FAQ

Unified communications (UC) is the integration of real-time communication services such as instant messaging (chat), presence information telephony (including IP Telephony), video conferencing, call control and speech recognition with non real time services such as unified messaging (integrated voicemail, email, SMS & fax). UC is not a single product, but a set of products that provides a consistent unified user interface and user experience across multiple devices and media types.

UC also refers to a trend to offer Business process integration i.e. to simplify and integrate all forms of communications in view to optimize business processes and reduce the response time, manage flows, and eliminate device and media dependencies.

UC allows an individual to send a message on one medium and receive the same communication on another medium. For example, one can receive a voicemail message and choose to access it through e-mail or a cell phone. If the sender is online according to the presence information and currently accepts calls, the response can be sent immediately through text chat or video call. Otherwise, it may be sent as a non-real-time message that can be accessed through a variety of media.

How to Introduce Unified Communications

1. Improve support for mobile workers
2. Bring telephony to the PC
3. Bring computer applications to the telephone
4. Establish Unified Messaging
5. Enterprise Instant Messaging Integration
6. Introduce Unified Conferencing
7. Add Video

The Session Initiation Protocol (SIP) is a signalling protocol used for establishing sessions in an IP network. You may be rapidly reaching the conclusion that everyone has too many devices, too many numbers, and too little time. Session Initiation Protocol is here to save the day. The SIP protocol is fast becoming the standard for complete service integration. Learn how the session initiation protocol can improve software communications architecture while simplifying your device-centric world.

Communications today are device-centric. Every device has its own phone number, address, or alias. The more devices people use, the more addresses you need to remember in order to reach them. And without presence, communication becomes a guessing game when trying to connect with people on whatever device they may be using.

With SIP, communications become user-centric once again. A SIP address of record or AOR, provides one unifying identifier that can be mapped across multiple devices and media types. With SIP, you no longer need to track multiple phone numbers, e-mail addresses, and IM contact names. SIP will do it for you.

Even more exciting, both users and applications can access presence information, providing the opportunity to create next generation converged communication applications. For example, your network can deliver new capabilities such as polite calling.

Based on information from her calendar application, voice calls to an executive can automatically be routed to an assistant if the executive is in a meeting.

VOIP is Voice over Internet Protocol. VoIP refers to a way to carry phone calls over an IP data network, whether on the Internet or your own internal network. A primary attraction of VoIP is its ability to help reduce expenses because telephone calls travel over the data network rather than the phone company’s network.

Encompasses the full suite of VoIP enabled services including the interconnection of phones for communications; related services such as billing and dialling plans; and basic features such as conferencing, transfer, forward, and hold. These services might previously have been provided by a PBX.

Business Phone System Smart Tips

Purchasing a phone system can be done through many ways.

• A smart way we recommend is by changing to a SIP phone Carrier. Huge savings are on offer. The savings made can easily cover the cost of a new phone system more
• A popular method has also been bundling your new phone system, in a rental contract with your phone carrier (for 3-5 years), with the promise of reduced call costs and call credits.
• Before deciding how to purchase, consider the expected life of the phone system. Ensure you are not buying a ‘run-out’ model, soon to be superseded
• You need to ensure that the system will cater for all growth of your business for the foreseeable future, say 5 years. Ensure the phone system has capacity to grow. And the functionality for your changing needs
• Is your Phone System Partner the right one for you? Do they have Technical Help Desks, Accredited, and Qualified Technicians? A big part of the purchase of a phone system is the aftercare support. That’s where you will get your value. You are not only buying a phone system but also technical expertise. Do they just sell one brand?
• Remote Support. Have your Phone System partner being able to access your phone system for remote changes will save both parties and ensure speedy service
• SIP enabled Phone Systems. Ensure your new phone system handles SIP phone calls
• Self Sufficiency. Ensure your new phone system enables you to make your own basic changes
• Unified Communications. Inquire how your business can benefit

The company has been owned by private equity firms Silver Lake Partners and TPG since a 2007 leveraged buyout for $8.2 billion. Being privately held means the company can act more strategically because it doesn’t have to respond to the quarterly expectations of Wall Street, says Avaya’s CEO

Major unified communications (UC) vendors including Alcatel-Lucent, Cisco, IBM, Microsoft, Shoretel, Siemens and others.

Avaya bought Nortel’s enterprise networking division and laid out a migration plan for bringing its VoIP/UC offerings in line with each other. Avaya has also set up a network infrastructure division to sell the former Nortel’s switches, wireless, security and other infrastructure hardware.

Avaya says it wants to advance UC via its software platforms, and is trying to draw customers of other telephony vendors into its fold. To do that, it is promoting interoperability of its call control server with Session Initiation Protocol (SIP) gear from other vendors, making it possible to bind together multivendor corporate networks immediately, and to transition over time to SIP-based UC.

Even for its current customers, Avaya is promoting the SIP-based Avaya Aura serve as a practical addition to existing voice networks that becomes the foundation on which to build UC networks as the need arises.

Beyond its private ownership, the company has been ranked No. 1or No. 2 in enterprise voice for years, and with the purchase of Nortel, which was reliably in the top four, it should be considered a top contender for grabbing customers and migrating them to embrace its UC vision.

The company is still in the first year of having bought Nortel’s enterprise division. This is a vulnerable time for any company that has made a sizeable acquisition, when it looks inward to organize and can lose focus on other top- and bottom-line aspects of the business.

The purchase of Nortel’s enterprise equipment division prompted Moody’s rating service to drop Avaya’s credit rating to B3 — one of the credit ratings commonly referred to as junk.

Avaya hangs its future on Avaya Aura, the company’s SIP-based communication software platform that will be sandwiched between communications infrastructure — such as PBXs — and services — such as voice, video, messaging, conferencing and mobility.

The company has hinted at a chameleon-like device that could perform many functions, some of them similar to what the iPad attempts to perform.

It has also announced a forthcoming data center switch that will be upgradable to support 40Gbps and 100Gbps Ethernet.

The Avaya IP Office’s default IP address is 192.168.42.1, and out-of-the-box, it only knows how to communicate with other devices in the same address range (192.168.42.x). Your PC probably has an IP address assigned outside this range. Program your PC to have a static IP address of 192.168.42.2 and try again.

There is a long list of mergeable and non-mergeable changes. A good rule of thumb is this: if you have ADDED something (ie. an extension, a hunt group, a new user.) you’ll have to reboot. If you’ve just CHANGED something (ie. button programming for a user, changed a short code, changed an incoming call route.) you can do a merge instead. You will automatically be prompted to merge or reboot in later releases of the IP Office software.

Make sure all of the following are in place in order to get your licenses to come up valid:

• Installation of Feature Key Dongle – Your feature key dongle must be securely connected to the PC that acts as your license server (often also the VoiceMail Pro server)
• You have installed the Feature Key Server software from the IP Office Admin CD on the PC with the dongle.
• The Key Server service is running on the key server PC (sometimes stopping and restarting this service helps).
• The correct IP address for the PC acting as the Key Server has been entered in Manager under System->License Server IP address.

Avaya IP Office Phone System Licenses are unique to a specific dongle. If you have multiple dongles, make sure you have the right licenses installed in Manager for the dongle you’re using.

Our lab test results show that Vonage Caller ID is not 100% compatible with the Avaya IP Office phone systems. Packet8 service, provided by 8×8 Inc. (NasdaqCM: EGHT), has proved to be a much better alternative to Vonage for voice over IP dial tone, as their Caller ID works so much more reliably when interfacing with the Avaya IP Office system in that it provides not only the callers’ name and number almost 100% of the time. On the other hand, Vonage tended to provide acceptable caller ID information, ie. callers’ phone number and name, once every ten calls at best.

Check the following:

• Make sure you have completed installation of the VoiceMail Pro software and the VoiceMail Pro Service is running on your voice mail server PC. Sometimes stopping and restarting the service will help.
• Make sure that your Voicemail Pro license is coming up as valid in Manager
• Make sure your voice mail server IP address is correctly programmed in Manager (set in systemàvoicemail tabàvoicemail server IP address. Make sure voice mail type is set to “PC” in this tab as well.)

You’re probably using an early version of the IP Office Manager that does not support Windows 2003 Server with VoiceMail Pro. Make sure you have upgraded to the latest Avaya IP Office software release.

Voice mail jitter can come from one of two things: an overloaded server and collisions between packets on your network. Check the following:

• If you are using a 403 base module, your network interface card in your Voice Mail Pro server PC needs to be set to 100/Half Duplex. This setting should also be used for any ports on 3rd party routers that the server and/or the IP Office are using as an uplink to the rest of your network.
• The voice mail server should be dedicated to voice mail and not running any other services. Do NOT install VoiceMail Pro on a server that is also running Exchange, hosting websites, or any other high-impact services.
• Make sure all screen savers, power save and hibernation modes are turned off.
• Anti-virus software can often impact the VoiceMail Pro service as well. Disable any background anti-virus software on the voice mail server PC.

You need to configure Phone Manager to log into the IP Office phone system to control your phone. The first time you run Phone Manager, do the following:

• Go to configureàPBX
• Enter the IP address of your IP Office in the PBX address field.
• When you enter the correct IP address, the Username box should automatically fill up with the names of every user in the system. If it does not, you have an incorrect IP address or your computer cannot communicate over the network with the IP Office.
• Select your name from the list and enter your user’s password where indicated. When you click ok, Phone Manager should come up, as “you” and all the buttons and menus will work.

Make sure that:

• You have valid licenses for Phone Manager Pro
• “Phone Manager type” is set to “Pro” under your user’s telephony settings in Manager.
• You don’t have more people trying to log in as Pro users than you have licenses for. If you have 5 Pro licenses, the 6th user to log in will get Phone Manager Lite, even if the above settings are correct.

Yes. However, the setting is not in Phone Manager. Open your phone system config in IP Office Manager, go to your lines area, and select a line. The “prefix” setting is designed to fix just this problem. Whatever you enter in the prefix area will pre pre-pended to the caller ID of every call that comes in on this line. Now the entire caller IDs in your logs will have 9’s in front of them and you can call them back using the “call number back” feature.

A short code can be programmed to support this functionality. Contact Mountain States Telecom for more information about this programming technique.

There are just a few simple steps to start dialing from Outlook:

• Make sure you have installed the TAPI drivers on your PC from the User CD.
• Make sure the TAPI driver is configured correctly. You can find the settings by going to the Windows control panelàphone and modem optionsàadvancedàAvaya IP Office TAPI2 service provider. (if you don’t see that last part, you have not installed the TAPI drivers).
• In the properties of the TAPI driver, make sure that “single user” is selected, and that the switch IP address, username, and password are identical to the settings you use under the configureàPBX window of phone manager.
• If you had to change settings in the TAPI driver, you must reboot your computer for them to take effect.

Once the driver is configured correctly, you have to tell a program from which you wish to dial to use the driver instead of your PCs modem. In Outlook, here’s how you do it:

• Open a contact you wish to dial. They must have at least one phone number field filled out.
• Click the auto-dialer button in the toolbar of the contact.
• Click the “dialing options” button
• Select “IP Office phone:” under the “connect using line” drop-down box.
• When you click “start call” your phone will automatically dial the contact number. Note that all rules about dialing 9 and/or 1 still apply.

We’ll make this easy – YES. Your Avaya IP Office system MUST have voice compression hardware in place in order to do any sort of VoIP telephony (IP phones and/or tying multiple IP Office units together.) All of the Avaya Small Office Edition control units come with VCM hardware built-in. On the Avaya 403, 406, and 412 models, you MUST purchase an additional VCM card in order to do Voice over IP functionality.

The confusion is arising around WHEN the system uses the VCM and when it does not. Depending on how you are going to use the system, we may be able to reduce the SIZE of your VCM, but not eliminate it all together. One of Mountain States Telecom’s knowledgeable engineers will be glad to help you determine how big a VCM you’ll need.

The phone looks for 2 files on the TFTP server every time it boots up: 46xxsettings.scr and 46xxupgrade.scr. By default, the upgrade script exists and the settings file does not. The phone doesn’t NEED the settings file to work, but it will give you a timeout error if it can’t find it anyway. To resolve this:

• On your TFTP server create a new file in c:\program files\Avaya\ip office\manager called 46xxsettings.scr.
• You can leave this file blank, or you can put in a line, starting with a “#” symbol, stating that the file is only there to get rid of timeout errors on the phones.

Your phone can’t communicate with the other end of the VPN tunnel – usually because of an incorrect “router” setting. Make sure that the IP address you are giving the phone for its router is the address of whatever device is handling your LOCAL end of the VPN tunnel.

Your phone can talk to the IP Office, but the IP office can’t talk back. This is due to incorrect or missing IP routes in the IP Office configuration. The actual IP routes you need to add will vary based upon your network topology, but a good rule of thumb is that you should ALWAYS have a default route:

• IP address = blank
• Mask = blank
• Gateway = address of router through which the IP Office can communicate with other networks (often this is the device handling the IP Office end of the VPN tunnel)
• Destination = LAN1

Chances are you are talking on two phones with a router between them. To fix the one-way talk issue:

• Open your phone system config in Manager
• Go to the extensions area
• Select the extensions in question. On each, go to the VoIP tab and make sure that “enable direct media path” is NOT checked.

This is one of the limitations of Voice over IP telephones. In the current state of the technology, group page simply does not work at this time.

Please call Mountain States Telecom at 719.635.0006 for support on the Small Office Edition.

We’ve seen this happen, too, and we’re not sure why. For this reason, Mountain States Telecom highly recommends that you download the free Avaya TFTP Suite Pro application and use THAT as your TFTP server application instead of leaving a Manager window open all the time.

Yes. While its true that you could probably get by without doing QoS on most small networks with just a few phones and minimal data traffic, the truth of the matter is that the 3616 and 3626 look for the SVP as part of their boot sequence. If they don’t see an SVP on the network they will give you a “No SVP found” error and die. The phones WILL NOT BOOT without an SVP visible on the network, period.

Paging systems are tricky. You will almost always need some sort of an interface unit to go between the IP Office and your paging amplifier. The best solution is to purchase a Universal Paging Access Module (UPAM) from Avaya. However, there is one trick you can try:

• Connect the POT extension on the IP Office you wish to use to a simple telephone cord splitter (available at Radio Shack, etc)
• Connect one of the outputs of the splitter to the input on your paging amplifier.
• Connect the other end of the splitter to a cheap, single-line analog telephone. Leave this phone OFF HOOK ALL THE TIME.

Yes. This is a common complaint when using a PRI. Open your telephone system configuration file, in Manager, go to the ‘system’ area, click the ‘telephony’ tab. These two values will be of interest to you:

• Dial Delay Time: This is the amount of time that the system will wait after the last key you press before attempting to interpret what you dialed. Increasing this value will make the system give you more time to dial a phone number.
• Dial Delay Count: This is the number of key presses the system will wait for before trying to interpret what you’ve dialed.

The optimal settings for these two values will depend on your environment (how your T1 works, whether you have to dial 10 digits on all outside calls or not) so there is no “correct” setting for all cases. With some simple trial-and-error, however, you will be able to use these two settings to optimize your dialing.

There are several places that people can log in and out of a group, and they will conflict with each other if set differently. You can enable/disable a user’s membership in a hunt group in any one of the following places:

• The “agent mode” tab in Phone Manager Pro.
• The properties window for a hunt group in Manager.
• Via a programmed soft key on the phone.
• Through the menu on the 44xx or 46xx series phones. (Avaya 4412, 4424 system telephones, the 5410, 5420 digital telephones and 5610 and 5620 IP telephones.)

That last item is often the problem and is the last place most people look. Press the menu button on your 44xx, 46xx, 54xx or 56xx-series phone and use the arrow keys to scroll over until you see “group” on the screen. Make sure there is a ” ^ ” character over the word “group.” If not, press the button under “group” to make the character appear. Otherwise, the phone will constantly be trying to log out of whatever groups it’s user belongs to.

The bandwidth used varies depending on the compression method chosen. Avaya IP Office supports a wide range of compression standards, including the most popular G.723.1 and G.729a. These will occupy approximately 10K and 13K of bandwidth respectively.

Try to keep the overall end-to-end delay to 150 milliseconds or below. An idea of the delay inherent in the network can be measured by carrying out a ping test and dividing the result by two. IP Office has built in echo cancellation to maximize speech quality.

Garble, clipping and some distortion quality problems are symptoms of variable delay and or packet loss. Variability in the delays of traffic is called jitter. Jitter and packet loss may be the result of switches and routers that are either faulty or working outside their design intentions. Avaya IP Office provides jitter buffers to compensate for a moderate amount of jitter found in networks. Voice traffic is quite tolerant of small amounts of packet loss so in most cases this may be ignored. Where packet loss is excessive (greater than 2% say) the cause should be established and fixed. This could be due to a fault or simply an over worked device discarding packets. Significant packet loss can cause perceptible losses in speech, to the extent that no speech may be heard either in one or both directions.

Each time speech is converted into a digital signal and back again, tiny difference from the original creeps in. The more times this happens on a single call, the bigger those differences can become. These differences can become perceptible as distortion. Ideally, the path speech takes should only require one ‘analog to digital to analog’ conversion and this will be the case in many instances. Exceptions to this occur when making calls to mobile telephones or voice mail systems where the analog to digital to analog conversion may occur twice (once on IP Office and once on the mobile network, etc). Different encoding methods will have different effects. Avaya IP Office supports a range of encoding methods to allow you to choose the one with the right quality versus bandwidth for your network. In general multiple conversions should be minimized wherever possible.

Delay in a network originates from a number of different sources and phenomena. A primary source of delay is the process of converting speech to VoIP traffic. The Avaya IP Office supports a number of standards based encoding methods to allow the optimum trade off between quality and bandwidth to be made. IP Office incorporates integral echo cancellation to minimize the effect of echo introduced in the Voice over IP conversion process.

Another source of delay comes from data and voice traffic queuing at the ports of switches, routers, gateways and or bridges that make up the network. It is possible that the traffic queuing at a port is minimal and no action needs to be taken. This would be the case if the available bandwidth far exceeded the demand. To overcome queuing bottlenecks in the network, IP Office prioritizes voice traffic using a standard known as DiffServ. This marks each IP packet carrying voice with a flag so that routers, etc. can force packets containing voice to the front of the transmission queue. An alternative method of prioritization that can be used by switches and routers, with an equally satisfactory result, is to look at what protocol is being used and prioritize this. All voice traffic is carried using two easily identifiable protocols, RTP and RTCP. Both methods are equally good, choose whichever method is the most cost effective and easiest to implement and manage.

A similar source of delay can be attributed to specific network nodes that convert from one network medium to another. For example T1 trunk lines may be carried across a high speed DSL like connection and converting from the high speed link back to T1 in the access gateway takes time to perform. Any Voice over IP ( VoIP ) traffic being carried through this link is therefore subject to the delay introduced by this conversion step. The delay may be minimized by ensuring that an appropriate QoS mechanism is enabled in the gateway to prioritize the Voice over IP traffic. IP Office incorporates integral echo cancellation to help minimize the effect of this kind of delay introduced by the network.

Delay can also be introduced as a consequent of collisions occurring on particular segments of the LAN. Collisions result when two devices on a shared switch port or segment try to transmit simultaneously. This causes all devices to stop transmitting for a period of time. This is a normal characteristic of many older Ethernet networks and, if occasional, may pass unnoticed. The more devices sharing a switch port, and the busier they are, the greater the opportunity for collisions. This is simply resolved by reducing the number of devices on each port, or by dedicating a port to each Voice over IP device. If you are just using Voice over IP telephony to link two IP Offices together, it’s well worth dedicating a port to each IP Office and router at either end of the link as the cost implications are likely to be minimal. In this regard it is important to dimension a network to cope with existing traffic demands as well as any future increases in traffic carrying capability.