Inter-tel has tried to thwart this problem by making a flag in the system that will force the user to have to hit another key first before entering the feature code.  This key is known as the ‘Special key‘.  It is a key with an infinity symbol on it (sideways 8, lazy 8).

I will explain SOME of the more popular feature keys below and possible symptoms:

• ACD Agent Login/Logout (326-328) – Used to login (326), logout(327), or toggle (328).  You’ll know if you’re logged out because your phone will not be ringing anymore in your respective hunt group.Automatic CO (Trunk) Access On/Off (360) – If this is off you will pick up the handset and still have to hit the flashing line key on incoming calls to answer the call.  Very annoying and common.

  Automatic Intercom Access On/Off (361) - Same as above but with intercom calls.

  Background Music On/Off (313) – Makes the on hold music come out of your phone.  (Hey, maybe you like your music on hold).

  Call Forward – All Calls (355) - This allows you to manually call forward your phone.  This will override system call forwarding.

  Call Forward – If Busy (357) - This allows you to manually call forward your phone only when you are on the phone.  This will override system call forwarding.

  Call Forward – If No Answer (356) - This allows you to manually call forward your phone after only after it has given you a chance to answer the phone first.  Usually 4 rings is the norm although it can be shorter or longer.

  Call Forward – If No Answer/Busy (358) – Combines the two previous conditions.

  Clear System Alarm (9850) – Sometimes there will be an alarm on the administrator’s phone.  (This is programmable, but usually the receptionist)  This will clear aforementioned alarm.  Can only be done from an administrator’s phone.

  Default Station (394) - When all else fails and your phone is acting funky use this feature code.  Most of the time it will fix most of the user caused issues.

  Display Time/Date (300) - If you have a message or something else on your screen but want to see the time and date or what extension you’re on this is your code.  The LCD will show you what you need to see for 5 seconds or so before reverting back.  Used a lot by techs in the field.

  Do-Not-Disturb (370-372) – turn it on (370), turn it off (371), toggle (372).  If it is on then no calls will come through.  Assuming your system forwarding is on, the call will usually go straight to voicemail.

  Handsfree On/Off (319) – When this is on and someone calls you the call will automatically connect and come through your speaker.  Some people love it because they don’t have to quit typing or what have you and can instantly talk to whoever is there.  Some people hate it because it can be a little intrusive.  Personal preferences vary.

  Headset On/Off (317) - VERY VERY common trouble call.  A customer’s speaker phone doesn’t seem to be working and there will be no dial tone on the handset.  The phone is obviously broken…… nope….. headset mode is on.  Very easy to fix and save yourself a headache.  Did I mention this is VERY VERY common.

  Hunt Group Remove and Replace (322-324) – Please see ACD feature codes at the top of list.

  Night Ring On/Off (9860) – Can only be done from an Administrator’s phone.  The display will say Night Mode On if night mode is on.  One common symptom is that when people are calling in they are getting the night greeting.  Obvious I know but you wouldn’t believe the calls one gets.

  Outgoing Call (8) - A lot of people get confused here because most of the time it is 9.

  Page On/Off (325) - All of a sudden you are not receiving pages through your phone.  You have removed yourself from the page zone.

  Queue Request (6) – If you call another phone internally (intercom) and they are busy you can use this and their phone will automatically call you back when they are done with their call.

  Review Keys (396) - Do you want to know what is programmed on a certain key?  Use this feature code and then hit the desired key.  It will let you know.

  System Forward On/Off (352-354) - Equally as common as the headset problem mentioned earlier.  You will know you have turned it off because your phone will ring forever and never forward to voicemail.  Very frequently addressing this issue.

These are just a few of the feature codes and scenarios I have encountered that could be easily fixed without making a trip.  There are tons that didn’t make it here. If any techs can think of any (regardless of the system) please feel free to share.

Resolution: The burnflash utility is used to manually update the switch with the current version firmware compatible with the ShoreWare server software.

When you install a new switch, ShoreTel’s TMS service is set up to detect the switch’s firmware version and automatically upgrade your hardware to the latest version.  For instances that require manual firmware upgrade, you could use Burnflash utility to manually upgrade switch firmware.  Please refer to Maintenance Guide for detail instruction.

A reliable, affordable and secure, fax-over-IP? You are probably thinking to yourself, “No way!”. Here at EtherSpeak we say, Yes way! EtherFax is the answer.

With respect to voice over IP, a codec is an algorithm used to encode and decode the voice conversation. Since voice and sound as we hear it is analogue, it needs to be converted (or encoded) to a digital format suitable for transmission over the Internet. Once at the other end, it needs to be decoded again so the other person can hear what you are saying. There are a variety of different ways this encoding and decoding can be done – many of which utilise compression in order to reduce the required bandwidth of the conversation. A key thing to remember with VoIP, is that encoding, particularly when heavy compression is used, takes time, which adds a delay to the conversation. Thus, the holy grail is a codec which not only maintains good quality with compression, but is able to do the encoding and decoding in a minimal amount of time.

These pages attempt to demistify codecs and give a brief overview of the different codecs and when they are used. It is important to keep in mind that different VoIP clients support different codecs, and each VoIP provider will only support a subset of the codecs too. Generally, when a VoIP call is established, you will need to use a codec that both parties and the provider support. No need to worry though, this sort of negotiation is handled automatically, but knowing the details will enable you to force or encourage certain codecs to be used. Understanding codecs will also help you understand why some VoIP clients sound better than others, and why voice quality with some providers, or through certain ISPs, are better than others.

If you would like to read up more about codecs with respect to VoIP, the following links may be of interest:

• Codecs page the VoIP Wiki
• Find out which hardware supports which codecs.

Codec Comparison

The following table lists the various codecs used in voice over IP, and in particular SIP. Many codecs come in a few varieties, and we have attempted to list all such version of each codec. If you would like to voice your opinion about a particular codec, or discuss the merits of one over another, feel free to do so in our voice over IP forums.

Notes

The information provided here is for information purposes only, if you find errors or ommissions, please report them in the relevant discussion forum.

Bandwidth

• Bandwidth values represent the amount of data in the payload of the IP packets.
• Bandwidth values indicate the bandwidth in each direction – not the sum of upstream and downstream bandwidths.
• Bandwidth values assume continuous transmission of voice in both direction with no silence suppression.
• The ‘nominal bandwidth’ column indicates the typical Ethernet bandwidth one can expect the codec to use.

Sampling Rate

The sampling rate is the rate at which the analogue audio signal is sampled. Nyquist’s Theorem states that in order to record a certain frequency, sampling must occur at at least twice that frequency. Thus, the higher the sampling rate, the greater the frequency range in the encoded audio stream. The human ear is capable of hearing from about 20Hz to about 20,000Hz. Typically, speech is around 100-4,000Hz. Thus, a sampling rate of at least 8kHz is required to accurately encode the human voice. Greater sampling rates will capture higher frequencies (this is useful, for example, if you are playing music down the phone), but will also increase bandwidth as there are more samples to encode and transmit.

Payload Size

The size of the payload of each encoded voice packet influences two things: lag and bandwidth. Every encoded packet that is sent incurs fixed bandwidth overheads (due to IP and other headers added to the data in the network). Thus, larger payloads incur a proportionately smaller overhead, thus reducing the nominal bandwidth utilisation. However, by using larger payloads, more audio (ie., a longer period of time) is required to construct a single packet, which in turn increases the amount of time it takes for even the beginning of the packet to reach the other end and be decoded, thus increasing the lag in the conversation. This is a typical trade-off in VoIP. Most codecs use payload sizes of 10-40ms.

When making a call over the Internet, the software (soft-phone) or hardware needs to use a codec to send/receive information in a certain format and convert it to the sounds you hear.

Generally, a codec with a higher bandwidth requirements provides better voice quality (If your Internet connection is fast enough to support the codec). Most VoIP providers/hardware/licensed software will support G.711 and G.729 (However be sure to check this before purchasing hardware, or signing up with a VoIP provider!). The G.711 codec requires a connection almost 3 times faster than that required by the G.729 codec. If you are using a free soft-phone, then G.729 may not be available to you; however, the GSM codec should be, and will give you similar call quality to that of a mobile phone. During a call, the following data would be used with the G.729 codec: 31.2Kbps up and 31.2Kbps down= 62.4Kbps in total 62.4 Kbps= .0076MBps (MickJT would like to see the figures here, it is neither .0076 in IEC nor SI standards, but however is the average between the two) 60seconds x .0076= .456MB per minute. So the G.729 codec uses roughly .5MB/min during a VoIP call. The following table shows bandwidth requirements for many common codecs. Codec.................Bandwidth Usage (Up/Down) G.711 (64 Kbps).......87.2 Kbps G.729 (8 Kbps)........31.2 Kbps G.723.1 (6.3 Kbps)....21.9 Kbps G.723.1 (5.3 Kbps)....20.8 Kbps G.726 (32 Kbps).......55.2 Kbps G.726 (24 Kbps).......47.2 Kbps G.728 (16 Kbps).......31.5 Kbps GSM (7 or kbps).......low ILBC (15 Kbps)........27.7 Kbps