phone services :

  • Download and print this authorization form for porting your existing number. Upon completion of this form, please send it to Aannex.
    ( NOTE : Without this document the order will not be processed, to port your existing phone number )
  • Send a copy of your recent phone bill to Aannex.
    ( NOTE : Without this document the order will not be processed, to port your existing phone number )

30 Day Money Back Gurantee

We offer a 30-day Money Back Guarantee ("MBG") if you terminate your service within 30 days from the date of your order. Your Order Date is the date you order service or the date we successfully process your payment, whichever is later. The MBG applies to the first line only and cover the monthly charge, taxes and or activation fees. No Shipping refundable, and administrative fees ( $10 to $15 will be charged based the plan) Additional lines are not eligible. Return ATA device return as good it was received. Any Long distance charges must be deductable from the total amount.

Our MBG does not cover any charges for international calls or usage outside of your service plan, calls to AANNEX toll free numbers, any monthly charge for Extensions and features or services not expressly included in your monthly plan fee. In addition, not all of the taxes that you paid may be refundable

Getting Started with Voicemail

1. To connect to voice mail, you can use any of the following methods:

  • Dial *40 from a phone connected via your VoIP device, then enter the assigned password.
  • Dial *42 from a phone connected via your VoIP device if you have multiple accounts, then enter the VoIP Phone number, followed by the assigned password when prompted.

You can also log in remotely from any phone by dialing your phone number and entering * during the greeting playback, followed by the assigned password when prompted.

2.Once you have logged into the voicemail system you can use the menu options below to check messages, manage messages or set personal greetings and preference options.

The voicemail system has standardized menus that operate much like the voicemail systems available with your mobile service, so the navigation may already seem familiar to you.

The buttons in darker blue below are top level menu options and the buttons in lighter blue are options available while in one of the top menu sections. A complete hierarchy of the menus is available on the following page.

How to use Phone Feature : Call Forwarding

Not at Home ? Have all your calls forwarded to another line. To setup "Call Forwarding":

  • 1. Simply log into your eCare Portal account and select call Preferences and choose the options for FORWARD features. Use any option that you like to forward your call accordingly.
  • 2. Select "call Forward" as the Forwarding Type.
  • 3. Select "unconditional Forward To" and Type the number you would like your calls forwarded to. (ie. Cell phone).
  • 4. Click Save.

Or you can use the quick "Call Forward" Menu :

To Activate To De-Activate
Call Forward All *72 + phone number *73
Call Forward on Busy *90 + phone number *91
Call Forward No Answer *92 + phone number *93

How to use Phone Feature : 3 way Calling

Connect two parties onto the same call so that all three of you can talk at the same time.

  • Establish a call with the first party
  • Press the telephone switch-hook or flash button to get a second dial tone
  • Dial the telephone number of the second party
  • Once the second party is connected, press the telephone switch-hook or flash button to conference all parties together

What is VOIP ?

Voice over IP is the same as Voice over Internet Protocol, and is better known as VOIP.

Voice over IP refers to the diffusion of voice traffic over internet-based networks. Internet Protocol (IP) was originally designed for data networking and following its success, the protocol has been adapted to voice networking.

Voice over IP (VOIP) can facilitate tasks and deliver services that might be cumbersome or costly to implement when using traditional PSTN:

  • More than one phone call can be transmitted on the same broadband phone line. This way, voice over IP can facilitate the addition of telephone lines to businesses.
  • Features that are usually charged extra by telecommunication companies, such as call forwarding, caller ID or automatic redialing, are simple with voice over IP technology.
  • Unified communications are secured with voice over IP technology, as it allows integration with other services available on the internet such as video conversation, messaging, etc.

These, and many other advantages of voice over IP, are making businesses adopt VOIP Phone systems at a staggering pace.

What is a VoIP Telephone ?

A VoIP telephone, also known as a SIP Phone or a softphone, allows the user to make phone calls to any softphone, mobile or landline by using voice over IP (VOIP). This way the voice is carried through the internet instead of the traditional PSTN system.

A VoIP telephone can be a simple software-based softphone or a hardware device that looks a lot like a common telephone.

Some of the common features of a VOIP Telephone are: caller ID, call park, call transfer and call hold.

3CX has developed a completely free VOIP telephone that can secure significant savings on telephone bills in a very simple way: all that the user requires is a broadband connection (DSL or cable), a connection to a VOIP phone or a SIP server, and a headset with a microphone and/or a sound card.

VOIP Definitions

  • VoIP - Voice over Internet Protocol (also called IP Telephony, Internet telephony, and Digital Phone) - is the routing of voice conversations over the Internet or any other IP-based network.
  • SIP - Session initiation protocol - is a protocol developed by the IETF MMUSIC Working Group and proposed standard for initiating, modifying, and terminating an interactive user session that involves multimedia elements such as video, voice, instant messaging, online games, and virtual reality.
  • PSTN - the public switched telephone network - is the concentration of the world's public circuit-switched telephone networks, in much the same way that the Internet is the concentration of the world's public IP-based packet-switched networks.
  • ISDN - Integrated Services Digital Network - is a type of circuit switched telephone network system, designed to allow digital (as opposed to analog) transmission of voice and data over ordinary telephone copper wires, resulting in better quality and higher speeds, than available with analog systems.
  • PBX - Private Branch eXchange (also called Private Business eXchange) - is a telephone exchange that is owned by a private business, as opposed to one owned by a common carrier or by a telephone company.
  • IVR - In telephony, Interactive Voice Response - is a computerised system that allows a person, typically a telephone caller, to select an option from a voice menu and otherwise interface with a computer system.
  • DID - Direct Inward Dialing (also called DDI in Europe) is a feature offered by telephone companies for use with their customers' PBX system, whereby the telephone company (telco) allocates a range of numbers all connected to their customer's PBX.
  • RFC - Request for Comments (plurals Requests for Comments but RFCs) is one of a series of numbered Internet informational documents and standards very widely followed by both commercial software and freeware in the Internet and Unix communities.

What is SIP ?

SIP is the internet standard for real time voice and video communication. SIP (Session Initiation Protocol) was developed by the IETF and published as RFC 3261.

SIP is an internet protocol for live communications used in setting up and tearing down voice or video calls. It is a signaling protocol used to create, modify, and terminate sessions with one or more participants in an IP network. A session can be a straightforward two-way phone call or it can be a multi-media conference session with many persons participating. SIP has made possible an array of services that seemed unthinkable just a few years ago: internet conferencing, IP telephony, instant messaging, presence, voice and video communication, data collaboration, online gaming, application sharing, and much more.

SIP is doing for real-time communications what HTTP did for the web and SMTP for email. It is the main driver in the acceleration of the IP Telephony revolution. With SIP Telephony, a viable alternative to traditional PBX has emerged. SIP telephone systems deliver features that enhance users' mobility and productivity, while securing substantial cost-saving advantages. This is making proprietary hardware based PBXs obsolete.

What is SIP forking ?

SIP forking refers to the process of "forking" a single SIP call to multiple SIP endpoints. This is a very powerful feature of SIP. A single call can ring many endpoints at the same time. With SIP forking you can have your desk phone ring at the same time as your softphone or a SIP phone on your mobile. For example, you would use SIP forking to ring your deskphone and your iphone at the same time, allowing you to take the call from either device easily. No forwarding rules would be necessary as both devices would ring. In the same manner SIP forking can be used in an office and allow the secretary to answer calls to the extension of his/her boss when he is away or unable to take the call.

AANNEX Phone System full supports SIP forking.

What is a SIP server ?

A SIP Server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network. A SIP server is also referred to as a SIP Proxy or a Registrar.

Although the SIP server is the most important part of the SIP based phone system, it only handles call setup and call tear down. It does not actually transmit or receive any audio. This is done by the media server in RTP.

What is an auto-attendant ?

Auto-attendant (or automated attendant) is a term commonly used in telephony to describe a voice menu system that allows callers to be transferred to an extension without going through a telephone operator or receptionist. The auto-attendant is also known as a digital receptionist.

For a caller to find a user on a phone system, a dial-by-name directory is usually available. This feature lists users by name, allowing the caller to press a key to automatically ring the extension of a user once his/her extension is announced by the auto attendant.

If a user is not available, the auto-attendant directs callers to the appropriate voice mailbox of the user to leave a voicemail message.

Having an auto-attendant in a phone system is a very useful and cost-effective feature for a business, as it replaces/helps the human operator by automating and simplifying the incoming phone calls procedure.

AANNEX includes a FREE auto-attendant feature.

What different types of CODECS are there ?

A Codec converts an analog signal to a digital one for transmission over a data network. The following Codecs are in use today :

  • GSM - 13 Kbps (full rate), 20ms frame size
  • iLBC - 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
  • ITU G.711 - 64 Kbps, sample-based. Also known as alaw/ulaw
  • ITU G.722 - 48/56/64 Kbps
  • ITU G.723.1 - 5.3/6.3 Kbps, 30ms frame size
  • ITU G.726 - 16/24/32/40 Kbps
  • ITU G.728 - 16 Kbps
  • ITU G.729 - 8 Kbps, 10ms frame size
  • Speex - 2.15 to 44.2 Kbps
  • LPC10 - 2.5 Kbps
  • DoD CELP - 4.8 Kbps

What is DID ( Direct Inward Dialing ) ?

DID - Direct Inward Dialing (also called DDI in Europe) is a feature offered by telephone companies for use with their customers' PBAX system, whereby the telephone company (telco) allocates a range of numbers associated with one or more phone lines.

Its purpose is to allow a company to assign a personal number to each employee, without requiring a separate phone line for each. That way, telephony traffic can be split up and managed more easily.

DID requires that you purchase an ISDN or Digital line and ask the telephone company to assign a range of numbers. You then need DID capable equipment at your premises which consists of BRI, E1 or T1 cards or gateways.

What is ECHO cancellation ?

Echo cancellation is the process of removing echo from a voice communication in order to improve the voice call quality. Echo cancellation is often needed because speech compression techniques and packet processing delays generate echo. There are 2 types of echo: acoustic echo and hybrid echo.

Echo cancellation not only improves quality but it also reduces bandwidth consumption because of its silence suppression technique.

What does ENUM mean ?

ENUM stands for Telephone Number Mapping. Behind this 'abbreviation' hides a great idea: To be reachable anywhere in the world with the same number - and via the best and cheapest route. ENUM takes a phone number and links it to an internet address which is published in the DNS system. The owner of an ENUM number can thus publish where a call should be routed to via a DNS entry. What's more, different routes can be defined for different types of calls - for example you can define a different route if the caller is a fax machine. ENUM does require the phone of the caller to support it.

You register an ENUM number rather like you register a domain. At present many registrars and VOIP Providers are providing this as a free service.

ENUM is a new standard, and is not that widespread yet. Though it looks set to become another revolution in communications and personal mobility.

How does FAX work in VOIP environment ?

FAX was designed for analog networks, and does not travel well over a VOIP network. The reason for this is that FAX communication uses the signal in a different way to regular voice communication. When VOIP technologies digitize and compress analog voice communication it is optimized for VOICE and not for FAX. Subsequently, there are a number of things you need to take note of when you move to a VoIP Phone System.

If you want to continue using your old fax machine, and you want to connect to your VoIP phone system, its best to use a VOIP GATWAY and an ATA that supports. T38 is a protocol designed to allow fax to 'travel' over a VoIP network.

It is also possible to convert to computer based fax and choose a VoIP phone system that supports fax. AANNEX Phone System for Windows includes a full feature FAX that receive faxes and forward them in PDF format to e-mail. Faxes can be sent from anywhere in the network using the Microsoft FAX client and Fax server(which comes free with Windows Server 2003 and 2008)

Another way to deal with fax when you switch to VOIP are to connect the fax machine directly to the existing analog phone line and bypass your VOIP system.

What is FOIP - Fax over IP ?

FOIP stands for Fax over IP and refers to the process of sending and receiving faxes via a VOIP network.

Fax over IP works via T38 and requires a T38 capable VOIP Gateway as well as a T38 capable fax machine, fax card or fax software. Fax Server software that can talk 'T38' allows sending and receiving faxes directly via a VOIP gateway and, consequently, does not need any additional fax hardware.

AANNEX includes a T38 compatible network Fax server in its. Faxes are converted to PDF files and forwarded via email. Outbound faxes are sent via Microsoft Fax from anywhere in the network. Other fax servers currently in the market require the use of separately licensed and expensive Dialogic SoftIP drivers.

What is RTCP - Real Time Transport Protocol ?

RTCP stands for Real Time Transport Protocol and is defined in RFC 3550. RTCP works hand in hand with RTP. RTP does the delivery of the actual data, where as RTCP is used to send control packets to participants in a call. The primary function is to provide feedback on the quality of service being provided by RTP.

What is SDP - Session Description Protocol ?

SDP, short for Session Description Protocol, is a format for describing streaming media initialization parameters. It has been published by the IETF as RFC 4566. Streaming media is content that is viewed or heard while it is being delivered.