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Grandstream BT100 Series

Manual

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Comments to date: 2. Page 1 of 1. Average Rating:
yassomak 12:59am on Tuesday, September 28th, 2010 
This is just what I needed! I actually use it in 2 different cars and it works great in both. If this item is refurbished, then it should be listed as such on the sales listing. Should not have to find this out once the product is received.
Daniel LEE 5:38pm on Monday, August 30th, 2010 
My husband bought this last year and I decided to buy one for myself and for my father. I think its pretty easy to set up. I just recently purchased the Jabra BT530 Bluetooth Wireless Headsets (Set of 2 in Bulk Packaging) for me and my wife.

Comments posted on www.ps2netdrivers.net are solely the views and opinions of the people posting them and do not necessarily reflect the views or opinions of us.

 

Documents

doc0

Make Calls using IP Address
Direct IP calling allows two parties, that is, a BudgeTone phone and another VoIP Device, to talk to each other in an ad hoc fashion without a SIP proxy. This kind of VoIP calls can be made between two parties if: Both BudgeTone phone and other VoIP Device(i.e., another IP Phone or BudgeTone SIP phone or other VoIP unit) have public IP addresses, or Both BudgeTone phone and other VoIP Device are on the same LAN using private or public IP addresses, or Both BudgeTone phone and other VoIP Device can be connected through a router using public or private IP addresses (with necessary port forwarding or DMZ).
To make a direct IP to IP call, first off hook, then press MENU key, then enter a 12-digit target IP address to make the call. If port is not default 5060, destination ports can be specified by using *4 (encoding for :) followed by the port number.

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BudgeTone-100 User Manual Examples:
If the target IP address is 192.168.0.10, the dialing convention is MENU_key followed by pressing the SEND key or wait for seconds in the No Key Entry Timeout. If the target IP address/port is 192.168.1.20:5062, then the dialing convention would be: MENU_key 192168001020*45062 followed by pressing the SEND key wait for seconds in the No Key Entry Timeout.

Answer an Incoming Call

There are two ways to answer an incoming call: 1. Pick up the handset to answer the call normally using handset, or 2. Press the SPEAKERPHONE button to answer in speakerphone or headset mode
Handset Mode, Speakerphone/Headset Mode
Handset mode and Speakerphone/Headset mode cannot be enabled at the same time. Pressing the hook-switch or Speakerphone button would toggle the phone between these two modes.

Call Hold

While in conversation, pressing the Hold button will put the remote end on hold. Pressing the Hold button again will release the previously Hold state and resume the bi-directional media.
Call Waiting and Call Flashing
If call waiting feature is enabled, while the user is in a conversation, he will hear a special stutter tone if there is another incoming call. User then can press FLASH button to put the current call party on hold automatically and switch to the other call. Pressing flash button toggles between two active calls.

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Call Transfer
Two transfer operations are supported. 5.3.7.1 Blind Transfer User can transfer an active call to a third party without announcement. User presses the TRANSFER button and if the other voice channel is available (i.e., there is no other active conversation besides the current one), user will hear a dial tone. User can then dial the third partys phone number followed by pressing SEND button. NOTE: Enable Call Feature has to be configured to Yes in web configuration page in order to make the features to work. A can hold on to the phone and wait for one of the three following behaviors: A quick confirmation tone (temporarily using the call waiting indication tone) follows by a dial tone. This indicates the transfer has been successful. At this point, the user can either hang up or make another call. A quick busy tone followed by a restored call (On supported platforms only). This means the transfer has failed due to the failed response sent from server and the phone will try to recover the call. The busy tone is just to indicate to the transferor that the transfer has failed. Busy tone keeps playing. This means the phone has failed to receive the final response and decide to time out. Be advised that this does not indicate the transfer has been successful, nor does it indicate the transfer has failed.
5.3.7.2 Attended Transfer User can transfer an active call to a third party with announcement. User presses the FLASH button and hears a dial tone, then dial the third partys phone number followed by pressing SEND button. If the call is answered, press TRANSFER to complete the transfer operation and hand up, if the call is not answered, pressing FLASH button to resume the original call. NOTE: When Attended Transfer failed, if A hangs up, the BudgeTone phone will ring user A back again to remind A that B is still on the call. A can pick up the phone to restore conversation with B.

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Conference Call
BudgeTone 100 phone supports 3-way conference. Assuming that call party A and B are in conversation. A wants to bring C in a conference: 1. 2. 3. 4. A presses the CONFERENCE button to get a dial tone and put B on hold A dials Cs number then SEND key to make the call If C answers the call, then A presses CONFERENCE button to bring B, C in the conference. If C does not answer the call, A can press FLASH back to talk to B.
NOTE: During the conference, if B or C drops the call, the remaining two parties can still talk. However, if A the conference initiator hangs up, all calls will be terminated.

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Call Features
Following table shows the available call features of BudgeTone 100 by using keypad star(*) code, if the VoIP service provider supports these call features in the server side:

*30 *31 *67 *82 *50 *51 *70 *71 *72
Block CallerID (for all-config change) Send CallerID (for all-config change) Block CallerID (per call base) Send CallerID (per call base) Disable Call Waiting (for all-config change) Enable Call Waiting (for all-config change) Disable Call Waiting. (Per Call) Enable Call Waiting (Per Call) Unconditional Call Forward. To use this feature, dial *72 and get the dial tone. Then dial the forward number followed by pressing SEND button and hear dial tone again to confirm the forward, then hang up. Cancel Unconditional Call Forward To cancel Unconditional Call Forward, dial *73 and get the dial tone to confirm the cancel, then hang up. Busy Call Forward To use this feature, dial *90 and get the dial tone. Then dial the forward number followed by pressing SEND button and hear dial tone again to confirm the forward, then hang up. Cancel Busy Call Forward To cancel Unconditional Call Forward, dial *91 and get the dial tone to confirm the cancel, then hang up Delayed Call Forward To use this feature, dial *92 and get the dial tone. Then dial the forward number followed by pressing SEND button and hear dial tone again to confirm the forward, then hang up. Cancel Delayed Call Forward To cancel this Forward, dial *93 and get the dial tone to confirm the cancel, then hang up When in conversation, this action will switch to the new incoming call if user hears the call waiting sound. When in conversation and no other incoming call, this action will switch to a new channel for a new call.

*73 *90

*91 *92

*93 Flash or Hook Flash

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Configuration Guide

Configuration with Keypad
When the phone is IDLE or On Hook, press the MENU button to enter key pad menu state. When the phone goes off-hook or a call comes in, the phone automatically exits the key pad menu state and prepare for the call. It also exits the key pad menu state if left idle for 20 seconds. Here are the key pad menu options supported:

Menu Item

Menu Functions
Display [1] dhcP On or [1] dhcP oFF for the current selection Press MENU key to enter edit mode Press or to toggle the selection Press MENU to save and exit Must recycle power to take effective!!! Display [2] IP Addr Press MENU to display the current IP address Enter new IP address if DHCP is OFF Press or to exit Press MENU to (save and) exit Must recycle power to take effective!!! Display [3] SubNet Press MENU to display the Subnet mask Enter new Subnet mask if DHCP is OFF Press or to exit Press MENU to (save and) exit Must recycle power to take effective!!! Display [4] routEr Press MENU to display the Router/Gateway address Enter new Router/Gateway address if DHCP is OFF Press or to exit Press MENU to (save and) exit Must recycle power to take effective!!!

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Display [5] dnS Press MENU to display the DNS address Enter new DNS address if DHCP is OFF Press or to exit Press MENU to (save and) exit Must recycle power to take effective!!! Display [6] tFtP Press MENU to display the TFTP address Enter new TFTP server IP address Press MENU to save Press or to exit Display [7] G-711u 2 Press MENU to select new codec Press or to browse a list of available codecs line 1 - G-711A - G-3 - G-4 - G-5 - G-6 - G-7 - iLBC 1 Press 1 to 9 to indicate number of frames per TX packet Press MENU to save and exit Must recycle power to take effective!!! Display [8] SIP SP-1 Reserve for future products.

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Display [9] codE rEL Press Menu to display the code releases Press or to browse line 1 b 2006-03-06 date: boot code 2 1. 0. 8. 11 version: boot code 3 P 2006-09-14 date: phone code 4 1. 0. 8. 32 version: phone code 5 c 2005-03-05 date: codec 6 1. 0. 1. 0 version: codec 7 h 2006-09-14 date: web server 8 1. 0. 8. 32 version: web server 9 1r 2004-05-12 date: 1st ring tone 10 1. 0. 0. 0 version: ring tone 11 2r 2005-07-21 date: 2nd ring tone 12 1. 0. 0. 0 version: ring tone 13 3r 0000-00-00 date: 3rd ring tone 14 0. 0. 0. 0 version: ring tone (all zeroes means unavailable or unsupported) Press MENU to exit Display [10] Phy Addr Press MENU to display the physical / MAC address Press or to exit Display [11] ring 0 Press MENU to hear the selected ring tone, press or to select the stored ring tones. Now only 3 are available, ring 0 (default), ring 1 and ring 2. ring 3 is unavailable or unsupported. Press MENU to select and exit Display -- rESEt --, please be very CAREFUL here. Only shown when key to pressed: Key in the physical / MAC address on back of the phone, Press MENU, phone will be reset to FACTORY DEFAULT setting, and all your setting will be erased. Press MENU key without key in anything, phone will function the same as power cycle or reboot. This is called soft reboot.

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When phone is powered on and time is displayed Press or , Display ring [4] , press or again to hear and adjust the ring tone volume, from 0 (off) to 7 (maximum), off and on hook to set Press SPEAKERPHONE button, or off hook and pick up handset, press or to adjust the speakerphone/headset or handset volume

Others

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Configuration with Web Browser
BudgeTone-100 series IP phone has an embedded Web server that will respond to HTTP GET/POST requests. It also has embedded HTML pages that allow a user to configure the IP phone through a Web browser such as Microsofts IE or Mozillas Firefox.

Access the Web Configuration Menu
The BudgeTone-100 IP Phone Web Configuration Pages can be accessed by input the phones IP into browsers URL address field like: http://Phone-IP-Address where the Phone-IP-Address is the IP address of the phone. There are two ways to retrieve this IP address from the phone: 1) When the phone is off-hook or in speakerphone mode, simply press MENU button. (This is most common way to get the IP address of the phone) 2) When the phone is on-hook, press MENU button and then the browsing arrow keys to [2] IP Addr, press MENU again. NOTE: To type IP address into browser to bring up the configuration pages, please strip out the leading 0 as the browser will parse in octet. e.g.: if the IP address is: 192.168.001.014, please type in: 192.168.1.14.
Once the correct IP address of the phone is input into browser and Enter key pressed, the web log in page will come up like following:
Grandstream Device Configuration

Password

All Rights Reserved Grandstream Networks, Inc. 2005

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The password is case sensitive with maximum length of 25 characters. The factory default password for End User is 123, for Administrator is admin respectively. Only administrator has the privilege to get access to ADVANCED SETTINGS configuration page. NOTE: If you cannot log into the configuration pages by using default password, please check with the VoIP service provider. Most likely, the service provider has already provisioned and automatically configured the device for you and has changed the default password.

Configuration Menu

After input the correct password into the login screen, the embedded Web server of the IP phone will respond with the Configuration Pages screen, which is explained in details below.

Status Page:

Grandstream Device Configuration STATUS BASIC SETTINGS ADVANCED SETTINGS
MAC Address: 00.0B.82.05.CA.91 IP Address: 192.168.1.101 Product Model: BT100 REV 2.0 Software Version: Program-- 1.0.8.32 Registered: Yes PPPoE Link Up: disabled NAT: detected NAT type is full cone NAT Mapped IP: 70.22.120.137 NAT Mapped Port: 60617 Total Inbound Calls: 7 Total Outbound Calls: 2 Total Missed Calls: 3 Total Call Time (in minutes): 19 Total SIP Message Sent: 428 Total SIP Message Received: 626 Total RTP Packet Sent: 38542 Total RTP Packet Received: 38514 Total RTP Packet Loss: 0 All Rights Reserved Grandstream Networks, Inc. 2005 Bootloader-- 1.0.8.11 HTML-- 1.0.8.32 VOC-- 1.0.1.0 System Up Time: 0 day(s) 18 hour(s) 1 minute(s)

Layer 2 QoS: 802.1Q/VLAN Tag Allow incoming SIP messages from SIP proxy only: No Yes

802.1p priority value

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Use DNS SRV: User ID is phone number: SIP Registration: Unregister On Reboot: Register Expiration: Early Dial: Allow outgoing call without Registration: Dial Plan Prefix: No Key Entry Timeout: Use # as Dial Key: local SIP port: local RTP port: Use random port: NAT Traversal:

No No Yes Yes

Yes Yes No No (in seconds. default 1 hour, max 45 days) Yes (use "Yes" only if proxy supports 484 response) Yes (this prefix string is added to each dialed number) (in seconds, default is 4 seconds)

5060 5004

Yes (if set to Yes, "#" will function as the Dial key) (default 5060) (1024-65535, default 5004) Yes
Yes, STUN server is: keep-alive interval: Use NAT IP

stun.fw dnet.net:3478

(URI or IP:port)
(in seconds, default 20 seconds) (if specified, this IP address is used in SIP/SDP
message) (if specified, the content will appear in ProxyRequire header)
Proxy-Require: Voice Mail UserID: SUBSCRIBE for MWI:
(User ID/extension for 3rd party voice mail system)
No, do not send SUBSCRIBE for Message Waiting Indication Yes, send periodical SUBSCRIBE for Message Waiting Indication
Auto Answer: Offhook Auto-Dial: Enable Call Features:

No offhook)

Yes (User ID/extension to dial automatically when
No Yes (if Yes, Call Forwarding & Call-Waiting-Disable are supported locally) No Yes

Disable Call-Waiting:

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Send DTMF: DTMF Payload Type: Send Flash Event: Onhook Threshold: NTP Server:

in-audio

via RTP (RFC2833)

via SIP INFO

800 ms time.nist.gov
Yes (Flash will be sent as a DTMF event if set to Yes)

(URI or IP address)

system ring tone custom ring tone 1, used if incoming caller ID is Default Ring Tone:
custom ring tone 2, used if incoming caller ID is
custom ring tone 3, used if incoming caller ID is
Send Anonymous: Anonymous Method: Time to ring: Special Feature: Syslog Server: Syslog Level:
Yes (caller ID will be blocked if set to Yes) Use Privacy Header

Use From Header

60 seconds Standard
Firmware Upgrade and Provisioning: Upgrade Via

fm.grandstream.com/gs

Firmware Server Path: Config Server Path: Firmware File Prefix: Config File Prefix: Automatic Upgrade: No
Firmware File Postfix: Config File Postfix:
Yes, check for upgrade every

minutes (default 7 days)

Always Check for New Firmware Check New Firmware only when F/W pre/suffix changes Always Skip the Firmware Check

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Firmware Key: Authenticate Conf File: Lock keypad update: Allow conf SIP Account in Basic Settings: Override MTU Size:

0 Update Cancel Reboot

(in Hexadecimal Representation) No Yes) No No Yes (configuration update via keypad is disabled if set to Yes) Yes Yes (cfg file would be authenticated before acceptance if set to

Admin Password

Administrator password. Only administrator can configure the Advanced Settings page. Password field is purposely blanked for security reason after clicking update and saved. The maximum password length is 25 characters. IP address or FQDN domain name provided by VoIP service provider. e.g., the following are some valid examples: sip.my-voip-provider.com or sip:my-company-sip-server.com or 192.168.1.200:5066 (where 5066 is the port number different to default 5060) IP address or FQDN domain name of Outbound Proxy (or called Media Gateway or Session Border Controller). Used by IP phone for firewall or NAT penetration in different network environment. If symmetric NAT is detected, STUN will not work and ONLY outbound proxy will provide solution for it. User account information, provided by VoIP service provider (ITSP), usually has the form of digits similar to phone number or actually a phone number ID used for authentication, usually same as SIP user ID, but could be different and decided by the ITSP Account information, password for IP Phone to register to (SIP) servers of ITSP. Maximum length is 25 characters. The BudgeTone IP phone supports up to 8 different codec types including G711-ulaw (PCMU), G711-alaw (PCMA), G723, G729A, G726-32 (ADPCM), G722, G728 and iLBC. A user can configure codecs in a preference list that will be included with the same preference order in SDP message. The first codec in this list can be entered by choosing the appropriate option in Choice 1. Similarly, the last codec in this list can be entered by choosing the appropriate option in Choice 8. Encoding rate for G723 codec. By default, 6.3kbps rate is set.

SIP Server

Outbound Proxy
SIP User ID Authentication ID Authenticate Password Preferred Vocoder

G723 Rate:

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BudgeTone-100 User Manual iLBC frame size iLBC payload type Silence Suppression
iLBC packet frame size. Default is 20ms. For Asterisk IP-PBX, 30ms might need to be configured for compatibility Payload type for iLBC. Default value is 97. The valid range is between 96 and 127. This controls the silence suppression/VAD feature of G723 and G729. If set to Yes, when a silence is detected, small quantity of VAD packets (instead of audio packets) will be sent during the period of no talking. If set to No, this feature is disabled.

Unregister On Reboot Register Expiration Early Dial
Allow outgoing call without Registration Dial Plan Prefix No key Entry Timeout Use # as Dial Key

Local SIP port

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BudgeTone-100 User Manual Local RTP port
This parameter defines the local RTP-RTCP port pair the IP phone will listen and transmit. It is the base RTP port for channel 0. When configured, channel 0 will use this port_value for RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP and port_value+3 for its RTCP. The default value is 5004.
Use Random port Default No. If set to Yes, the device will pick randomly generated SIP and RTP ports. This is usually necessary and useful when multiple IP Phones are behind the same full cone NAT router. NAT Traversal Defines whether the NAT traversal mechanism is activated. It should be set to YES if the device is behind NAT router. If Outbound Proxy is NOT configured, STUN server needs to be set to activate STUN detection mechanism. Usually ITSP will provide these settings for device to work properly behind NAT/Firewall If this field is set to Yes without STUN server, then the device will periodically (every Keep-alive interval) send a dummy UDP packet to the SIP server to pinhole the NAT in the router side. Default is 20 seconds. The interval of sending dummy UDP packet to keep NAT pin hole open in the router side. Min. value is 10 seconds. NAT IP address (WAN side) used in SIP/SDP message. Default is blank. SIP Extension to notify SIP server that the unit is behind the NAT/Firewall. Required by some soft switch vendor like Nortel MCS. User ID (extension or access number) of a 3rd party VoiceMail system where the user may have an account. By defining it, user presses the MESSAGE button on the phone, an INVITE message will send to that ID/number to allow the user to retrieve VM. Default is No. When set to Yes, a SUBSCRIBE for Message Waiting Indication will be sent periodically to server. BT-100 support both synchronize and nonsynchronized SUBSCRIBE SIP message. Default is No. When set to Yes, the phone will automatically pick up the call after a short beep and turn on the speaker. This parameter allows the user to configure a User ID or extension number to be automatically dialed upon off hook (like hot line). Please note that only the user part of a SIP address needs to be entered here. The phone will automatically append the @ and the host portion of the corresponding SIP address. Default is Yes. Advance call features or feature codes functions (Star code, see Section 5.4 of this manual) are supported locally Default is No. User can use * code to use this feature per call basis.

Keep alive interval Use NAT IP Proxy-Require Voice Mail User ID
Subscribe for MWI Auto Answer Offhook Auto-Dial
Enable call features Disable Call Waiting

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BudgeTone-100 User Manual Send DTMF
This parameter specifies the mechanism to transmit DTMF digits. There are 3 modes supported: in audio, which means DTMF is combined in inband audio signal (not very reliable with low-bit-rate codec); via RTP (RFC2833); or via SIP INFO. Make sure this setting is synchronized with SIP server otherwise will not work. This parameter sets the payload type for DTMF using RFC2833 Default is 101

DTMF Payload Type

Send Flash Event Default is NO. If set to Yes, flash will be sent as DTMF event therefore the FLASH key to switch calls will NOT work. Onhook Threshold NTP server Default Ring Tone Time period when cradle pressed (Hook Flash) to simulate FLASH. Adjust this time value will prevent user on hook carelessly to activate Flash/Hold and cause the phone ring back. Default is 800ms. URI or IP address of the NTP (Network Time Protocol) server, which will be used by the phone to synchronize the date and time. By default System Ring Tone is selected. All calls received that do not match any incoming caller ID number will generate the selected ring tone - system, 1, 2 or 3. The user can setup up one number on each of the ring tones 1, 2, and 3. When a call is received from one of these numbers, the respective ring tone will be generated. If the ring tone does not exist, the system default ring tone will be played. For BT-100, ring 3 is not supported.
Send Anonymous If this parameter is set to Yes, user ID will be sent as anonymous, essentially block the Caller ID from displaying. Anonymous Method Time to ring Special Feature This is decided by supported SIP server/proxy. Either Use From Header or Use Privacy Header depends on how SIP sever/proxy process the SIP header. Allow user to adjust the ring time of the phone. Default is 60 seconds Default is Standard. Choose the selection to meet some special requirements from Soft Switch vendors like Nortel, Sonus, Broadsoft, etc. End user please do not touch this parameter. The IP address or URL of System log server, especially useful for ITSP (Internet Telephone Service Provider). End user dont encourage to touch this part

Syslog Server

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BudgeTone-100 User Manual Syslog level
Select the level for the phone to report the log messages. Default is NONE. Useful for ITSP. The level is one of DEBUG, INFO, WARNING or ERROR. Syslog messages are sent based on the following events:

For example, if the MAC address is 000b8200e395, it should be encoded as 0002228200333395. Step 3: To perform factory reset: a. b. c. d. e. Press the MENU button for Key Pad Menu options. Press the Up or Down button to see reset. Enter the encoded MAC address. Press the MENU button again Wait for phone reboot and the LCD backlight finish flashing.

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Headset Connection
The BT-100 phone has a headset socket allowing user to plug the headset into the phone. The picture below shows the handset and headset connectors wiring schema.
As show in the schema, the left side is pin assignment for a RJ11 interface headset; while the right side is showing a normal 3.5mm headset plug. A 3.5mm to 2.5mm plug converter is required if user want to user normal 2.5mm cell phone headset. The plug converter can be purchased from any electronics component store.

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Cross Over Cable For Plantronic Headset
Some users want to use headset products with RJ22 plug, like the Plantronic headset. In this case, a special cable is required. The handset twisted cable is a roll-over cable standard for ALL handset, using RJ22 plug. Since the default handset plug lay out is Asia standard which is just reversed to North American or Europe standard, therefore US and Europe customer can not use popular Plantronic Handset without tweaking the connection wiring. The quick and easy solution will be a special cable: a cross-over cable. Please ask for help from electrician if user can not understand this part. Here is the example and instruction to hand made such an adapter or cable and confirmed to work with Plantronic M12 headset with amplifier, which is most popular headset used in call centers. Here are the schemas of the two cables; the plug is viewed with Pin facing user, with PINs as specified above: d: SP + a: SP c: Mic + b: Mic -
Roll Over Cable: (already provided, to connect the handset to phone base)

a b c d

d c b a

One End

Another End

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Cross Over Cable: (required to allow Plantronic Headset to work with the phone)

a b c d b a d c

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10 Glossary of Terms

ADSL Asymmetric Digital Subscriber Line: Modems attached to twisted pair copper wiring that transmit from 1.5 Mbps to 9 Mbps downstream (to the subscriber) and from 16 kbps to 800 kbps upstream, depending on line distance. AGC Automatic Gain Control, is an electronic system found in many types of devices. Its purpose is to control the gain of a system in order to maintain some measure of performance over a changing range of real world conditions. ARP Address Resolution Protocol is a protocol used by the Internet Protocol (IP) [RFC826], specifically IPv4, to map IP network addresses to the hardware addresses used by a data link protocol. The protocol operates below the network layer as a part of the interface between the OSI network and OSI link layer. It is used when IPv4 is used over Ethernet ATA Analogue Telephone Adapter. Covert analogue telephone to be used in data network for VoIP, like Grandstream HT series products. CODEC Abbreviation for Coder-Decoder. It is an analog-to-digital (A/D) and digital-to-analog (D/A) converter for translating the signals from the outside world to digital, and back again. CNG Comfort Noise Generator, generate artificial background noise used in radio and wireless communications to fill the silent time in a transmission resulting from voice activity detection. DATAGRAM A data packet carrying its own address information so it can be independently routed from its source to the destination computer DECIMATE To discard portions of a signal in order to reduce the amount of information to be encoded or compressed. Lossy compression algorithms ordinarily decimate while subsampling. DECT Digital Enhanced Cordless Telecommunications: A standard developed by the European Telecommunication Standard Institute from 1988, governing pan-European digital mobile telephony. DECT covers wireless PBXs, telepoint, residential cordless telephones, wireless access to the public switched telephone network, Closed User Groups (CUGs), Local Area

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Networks, and wireless local loop. The DECT Common Interface radio standard is a multicarrier time division multiple access, time division duplex (MC-TDMA-TDD) radio transmission technique using ten radio frequency channels from 1880 to 1930 MHz, each divided into 24 time slots of 10ms, and twelve full-duplex accesses per carrier, for a total of 120 possible combinations. A DECT base station (an RFP, Radio Fixed Part) can transmit all 12 possible accesses (time slots) simultaneously by using different frequencies or using only one frequency. All signaling information is transmitted from the RFP within a multi-frame (16 frames). Voice signals are digitally encoded into a 32 kbit/s signal using Adaptive Differential Pulse Code Modulation. DNS Short for Domain Name System (or Service or Server), an Internet service that translates domain names into IP addresses DID Direct Inward Dialing Direct Inward Dialing. The ability for an outside caller to dial to a PBX extension without going through an attendant or auto-attendant. DSP Digital Signal Processing. Using computers to process signals such as sound, video, and other analog signals which have been converted to digital form. Digital Signal Processor. A specialized CPU used for digital signal processing. Grandstream products all have DSP chips built inside. DTMF Dual Tone Multi Frequency The standard tone-pairs used on telephone terminals for dialing using in-band signaling. The standards define 16 tone-pairs (0-9, #, * and A-F) although most terminals support only 12 of them (0-9, * and #). FQDN Fully Qualified Domain Name A FQDN consists of a host and domain name, including top-level domain. For example, www.grandstream.com is a fully qualified domain name. www is the host, grandstream is the second-level domain, and.com is the top level domain. FXO Foreign eXchange Office An FXO device can be an analog phone, answering machine, fax, or anything that handles a call from the telephone company like AT&T. They should also operate the same way when connected to an FXS interface.

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An FXO interface will accept calls from FXS or PSTN interfaces. All countries and regions have their own standards. FXO is complimentary to FXS (and the PSTN). FXS Foreign eXchange Station An FXS device has hardware to generate the ring signal to the FXO extension (usually an analog phone). An FXS device will allow any FXO device to operate as if it were connected to the phone company. This makes your PBX the POTS+PSTN for the phone. The FXS Interface connects to FXO devices (by an FXO interface, of course). DHCP The Dynamic Host Configuration Protocol (DHCP) is an Internet protocol for automating the configuration of computers that use TCP/IP. DHCP can be used to automatically assign IP addresses, to deliver TCP/IP stack configuration parameters such as the subnet mask and default router, and to provide other configuration information such as the addresses for printer, time and news servers. ECHO CANCELLATION Echo Cancellation is used in telephony to describe the process of removing echo from a voice communication in order to improve voice quality on a telephone call. In addition to improving quality, this process improves bandwidth savings achieved through silence suppression by preventing echo from traveling across a network. There are two types of echo of relevance in telephony: acoustic echo and hybrid echo. Speech compression techniques and digital processing delay often contribute to echo generation in telephone networks. H.323 A suite of standards for multimedia conferences on traditional packet-switched networks. HTTP Hyper Text Transfer Protocol; the World Wide Web protocol that performs the request and retrieve functions of a server IP Internet Protocol. A packet-based protocol for delivering data across networks. IP-PBX IP-based Private Branch Exchange

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IP Telephony (Internet Protocol telephony, also known as Voice over IP Telephony) A general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). The basic steps involved in originating an IP Telephony call are conversion of the analog voice signal to digital format and compression/translation of the signal into Internet protocol (IP) packets for transmission over the Internet or other packet-switched networks; the process is reversed at the receiving end. The terms IP Telephony and Internet Telephony are often used to mean the same; however, they are not 100 per cent interchangeable, since Internet is only a subcase of packet-switched networks. For users who have free or fixed-price Internet access, IP Telephony software essentially provides free telephone calls anywhere in the world. However, the challenge of IP Telephony is maintaining the quality of service expected by subscribers. Session border controllers resolve this issue by providing quality assurance comparable to legacy telephone systems. IVR IVR is a software application that accepts a combination of voice telephone input and touchtone keypad selection and provides appropriate responses in the form of voice, fax, callback, email and perhaps other media. MTU A Maximum Transmission Unit (MTU) is the largest size packet or frame, specified in octets (eight-bit bytes), that can be sent in a packet- or frame-based network such as the Internet. The maximum for Ethernet is 1500 byte. NAT
Network Address Translation
NTP Network Time Protocol, a protocol to exchange and synchronize time over networks The port used is UDP 123 Grandstream products using NTP to get time from Internet OBP/SBC Outbound Proxy or another name Session Border Controller. A device used in VoIP networks. OBP/SBCs are put into the signaling and media path between calling and called party. The OBP/SBC acts as if it was the called VoIP phone and places a second call to the called party. The effect of this behavior is that not only the signaling traffic, but also the media traffic (voice, video etc) crosses the OBP/SBC. Without an OBP/SBC, the media traffic travels directly between the VoIP phones. Private OBP/SBCs are used along with firewalls to enable VoIP calls to and from a protected enterprise network. Public VoIP service providers use OBP/SBCs to allow the use of VoIP protocols from private networks with internet connections using NAT.

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BudgeTone-100 User Manual PPPoE
Point-to-Point Protocol over Ethernet, is a network protocol for encapsulating PPP frames in Ethernet frames. It is used mainly with cable modem and DSL services. PSTN Public Switched Telephone Network i.e. the phone service we use for every ordinary phone call, or called POT (Plain Old Telephone), or circuit switched network. RTCP Real-time Transport Control Protocol, defined in RFC 3550, a sister protocol of the Real-time Transport Protocol (RTP), It partners RTP in the delivery and packaging of multimedia data, but does not transport any data itself. It is used periodically to transmit control packets to participants in a streaming multimedia session. The primary function of RTCP is to provide feedback on the quality of service being provided by RTP. RTP Real-time Transport Protocol defines a standardized packet format for delivering audio and video over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first published in 1996 as RFC 1889 SDP Session Description Protocol, is a format for describing streaming media initialization parameters. It has been published by the IETF as RFC 2327. SIP Session Initiation Protocol, An IP telephony signaling protocol developed by the IETF (RFC3261). SIP is a text-based protocol suitable for integrated voice-data applications. SIP is designed for voice transmission and uses fewer resources and is considerably less complex than H.323. All Grandstream products are SIP based STUN Simple Traversal of UDP over NATs, is a network protocol allowing clients behind NAT (or multiple NATs) to find out its public address, the type of NAT it is behind and the internet side port associated by the NAT with a particular local port. This information is used to set up UDP communication between two hosts that are both behind NAT routers. The protocol is defined in RFC 3489. STUN will usually work well with non-symmetric NAT routers. TCP Transmission Control Protocol, is one of the core protocols of the Internet protocol suite. Using TCP, applications on networked hosts can create connections to one another, over which they can exchange data or packets. The protocol guarantees reliable and in-order delivery of sender to receiver data.

 

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