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1. Purpose EE Key 7 Name UserID Time Zone GMT 8 00 US Pacific Time Los Angeles hi Daylight 2o NO L Yes if set to Yes display time will be 1 hour ahead of normal savings 4 1 7 2 0 10 1 7 2 0 60 Time time Optional Rule 7 7 LCD Backlight amp No E Yes Always On Time Display 12 HOUR 24 HOUR Format Date E Year Month Day Display E Month Day Year BORSE Day Month Y ear Display Clock C Eo instead of No nes Date System Device Mode Device vee Switch default NAT Router NAT Router Configuration 29 WAN side http E No L Yes WAN side access to http server will be rejected if set access to No Reply to ICMP E No E Yes Unit will not respond to PING from WAN side if set to on WAN port No Cloned WAN Zz mM mM mM Jin hex format MAC Addr LAN Subnet M default is 255 255 255 0 Mask LAN DHCP base IP for the LAN port default is Base IP 192 168 2 1 DHCP IP E 120 in units of hours default is 120 hours or 5 days DMZ IP WAN port LAN IP LAN port Protocol T o wm a wm Port o UDPOnly Forwarding o o UDPOnly Po o a v m T o e TTL End User This contains the password to access the Web Configuration Menu Password This field is case sensitive with a maximum length of 25 characters 30 IP Address There are two modes under which the GXP 2000 can operate e If DHCP mode is enabled then all the field values for the Static
2. Press Menu button to display the Instant Messages received Press or T to toggle the selection Press Menu button to choose the menu item Press button to return to the upper menu Note GXP2000 only supports the function of receiving of Instant Messages Display Direct IP Call Press Menu button to display the direct IP call interface Enter 12 digit IP address For example 10 10 1 2 could be entered like 010010001002 Press or gt to move the cursor or toggle the selection Press Menu button to confirm 24 Menu Functions Display Preference Press Menu button to enter this sub menu including Do NOT Disturb or Ring Tone or Ring Volume or Download SCR XML or Erase Custom SCR or Back DND Do NOT Disturb function could be turned on or off in the DO NOT Disturb menu Choose different ring tones you prefer in the Ring Tone menu Adjust ring volume in the Ring Volume menu by using and gt button Press or T to toggle the selection Press Menu button to choose the menu item you want Press to return to the upper menu Display Configure Press Menu button to display the configuration items Network or SIP or Audio or Upgrade or Factory Reset Please check the web configuration page for more detail information about these items Press or T to toggle the selection Press to return the u
3. 1 Make Handset SPEAKER Headset off hook or press the available LINE key to select a SIP account the corresponding LINE LED will light up in solid red Enter the phone numbers and press the SEND key 2 Make Handset SPEAKER Headset off hook or press the available LINE key the corresponding LINE LED will light up in solid red Press the SEND button to redial the last number called 3 Make Handset SPEAKER Headset off hook or press the available LINE key the corresponding LINE LED will light up in solid red Press the Speed Dial key to call the preset calling party number 4 Press the DOWN button then select the number in the Phone Book menu you want to call by pressing the Menu button and then press the Menu button again to call this number 5 Press the UP button then select the number in the Missed Calls you want to call by pressing the Menu button and then press the Menu button again to call this number Note e Once pressed the dialed number is displayed on the LCD as the corresponding DTMF tone is played out e Ifthe SEND button is not pressed after the phone number the phone will wait for 4 seconds before initiating the call 4 4 4 Making Calls using IP Address Direct IP calling allows two phones to talk to each other in an ad hoc fashion without a SIP proxy VoIP calls can be made between two phones if e Both phones have public IP addresses or e Both phones are on a same LAN using private or public IP addresses o
4. At any time you can trigger an immediate download from by choosing the Download Phonebook in the GUI Phone Book Menu you can use the down arrow key when the phone is on hook Example XML file of gs phonebook xml lt ym versions 07 gt lt AddressBook gt lt Contact gt Last Name Crai g Last Name FirstName Ri chard Fi rst Name Phone lt phonenumber gt 6910 lt phonenumber gt lt accountindex gt 0 lt accountindex gt lt Phone gt Contact lt COML act Last Name Peterson Last Name lt FirstName gt Susan lt FirstName gt lt Phone gt lt phonenumber gt 6911 lt phonenumber gt lt accountindex gt 0 lt accountindex gt lt Phone gt Contact gt lt Contact gt Last Name Mayer Last Name FirstName Ci ndy First Name gt Phone lt phonenumber gt 6915 lt phonenumber gt lt accountindex gt 0 lt accountindex gt lt Phone gt Contact gt AddressBook Note This feature is designed for enterprise user only Please contact Grandstream for more details 40 Idle Screen XML Download DTMF Payload Type NTP server Distinctive Ring Tone The feature will be activated when Enable Idle Screen XML Download is set to YES HTTP or TFTP AND a valid Idle Screen XML Path 1s set This feature does not automatically download the gs screen xml file in the path even when activated Because the LCD is composed of 130 64 mono pixels
5. ThdisercConlsurdadollussbesodeie uade e ui uot aas aaa M nu Lied SU 2 25 2 9 Advanced User Cono UutatlOfls cs eec E Een nodos eoe R ER PNEU NE IDtET eee 35 5 2 4 Saving the Configuration Changes eeeeeeeeeeeeeeeeeeeeee nnne 49 5 2 5 Rebooting the Phone from Remote eeeeeessssseeeeeeeneerrne 50 5 3 Configuration through Central Provisioning Server eeeeeeeeeee 50 G Pinnyare Upo 10 cae nen enn ne Ree RUD Sore E eee ee 52 6 1 Firmware Upgrade through TFTP HTTP 20 0 cc ccccccssssseeeeeceeeeeeeaeneeeees 52 0 2 Conteurauon File Download usce eee tbe SEE ruit eoe EE 53 6 3 Firmware and Configuration File Prefix and Postfix essssss 53 6 4 Managing Firmware and Configuration File Download 09 f Restore Factory Default SEINE oats sien a ts orto aito Sepe cnm E gie hue d e Deas ees iUe dits EE D ERU 54 Appendixcl Glossary REESE OE mE 55 Appendix II AUT Wem C Batts s et etia ands hee eO Cte vote vp ae to meinte Va oup rue 62 1 Welcome Thank you for purchasing Grandstream award winning GXP 2000 Enterprise IP Phone You made an excellent choice and we hope you will enjoy all its capabilities Grandstream s award wining GXP 2000 SIP IP phone is the innovative enterprise IP telephone that offers a rich set of functionality and superb sound quality They are fully compatible with SIP industry standard and can interoperate with
6. IP mode are not used even though they are still saved in the Flash memory The GXP 2000 will acquire its IP address from the first DHCP server it discovers from the LAN it is connected e To use the PPPoE feature the PPPoE account settings need to be set The GXP 2000 will attempt to establish a PPPoE session if any of the PPPoE fields is set e If Static IP mode is enabled then the IP address Subnet Mask Default Router IP address DNS Server 1 primary DNS Server 2 secondary fields will need to be configured These fields are set to zero by default Speed Dial There are 7 speed dial fields that can be configured e Name field is used to identify the person It will be displayed on LCD when pressing the corresponding Key e UserID field is the number configured e Account field is the SIP account associated with the number Asterisk BLF Asterisk Busy Line Field feature needs the support of Asterisk PBX Please check Asterisk for more details Presence Watcher This feature is used to monitor the status of other SIP devices which supports SIP PUBLISH for Presence RFC 3903 If SIP PUBLISH for Presence on GXP2000 is enabled Please refer to PUBLISH for Presence option at account configuration pages the status of GXP2000 can be monitored by other SIP devices The status of GX P2000 can be changed by enable disable Do Not Disturb via the GUI Menu The default status of GXP2000 is Do Not Disturb disabled The LED of t
7. Its power adaptor 1s compliant with UL standard The phone should only be operated with the universal power adaptor provided with the package Damages to the phone caused by using other unsupported power adaptors are not covered by the manufacturer s warranty 2 6 Warranty Grandstream has a reseller agreement with our reseller customer End user should contact the company from whom you purchased the product for replacement repair or refund If you purchased the product directly from Grandstream contact your Grandstream sales and Service Representative for a RMA Return Materials Authorization number Grandstream reserves the right to remedy warranty policy without prior notification Warning Please do not attempt to use a different power adaptor Using other power adaptor may damage the GXP 2000 and will void the manufacturer warranty Caution Changes or modifications to this product not expressly approved by Grandstream or operation of this product in any way other than as detailed by this User Manual could void your manufacturer warranty Information in this document is subject to change without notice No part of this document may be reproduced or transmitted in any form or by any means electronic or mechanical for any purpose without the express written permission of Grandstream Networks Inc 10 3 Product Overview GXP 2000 series IP phone is designed to be an enterprise telephone which could also be used in r
8. a very simple file transfer protocol with the functionality of a very basic form of FTP It uses UDP port 69 as its transport protocol UDP User Datagram Protocol UDP is one of the core protocols of the Internet protocol suite Using UDP programs on networked computers can send short messages known as datagrams to one another UDP does not provide the reliability and ordering guarantees that TCP does datagrams may arrive out of order or go missing without notice However as a result UDP is faster and more efficient for many lightweight or time sensitive purposes VAD Voice Activity Detection or Voice Activity Detector is an algorithm used in speech processing wherein the presence or absence of human speech is detected from the audio samples VLAN A virtual LAN known as a VLAN is a logically independent network Several VLANs can co exist on a single physical switch It 1s usually refer to the IEEE 802 1Q tagging protocol VoIP Voice over IP VoIP encompasses many protocols All the protocols do some form of signalling of call capabilities and transport of voice data from one point to another e g SIP H 323 etc 60 61 Config Phone Book Call History Direct IP Call Preference Status Factory Functions Reboot Exit Appendix If GUI Menu Chart Call History 62 Back Delete All NOTE Tao ths sage need press ROUND button VERY QUICKLY TWICE to bring up the input field page 63
9. be stored in one directory In addition when the field Check New Firmware only when F W pre suffix changes is set to Yes the device will only issue firmware upgrade request if there are changes in the firmware Prefix or Postfix 6 4 Managing Firmware and Configuration File Download When Automatic Upgrade is set to Yes Service Provider can use P193 Auto Check Interval in minutes default and minimum is 60 minutes to have the devices periodically check with either Firmware Server or Config Server whenever they are defined This allows the device periodically check if there are any new changes need to be taken on a scheduled time By defining different intervals 1n P193 for different devices Server Provider can spread the Firmware or Configuration File download in minutes to reduce the Firmware or Provisioning Server load at any given time 53 7 Restore Factory Default Setting Warning Restore the Factory Default Setting will DELETE all configuration information of the device Please backup or print out all the settings before you approach to following steps Grandstream will not take any responsibility if you lose all the parameters of setting and cannot connect to your service provider Disconnect network cable and power cycle the unit before resetting factory default settings 1 Step 1 Press round menu key to bring up the key pad configuration Ul menu select Config press OK to enter submenu select
10. between Calls esses 20 4A MERO IM CIT TIT A E AE 20 MEE EE VICO EE 2 4 4 10 Checking Message and Message Waiting Indication 2 ZI Mae dicPIelele unica E Cero te een term be 21 4 4 12 Multi functional Button Extensional board seeesssss 22 24 12 Me Speed Did setis dotem nats muet ue aa reditu mutet ON qeu dd 22 4 4 12 2 Asterisk Biy Line Field erneer eerte P S oo ERE DEREN t2 CHI o ors 22 22 12 35 EXEASION BOITO nssocccotseulociyeauna ect dotati bet baa qt addo aec Pvt au San ces odd 22 The fifty six Multi functional buttons on Extension board function the same as Multi functional buttons on the phone base station except that they can not be IE dq m 22 HA LC ED cou m m 22 d 5 Call Beadbir65 ecrit Seed aad m Ea ML E DE AM UE E EE and LR MAE 22 46 Customizeddqdie SCIeefilssoaeetii uei pU une iei antcm tateeuti itu oss tor ite ui DE E uq 23 4 7 XML phone book downloading ssseeeeeenneeeeneee 23 SX Uu HON SUMMER M 24 od Conbieuraduonm Will S Eya deene tori eve erant Pio red ERN VEDI Ute OD aseo Du UE D 24 5 2 Configuration with Web Browser cccccccccccccsessseeeeccccceeeeasaaeseseeeeceeeeeeeaaaes 27 5 2 1 Access the Web Configuration Menu ccccccccccccceceeeeeeeesssesseeeeseeeeeees 27 942 2
11. button will automatically initiate a call to the destination extension Note e When an incoming call arrives while all of the 4 LINE 1 4 channels are in use the Multi functional buttons will function as LINE keys and flash the light for the next incoming call User can press the button to pick up the call e When any one of the 7 functions keys is associated with a call they function as LINE keys In this case speed dial BLF function will not work For ex when first Multi functional button 1s in use you cannot use it for speed dial BLF 4 4 12 2 Asterisk Busy Line Field The seven Multi functional buttons can be configured for Asterisk Busy Line Field function with a specified account When Asterisk BLF is configured on one of the Multi functional buttons Speed Dial function on it will still work when it shows idle status 4 4 12 3 Extension Board The fifty six Multi functional buttons on Extension board function the same as Multi functional buttons on the phone base station except that they can not be Line keys 4 4 13 Page Dialing When off hook you see LINEx DIAL USING you can press the ROUND button and you will see LINEx PAGE USING you can toggle between the modes by pressing the button BEFORE any DTMF digits are dialed On the called party Allow Auto Answer by Call Info on advance page needs to be set to Yes for Paging function The called party will receive an INVITE with call info header that has answer after
12. enable automatic upgrade and provisioning In Check for new firmware every field enter the number of days to enable GXP 2000 to check the server for firmware upgrade or configuration in the defined period of days When set to No GXP 2000 will only do upgrade once at boot up Always check for New Firmware Check New Firmware only when F W pre suffix changes Always Skip the Firmware Check if set to Yes cfg file would be authenticated before acceptance This mechanism 1s useful for the protection of configuration on the device from unauthorized change 39 Phonebook The feature will be activated when Enable Downloadable Phonebook is set to XML Download YES HTTP or TFTP AND a valid Phonebook XML Path is set When the device boot up and completed the provisioning routine it will attempt to download the gs phonebook xml file specified in Phonebook XML Path During this process the LCD should display some messages to indicate that XML Phonebook download is in progress If the Phonebook Download Interval is set to a non zero value GXP2000 will periodically check and download the updated phonebook available If the Remove manually edited entries on download option is set to NO by default GXP2000 will keep ALL entries edited manually and insert the downloaded entries and then save the phonebook Otherwise ALL the existing phonebook entries on the phone will be erased and filled with the downloaded entries
13. on automatically and brings the user s attention Icon LCD Icon Definitions Network Status Icon FLASH in the case of Ethernet link failure OFF if IP address or SIP server is not found ON if IP address and SIP server are located Phone Status Icon OFF when the handset 1s on hook ON when the handset 1s off hook Speaker Phone Status Icon FLASH when phone rings or a call is pending OFF when the speakerphone is off ON when the speakerphone is on Handset Speakerphone and Ring Volume Icon 0 7 scales to adjust handset speakerphone ring volume Real time Clock Synchronized to Internet time server Time zone configurable via web browser Time Icon AM for the morning PM for the afternoon 4 2 Getting Familiar with Keypad E Message Waiting Indicator La gt Ari M ES zs Line 1 4 Keys ELMEEEDUS Ot T c SE E gt Menu Keys Dn QNNM S Speed Dial Hi Configurable 1111 LIIULUALULD Mute Delete lt A o2xu Message m gt Conference EB gt Transfer q lt 4 RJ11 Hold Speaker Send Re Dial Standard Keypad 16 GXP 2000 phone has 35 key buttons Key Button LINE1 LINE4 MULTI FUNCTION UP 1 DOWN LEFT RIGHT gt MENU e IRNF CONF MSG MUTE DEL HOLD SPEAKER SEND 0 9 Key Button Definitions 4 Line keys with LED can be extended to 11 Lines with the use of 7 Speed Dial Key
14. or HTTP The corresponding configuration settings are in the ADVANCED SETTINGS configuration page 6 1 Firmware Upgrade through TFTP HTTP To upgrade via TFTP or HTTP the Firmware Upgrade and Provisioning upgrade via field needs to be set to TFTP or HTTP respectively Firmware Server Path needs to be set to a valid URL of a TFTP or HTTP server server name can be in either FQDN or IP address format Here are examples of some valid URL NOTES e g firmware mycompany com 6688 Grandstream 1 1 1 14 e g 168 75 215 189 TFTP server in IP address format can be configured via GUI menu or web configuration page If TFTP server is in FQDN format it must be set via web configuration interface Once a Firmware Server Path is set user needs to update the settings and reboot the device If the configured firmware server is found and a new code image is available the GXP2000 will attempt to retrieve the new image files by downloading them into the GXP2000 s SRAM Upon verification of checksum the new code image will then be saved into the Flash During the firmware upgrading process the GXP2000 s LCD will display the message about the firmware loading progress If TFTP HTTP fails for any reason e g TFTP HTTP server is not responding there are no code image files available for upgrade or checksum test fails etc the GXP2000 will stop the TFTP HTTP process and simply boot using the existing code image in the flash Firmware u
15. together in a conference If the conference holder wishes to end a conference simply press HOLD which breaks the conference and places both parties on hold User can then talk to each individual party by selecting the corresponding LINE 4 4 10 Checking Message and Message Waiting Indication When GXP 2000 is on hook pressing the MSG button will trigger the phone to call the VM Server VMS configured for the primary account If a line account is selected first it dials the VMS configured for that account The MWI Message Waiting Indicator LED will flash in red color in three quarters of a second when voicemail server sends message waiting information to GXP 2000 4 4 11 Mute and Delete When in conversation with an ACTIVE LINE pressing MUTE DEL will mute the conversation that is you can hear the other party but the other party cannot hear you Pressing the button again will resume the conversation When dialing a number pressing MUTE DEL button will delete the last entered digit When the phone is in idle status pressing MUTE DEL button will activate Do not Disturb function on the phone Pressing MUTE DEL button again will deactivate DND function 2 4 4 12 Multi functional Button Extensional board 4 4 12 1 Speed Dial The seven Multi functional buttons can be configured for speed dial function A vertical rectangle pad on the keypad is provided to label Speed Dial numbers Pressing the speed dial
16. 0 then it will answer the call automatically Note this does not work with Asterisk or other proxies that does not pass along the Call Info header There are workarounds for Asterisk server side setup 4 5 Call Features 22 GXP 2000 series phone supports a list of call features Caller ID Block or Anonymous Call Disable Enable Call Waiting Call Forward on Busy Delay or Unconditional etc Following table shows the call features of GXP 2000 series phone 9 02 Enable Call Waiting Per Call Unconditional Call Forward To use this feature dial 72 and get the dial tone Dial the forward number and for a dial tone and then hang up A call forward icon on status line will be seen when account is set to unconditional call forward Cancel Unconditional Call Forward To cancel Unconditional Call Forward dial 73 and get the dial tone then hang up Busy Call Forward To use this feature dial 90 and get the dial tone Dial the forward number and for a dial tone and then hang up Note Busy forward functions only when all media channels are in use Since GXP 2000 supports up to 11 lines this function will not Cancel Busy Call Forward To cancel Busy Call Forward dial 91 and get the dial tone then hang up Delayed Call Forward To use this feature dial 92 and get the dial tone Dial the forward number and for a dial tone and then hang up Cancel
17. 1s 60 minutes or 1 hour The maximum interval is 65535 minutes about 45 days This parameter defines the local SIP port the GXP 2000 will listen and transmit The default value for Account 1 is 5060 It is 5062 5064 5066 for Account 2 Account 3 and Account 4 respectively T1 is an estimate of the round trip time RTT between the client and server transactions If the network latency is high select bigger value for reliable usage This element sets the value of the SIP protocol T2 timer in seconds Timer T2 defines the retransmit interval for INVITE responses and non INVITE requests The SIP protocol default value 1s 4 seconds 46 NAT Traversal Subscribe for MWI PUBLISH for Presence Proxy Require Voice Mail User ID Send DTMF Early Dial Dial Plan Prefix Enable Call Features This parameter defines whether the GXP 2000 NAT traversal mechanism will be activated or not If activated by choosing Yes and a STUN server is also specified then the GXP 2000 will behave according to the STUN client specification Under this mode the embedded STUN client inside the GXP 2000 will attempt to detect if and what type of firewall NAT it is sitting behind through communication with the specified STUN server If the detected NAT 1s a Full Cone Restricted Cone or a Port Restricted Cone the GXP 2000 will attempt to use its mapped public IP address and port in all of its SIP and SDP messages If the NAT Traversal field 1s
18. Delayed Call Forward To cancel this Forward dial 93 and get the dial tone then hang up 4 6 Customized idle screen GXP 2000 supports the feature that allows customer to customize the idle LCD screen display Please send email to support grandstream com for detailed information 4 7 XML phone book downloading GXP 2000 supports the feature that allow the user to download XML phone book Please send email to support grandstream com for detailed information 23 5 Configuration Guide 5 1 Configuration with Keypad When the phone is on hook press the MENU button to enter MENU mode When the phone goes off hook or a call comes in the phone automatically exits the MENU state and prepares for the call Here are the Menu options supported Menu Functions Display Call History Press Menu button to enter this menu including Received Calls or Dialed Calls or Missed Calls or Back Press or T to toggle the selection Press to return to the upper menu Display Status Press Menu button to enter this menu to see the status of the phone Press or T to toggle the selection Press Menu or amp button to exit Display Phone Book Press Menu button to display the phone book including Download Phonebook Press or T to toggle the selection Press Menu button to choose the menu item Press button to return to the upper menu Display Instant Messages
19. Factory Reset 2 Step 2 Key in the MAC address printed on the bottom of the sticker on the back Please use the following mapping 0 9 0 9 22 press the 2 key twice A will show on the LCD 222 2222 33 press the 3 key twice D will show on the LCD 33 293 0ooocp A B C D E 3 xt For example if the MAC address is 000582006395 it should be key in as 0002228200333395 NOTE lf there are digits like 22 in the MAC you need to type 2 then press gt right arrow key to move the cursor or wait for 4 seconds to continue to key in another 2 3 Step 3 Press the round menu key again to move the cursor to OK button Press round menu key again to confirm If the MAC address is correct the phone will reboot Otherwise it will exit to previous keypad menu interface 54 Appendix I 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 pecifically IPv4 to map IP network addresses to t
20. IP T2 Interval NAT Traversal STUN PUBLISH for Presence L NO G Yes ay e g MyCompany alk bagi asl ed e g sip nycompany com or IP address e g proxy myprovider com or IP address if any the user part of an SIP address can be identical to or different from SIP User D purposely not displayed for securit protection optional e g John Doe U No E ves U No LE ves UL No E ves U No E ves 9 dn minutes default 1 hour max 45 days 99 default 5060 L No E No but send keep alive u Yes Ee No E ves I 43 SUBSCRIBE for MWI Proxy Requtre Voice Mail UserID Send DTMF Early Dial Dial Plan Prefix Enable Call Features Disable Missed Call Session Expiration Min SE Caller Request Timer Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher Force INVITE Enable 100rel Account Ring Tone E No L Yes User ID extension for 3rd party voice mail system L in audio via RTP RFC2833 via SIF INFO No L Yes use Yes only if proxy supports 484 response this prefix string 1s added to each dialed number E No L Yes Gf Yes Call Forwarding amp Call Waiting Disable are supported locally U No E ves Missed calls NOT recorded 10 a in seconds default 180 seconds E 1n seconds default and minimum 90 seconds I5 No Li Yes Request for timer when making outbound calls No requ
21. New Firmware Li Check New Firmware only when F W pre suffix changes s Always Skip the Firmware Check Authenticate Conf File E No Yes cfg file would be authenticated before acceptance if set to Yes Enable Phonebook XML Download L No vgs urTP YES TFTP Phonebook XML Server Path Phonebook Download Interval 0 720 Remove Manually edited entries on Download s No L ES Phonebook XML Download 36 Enable Idle Screen XML Download Idle Screen XML No E YES HTTP YES TFTP Download Idle Screen XML Server Path DTMF Payload Type id Syslog Server Po Syslog Level NONE e NTP Server timenistgov URI or IP address Allow DHCP Option 42 to override NTP server No Yes Custom ring tone 1 used if incoming caller ID is in ae Custom ring tone 2 used 1f incoming caller ID 1s Distinctive Ring Tone TANE ORE AU g Custom ring tone 3 used if incoming caller ID is Disable Call Waiting E yy E yes Disable Call Waiting tz Tone Use Quick IP call mode NO s Yes s No s Yes Lock keypad update No Yes configuration update via keypad is disabled if set to Yes Admin Administrator password Only administrator can configure the Advanced Settings Password page Password field is purposely left blank for security reason after clicking update and saved The maximum password length is 25 characters G723 rate Encoding rate for G723 codec By defau
22. P 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 behaviour 1s 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 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 ie 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 1s 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 standar
23. User Manual GXP 2000 Enterprise IP Phone For Firmware Version 1 1 1 14 Grandstream Networks Inc www grandstream com NETWORKS e A stea Table of Contents Lo GG OM aE tester olas bitte teur E UMP Ra I EH NEN 4 PB EIS TT em 5 21 Whatis Included im the Pack aCe usse ee Ere ert oU None ES Eee cx D Vere iones 5 2 2 COnme Cin amp Your PHONG i ssissssvecseconsaossedasssunconnsenanasveannensanseacnsgeassnsiee totg ER CEP V Ia ERE d 5 Ps NAIM OI NN 6 24 CGXAP2000 Extension Board asera N N 7 2 Sale COMPO MAN CSS aroen a 9 ZO Wara eiee Ta tire N E T E EN AEE 9 PEOC EON IVA N a E E E E E OAO 11 SS EE Ari E o SES 12 3 2 HardWare SpEcCITICdLTODisse donet eia diesen Cete Batu E n Dae Men Odes drum 13 Z sme GX P 2 O00 IP PRONG iseci A 15 dE TS sake AM ee E bbb EE D see cesses tes ues erase Peu orale stib P on Sonst iu obddru d dose 15 42 Getting Familiar with Keypdadasascsasnmevneiceiutea en doter e baa exo ot mtra Ea 16 A MEE GEI EIUS ODE BIG US 18 4 4 Making and Answering Phone Calls eeeeeeeeeeeeseeeeeeeernnnn 18 4 4 1 Handset Speakerphone and Headset Mode eeeeeeeesssss 18 4 4 2 Multiple SIP Accounts and Lines seen 18 AAS MaRS C AI REUNIR RI TET 19 4 4 4 Making Calls using IP Address ecccceccccceeeeceesseeeeeccceeeeeeaaaeneeees 19 BD SIC CS URGE T TUTTO 20 OR RS UN TAO es AAE E AAT EE EAA A E E A E 20 4 4 7 Call Waiting and Switch
24. an enterprise with large deployment of GXP 2000 can easily manage the configuration and service provisioning of individual devices remotely from a central server Grandstream provides a licensed provisioning system that can be used to support automated configuration of GXP 2000 It uses enhanced NAT friendly TFTP or HTTP thus no NAT issues and other communication protocols to communicate with each individual GXP 2000 for firmware upgrade remote reboot etc Grandstream provide redirection service to VoIP service providers It could be either simple redirection or with certain special provisioning settings Initially upon booting up Grandstream devices by default point to Grandstream provisioning server based on the unique MAC address of each device our redirection service provision the devices with redirection settings so that they will be redirected to customer s TFTP or http server for further provisioning Grandstream also provide GAPSLite software package which contains our NAT friendly TFTP server and a configuration tool to facilitate the task of generating device configuration files The GAPSLite configuration tool is now free to end users The tool and configuration templates can be downloaded from http www grandstream com DOWNLOAD Configuration_Tool For details on how redirection service works please refer to the documentation of provisioning product 51 6 Firmware Upgrade Software upgrade can be done via either TFTP
25. an hear what you said from the speaker of the handset audio part of your phone works fine Press Menu button to exit the mode Display Diagnostic Mode Press Menu button to enter this mode all LEDs will light up Press any key on the keypad the button name will be displayed in the LCD Lift and put back the handset or press Menu button to exit the diagnostic mode Display Factory Reset please be very CAREFUL here Key in the physical MAC address on back of the phone Press Menu button phone will be reset back to FACTORY DEFAULT setting and all your setting will be erased Please refer to Section 7 for complete details 26 5 2 Configuration with Web Browser GXP 2000 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 Microsoft s IE 5 2 1 Access the Web Configuration Menu The IP Phone Web Configuration Menu can be accessed by the following URI http Phone IP Address where the Phone IP Address is the IP address of the phone When the phone is on hook press Menu button and then select the Status item to see IP IP Address NOTE e To type IP address into browser to get into the configuration page 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 192 166 1 14 5 2 2 End User Configur
26. ath is the TFTP or HTTP server path for configuration file It needs to be set to a valid URL either in FQDN or IP address format The Config Server Path can be same or different from the Firmware Server Path A configuration parameter is associated with each particular field in the web configuration page A parameter consists of a Capital letter P and 2 to 3 Could be extended to 4 1n the future digit numeric numbers 1 e P2 is associated with Admin Password in the ADVANCED SETTINGS page For a detailed parameter list please refer to the corresponding firmware release configuration template When Grandstream Device boots up or reboots it will issue request for configuration file named C gXXXXXXXXXXXX Where C xxxxxxxxxxxx is the MAC address of the device 1e cfg000b820102ab The configuration file name should be in lower cases 6 3 Firmware and Configuration File Prefix and Postfix The firmware release 1 1 1 14 allow user to customize the firmware file name by adding prefix and postfix for both firmware and configuration file Firmware Prefix and Postfix allows device to download the firmware name with the matching Prefix and Postfix This makes it the possible to store ALL of the firmware with different version in one single directory Similarly Config File Prefix and Postfix allows device to download the configuration file with the matching Prefix and Postfix Thus multiple configuration files for the same device can
27. ation Once this HTTP request is entered and sent from a Web browser the GXP 2000 will respond with the following login screen Password Login The password is case sensitive with maximum length of 25 characters and the factory default password for End User is 123 2l After a correct password is entered in the login screen the embedded Web server inside the GXP 2000 will respond with the Configuration page which is explained in details below RN End User l purposely not displayed for security protection Password A dynamically assigned via DHCP default or PPPoE will attempt PPPoE if DHCP fails and following is non blank PPPoE account ID PPPoE password Preferred DNS server o fe Jo Jo La statically configured as IP Address 192 168 jo 160 Subnet Mask Jo o jo Jo Default Router fo fo jo DNS Server 1 jo o J Jo DNS Server 2 Jo o fo Jo Multi Key Mode Speed Dial Account Account 1 Purpose Key 1 Name UserID Multi Key Mode Speed Dial Account Account 1 Purpose l l Key 2 Name UserID Key 3 Name UserID Multi Key Mode Speed Dial Account Account Purpose Key 4 Name UserID 28 Multi Purpose Key 5 Multi Purpose Key 6 Multi Speed Dial M Account 1 M Key Mode Account Name UserID Key Mode Speed Dial B Account 1 Name UserID Speed Dial Account 1 Key Mode Account
28. by GXP 2000 to display the current date time Customer Ring Tone to 3 with associate Caller ID when selected if Caller ID is configured then the device will ONLY sound this ring tone when the incoming call is from the Caller ID device will use System Ring Tone for all other calls When selected but no Caller ID 1s configured the selected ring tone will be used for all incoming calls Disable Call Default is No Waiting 4 Disable Call Waiting Tone Quick IP call mode Lock keypad update Syslog Server Syslog Level Allow DHCP Option 66 to override server Allow DHCP Option 42 to override NTP server Default is No This feature is dedicated for busy sales service call centers The call waiting sound is turned off but LINE LED flashing still can be seen and can take the call by pressing the LINE button Please refer user manual chapter 4 4 4 If this parameter is set to Yes the configuration updates via keypad for Menu Item 7 9 12 are disabled The IP address or URL of System log server This feature is especially useful for ITSP Internet Telephone Service Provider Select the ATA to report the log level Default is NONE The level is one of DEBUG INFO WARNING or ERROR Syslog messages are sent based on the following events e product model version on boot up INFO level e NAT related info INFO level e sent or received SIP message DEBUG level e SIP message summary INFO level e i
29. d news servers ECHO CANCELLATION H 323 HTTP 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 A suite of standards for multimedia conferences on traditional packet switched networks oy IP Hyper Text Transfer Protocol the World Wide Web protocol that performs the request and retrieve functions of a server Internet Protocol A packet based protocol for delivering data across networks IP PBX IP based Private Branch Exchange IP Telephony IVR MTU NAT NTP 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 transla
30. dard is a multicarrier time division multiple access time division duplex MC TDMA TDD radio 55 DNS DID DSP DIMF FQDN FXO 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 multiframe 16 frames Voice signals are digitally encoded into a 32 kbit s signal using Adaptive Differential Pulse Code Modulation Short for Domain Name System or Service or Server an Internet service that translates domain names into IP addresses 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 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 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 a
31. dized 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 REC 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 59 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 1s defined in RFC 3489 STUN will usually work good 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 TFTP Trivial File Transfer Protocol is
32. e message Special Feature Default 1s Standard Choose the selection to meet some special requirements from Soft Switch vendors like Nortel Broadsoft etc 5 2 4 Saving the Configuration Changes Once a change is made the user should press the Update button in the Configuration Menu The IP phone will then display the following screen to confirm that the changes have been saved 49 Your configuration changes have been saved They will take effect on next reboot Reboot User is recommended to power cycle the IP phone after seeing the above message 5 2 5 Rebooting the Phone from Remote The administrator of the phone can remotely reboot the phone by pressing the Reboot button at the bottom of the configuration menu Once done the following screen will be displayed to indicate that rebooting is underway The device is rebooting now You may relogin by clicking on the link below in 30 seconds Click to relogin At this point user can relogin to the phone after waiting for about 30 seconds 5 3 Configuration through Central Provisioning Server 50 Grandstream GXP 2000 can be automatically configured from a central provisioning system When GXP 2000 boots up it will send TFTP or HTTP request to download configuration file cfg000b82xxxxxx where 000b82xxxxxx is the MAC address of the GXP 2000 The configuration files can be downloaded via TFTP or HTTP from the central server A service provider or
33. esidential household The following photo illustrates the appearance of a GXP 2000 IP phone Back View 11 3 1 Key Features Grandstream GXP 2000 IP Phone is a next generation enterprise IP telephone based on industry open standard SIP Session Initiation Protocol Built on innovative technology Grandstream IP Phone features market leading superb sound quality and rich functionalities at mass affordable price Software Feature Support SIP 2 0 TCP UDP IP PPPoE RTP RTCP SRTP by SDES HTTP ARP RARP ICMP DNS DHCP NTP TFTP STUN etc Support up to 4 SIP accounts and up to 11 media channels The two Ethernet ports can be configured to function as NAT router Powerful digital signal processing DSP to ensure superb audio quality advanced adaptive jitter control and packet loss concealment technology Support popular codecs including G711 a law and u law G 723 1 6 3K G 726 32K G 729A B and GSM Support Caller ID name display or block Call waiting caller ID Hold Call Waiting Call Transfer consultative blind 3 way conference Call Forward in band and out of band DTMF Do not Disturb SIMPLE PRESENSE phone book downloading customized idle screen Asterisk BLF speed dial paging message waiting indicator etc Support Silence Suppression VAD Voice Activity Detection CNG Comfort Noise Generation Line Echo Cancellation G 168 and AGC Automatic Gain Control Support standard encryption and authenticati
34. est one U No L Yes Use timer even when remote party does not support L uac b uas omit Recommended E yac E UAS When UAC did not specify refresher L Yes When caller supports timer but did not tag E No E ves Always refresh with INVITE insteac of UPDATE U No E Yes La system ring tone E custom ring tone 1 L custom ring tone 2 44 custom ring tone 3 Send Anonymous amp L Yes caller ID will be blocked if set to Yes Auto Answer Li Yes Allow Auto Answer by Call E Info Yes Turn off speaker on remote disconnect Check SIP User ID for E Yes incoming INVITE s Yes PCMU choice 1 choice 5 Preferred Vocoder choice 2 ici choice 6 in listed order choice 3 6725 choice 7 choice 4 ae p choice 8 SRTP Mode And s Enabled but not forced s Enabled ang orce Special Feature Standard 7 Individual Account Settings Account Active This field indicates whether the account 1s active or not The default value for the primary account Account is Yes The default values for the other three accounts are No Account Name A name to identify an account which will be displayed in LCD SIP Server SIP Server s IP address or Domain name provided by VoIP service provider Outbound Proxy IP address or Domain name of Outbound Proxy or Media Gateway or Session Border Controller Used by GXP 2000 for firewall or NAT penetration in different network environment If symmet
35. evice Status page The following is a screen shot of the device Status page Details are explained next MAC Address 00 0B 82 05 11 BC IP Address 10 10 1 3 Product Model GXP2000 Software Version Program 1 0 2 6 Bootloader 1 0 2 3 System Up Time 0 day s 5 hour s 56 minute s Registered Account 1 Yes Account 2 No Account 3 No Account 4 Yes PPPoE Link Up disabled detected NAT type is full cone MAC Address The device ID in HEX format This is a very important ID for ISP troubleshooting IP Address This field shows LAN IP address of GXP 2000 Product Model This field contains the product model info Software Version e Program This is the main software release its number is always used for firmware upgrade e Bootloader This is normally not changed System Up Time This field shows system up time since the last reboot Registered This field indicates whether the device is registered to the SIP server s PPPoE Link Up This field shows whether the PPPoE connection is up if connected to DSL modem 34 Detected NAT Type This field shows what kind NAT the GXP 2000 is connected to via its LAN port It is based on STUN protocol 5 2 3 Advanced User Configuration To login to the Advanced User Configuration page please follow the instructions in section 5 2 1 to get to the following login page The password is case sensitive with a maximum length of 25 characters and the factory default password for Advanced User
36. he current line on HOLD by pressing another available LINE for making or receiving other phone calls 4 4 7 Call Waiting and Switch between Calls GXP 2000 can support up to 11 Lines user can switch to another line for making or answering calls by pressing the line button and automatically put an ACTIVE call on Hold When receiving second or more incoming calls besides playing a stutter Call Waiting tone GXP 2000 will pick up the corresponding account or the next available LINE as described in section 4 4 2 4 4 8 Call Transfer GXP 2000 supports both BLIND and ATTENDED Transfer 1 Blind Transfer When a LINE is ACTIVE user will get a dial tone by pressing the TRNF button and then dial the number and press the SEND button This will transfer the other party in the corresponding LINE to the dialed number 20 2 Attended Transfer When in conversation with an ACTIVE LINE as defined in section 4 3 2 and another LINE that 1s put on hold user can press TRNF button then press the intended LINE that is on HOLD If there is no LINE on HOLD user will need to make a call and thus automatically puts the current ACTIVE LINE on HOLD NOTE e Transferring calls across SIP domains needs to be supported by SIP services 4 4 9 3 Way Conferencing GXP 2000 supports 3 way conferencing With one LINE ACTIVE and another LINE on HOLD press the CONF button then the LINE that 1s on HOLD will join the three parties
37. he dialing account before dialing any digits by pressing the same LINE button one or more times If user continues to press one LINE the selected account will circulate among the registered accounts For example when LINE is pressed LCD displays FWD If LINEI is pressed again LCD displays SIPPHONE and the subsequent call will be made through SIP account 2 For incoming calls if an account is configured and registered all incoming calls for that account will attempt to use its corresponding LINE if it is not in use When the virtually mapped line is in use GXP 2000 will flash the next available LINE from Left to Right then Top to Bottom in red color LINE 5 to 11 cannot be picked like LINE 1 to 4 This happens automatically When an incoming call arrives while all of the 4 LINE 1 4 channels are in use LINES will be selected When all 4 LINE 1 4 channels are in use and user places an active call on hold user can on hook and off hook to activate the next available channel LINES or whatever the next one When any one of the 7 functions keys is associated with a call they function as LINE keys otherwise they function as speed dial keys So when LINE 5 1s in use you cannot use speed dial 1 but speed dial 2 7 still work 18 A LINE is defined as ACTIVE when it is making or receiving a call and its corresponding LINE LED is lit up in solid RED 4 4 3 Making Calls There are many ways to make phone calls
38. he 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 s 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 geneate 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 Networks and wireless local loop The DECT Common Interface radio stan
39. he multi purpose Key on the watcher side will be solid on Once Do not disturb is enable the device will send out a PUBLISH message with status update and the LED on the Watcher side will be turn off Time Zone This parameter controls how the date time is displayed according to the specified time zone 3l Daylight Savings Time This parameter controls whether the displayed time will be daylight savings time or not If set to Yes and the Optional Rule is empty then the displayed time will be 1 hour ahead of normal time The Automatic Daylight Saving Time Rule shall have the following syntax start time end time saving Both start time and end time have the same syntax month day weekday hour minute month 1 2 3 12 for Jan Feb Dec day 1 2 3 31 weekday 1 2 3 7 for Mon Tue Sun or 0 which means the daylight saving rule is not based on week days but based on the day of the month hour hour 0 23 minute minute 0 59 If weekday is 0 it means the date to start or end daylight saving is at exactly the given date In that case the day value must not be negative If weekday is not zero and day is positive then the daylight saving starts on the first day th iteration of the weekday 1st Sunday 3rd Tuesday etc If weekday is not zero and day is negative then the daylight saving starts on the last day th iteration of the weekday last Su
40. irst vocoder choice The maximum value for PCM is 10 x10ms frames for G726 it is 20 x10ms frames for G723 it is 32 x30ms frames for G729 G728 64 x10ms and 64 x2 5ms frames respectively This field defines the layer 3 QoS parameter which can be the value used for IP Precedence or Diff Serv or MPLS Default value is 48 This contains the value used for layer 2 VLAN tag Default setting 1s blank Default is 4 seconds This parameter allows users to configure the key to be used as the Send or Dial key If set to Yes pressing this key will immediately trigger the sending of dialed string collected so far In this case this key is essentially equivalent to the Re Dial key If set to No this 7 key will then be included as part of the dial string to be sent out This parameter defines the local RTP RTCP port pair the GXP 2000 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 This parameter when set to Yes will force random generation of both the local SIP and RTP ports This is usually necessary when multiple GXP 2000s are behind the same NAT This parameter specifies how often the GXP 2000 sends a blank UDP packet to the SIP server in order to keep the hole on the NAT open Default i
41. is admin Password Login Advanced User configuration includes not only the end user configuration but also advanced configuration such as SIP configuration Codec selection NAT Traversal Setting and other miscellaneous configuration Following is a screen shot of the advanced configuration page purposely not displayed for security protection Admin Password G723 rate 6 3kbps encoding rate H 5 3kbps encoding rate Silence Suppression No LU Yes 2 up to 10 20 32 64 for G711 G726 G723 other codecs respectively Layer 5 QoS ii Diff Serv or Precedence value Layer 2 QoS ET VLAN Tag i 802 1p priority value 0 Voice Frames per TX 35 No Key Entry Timeout k in seconds default is 4 seconds Use as Dial Key L No E Yes 1f set to Yes will function as the Re Dial key local RTP port 1024 65535 default 5004 Use random port N E Yes keep alive interval 2 in seconds default 20 seconds if specified this will be used in SIP SDP message STUN server URI or IP port Firmware Upgrade and Upgrade Via E TprTp E HTTP Provisioning Use NAT IP Firmware Server Path Config Server Path Firmware File Prefix Firmware File Postfix Config File Prefix Config File Postfix Allow DHCP Option 66 to override server U No E ves Automatic Upgrade me E No E Yes check for upgrade every E minutes default 7 days 9 Always Check for
42. is feature needs the support of IP PBX Turn off speaker on Default is No If set to Yes the speaker will turn off and the phone will go remote disconnect back to idle status after the other party of the call hands up Check SIP User ID Default is No If set to Yes this account will deny direct IP call and the call for incoming to previous configured account will be refused also INVITE Preferred Vocoder The GXP 2000 supports up to 5 different Vocoder types including G 711 A U law GSM G 723 1 G 729A B User can configure Vocoders in a preference list that will be included with the same preference order in SDP message The first Vocoder in this list can be entered by choosing the appropriate option in Choice 1 Similarly the last Vocoder in this list can be entered by choosing the appropriate option in Choice 8 SRTP Mode e Disabled GXP2000 will use RTP for both inbound and outbound calls Default setting e Enabled but not forced GXP2000 provides crypto suites to others SIP phones If they support SRTP then use SRTP otherwise use RTP e Enabled and forced All calls using this account must use SRTP If the other party doesn t support SRTP GXP2000 will decline the call If we invite others and receive a 200 Ok message without crypto suite we will send a BYE message followed by the ACK message to terminate the call If someone invites us without crypto suite we will response a 488 Not Acceptable Her
43. l transfer Call Forwarding amp Do Not Disturb are supported locally Disable Missed Call Default is No If set to Yes missed calls will not be recorded for your review 47 Session Expiration Min SE Caller Request Timer Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher Force INVITE Enable 100rel Account Ring Tone Send Anonymous Grandstream implemented SIP Session Timer The session timer extension enables SIP sessions to be periodically refreshed via a SIP request UPDATE or re INVITE Once the session interval expires if there is no refresh via a UPDATE or re INVITE message the session will be terminated Session Expiration is the time in seconds at which the session is considered timed out 1f no successful session refresh transaction occurs beforehand The default value is 180 seconds The minimum session expiration in seconds The default value is 90 seconds If selecting Yes the phone will use session timer when it makes outbound calls if remote party supports session timer If selecting Yes the phone will use session timer when it receives inbound calls with session timer request If selecting Yes the phone will use session timer even if the remote party does not support this feature Selecting No will allow the phone to enable session timer only when the remote party support this feature To turn off Session Timer select No f
44. ll Handset Rest Tab E a Tab with Tab with extension down extension up After wall mounting the main body of GXP 2000 user will need to pull out the tab extension downward from handset cradle on the top of the handset rest and rotate the tab and plug it into the slot with the extension up for handset holding 2 4 GXP2000 Extension Board GXP 2000 supports up to 2 extension boards from firmware 1 1 1 1 GXP2000 Extension Board expands more Multi Purpose Keys for advanced business phone applications One GXP2000 Extension Board carries 56 Multi Purpose Keys Simply connect the extension board with GXP2000 via PS2 cable in the same package the GXP2000 will automatically reboot itself and initiate the extension board The GXP2000 will boot up first and all LED indicators on GXP2000 will be solid on for a while the status light at the top right corner of the extension board will blink in red and then all of the LED indicators on the extension board will flash three times and then the status light at the top right corner of the extension board will turn to solid green Note If you are using GXP2000 with HWO 4 power supply in the same package should be plugged in the extension board Hardware version can be found in the GUI STATUS Menu you can use the down arrow key to browse GUI menu when the phone is on hook 2 5 Safety Compliances The GXP 2000 phone is compliant with various safety standards including FCC CE
45. lt 6 3kbps rate is set Silence This controls the silence suppression VAD feature of G723 and G729 If set to Suppression 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 37 Voice Frames per TX Layer 3 QoS Layer 2 QoS No Key Entry Timeout Use as Send Key Local RTP port Use Random Port Keep alive interval Use NAT IP STUN Server This field contains the number of voice frames to be transmitted in a single packet When setting this value the user should be aware of the requested packet time used in SDP message as a result of configuring this parameter This parameter is associated with the first vocoder in the above vocoder Preference List or the actual used payload type negotiated between the 2 conversation parties at run time e g if the first vocoder is configured as G723 and the Voice Frames per TX is set to be 2 then the ptime value in the SDP message of an INVITE request will be 60ms because each G723 voice frame contains 30ms of audio Similarly if this field is set to be 2 and if the first vocoder chosen 1s G729 or G711 or G726 then the ptime value in the SDP message of an INVITE request will be 20ms If the configured voice frames per TX exceeds the maximum allowed value the GXP 2000 will use and save the maximum allowed value for the corresponding f
46. lthough most terminals support only 12 of them 0 9 and 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 56 FXS DHCP 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 amp T They should also operate the same way when connected to an FXS interface An FXO interface will accept calls from FXS or PSTN interfaces All countries and regions have their own standards FXO 1s complimentary to FXS and the PSTN 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 1t 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 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 an
47. many other SIP compliant devices and software on the market Grandstream GXP 2000 has been awarded the Best of Show product in 2005 Internet Telephony Conference and Expo This document is subject to changes without notice The latest electronic version of this user manual is available for download from the following location http www grandstream com user manuals GXP2000 pdf CONFERENCE amp EXPO 2 Installation 2 1 What is Included in the Package The GXP 2000 phone package contains 1 One GXP 2000 Main Case 2 One Handset 3 One Phone Cord 4 One Universal Power Adaptor 5 One Ethernet Cable 2 2 Connecting Your Phone Following is a backside picture of GXP 2000 each connection port is labeled with the name in the following table EXT PC LAN PoE POWER HEADSE The table below describes the connectors on the GXP 2000 phone Extension connection for extended keypad will be EXT implemented in the future 10 100 Switch LAN port for connecting to Ethernet LAN PoE Support PoE 802 3af Draws power from both spare line and signal line PC 10 100 Switch port for connecting PC POWER SV power port HEADSET 3 5mm Headset port 2 3 Wall Mount GXP 2000 can be wall mounted There are two wall mount holes on the bottom of the GXP 2000 main body Top Wall Mount hole Bottom Wall Mount hole User can simply place the device against the wall with two holes placed to the fixed hanger position on the wa
48. mpliance GXP 2000 2xRJ45 10 100Base T with PoE 802 3af IEEE 802 3af standard can draw power from both spare lines or signal lines from Ethernet 3 5mm Headset port 11 LED with different light pattern in RED color Input 100 240VAC 50 60 Hz Output SVDC 1200mA UL certified 215mm W 220mm D 57mm H 0 82kg 1 81bs 40 130 F 5 45 C 10 90 non condensing FCC CE C Tick The picture below shows the handset and headset connectors wiring schema 13 GAPZOO0 handset jack 1 gt P eo m mac d gt mict 4 gt SPK GAP2000 handset plug l Abed Inu d e Le 4 Bey bec mic c mic d gt SPK4 GXP2000 power jack headset Jack Bottom view pc 5y 4 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 1s required if user want to user normal 2 5mm cell phone headset The plug converter can be purchased from any electronics department store like Radio Shack 14 4 Using GXP 2000 IP Phone 4 1 Getting Familiar with LCD GXP 2000 phone has a numeric LCD of 64 rows x 131 columns in pixels Here is the display when all segments illuminate e will 1B SR The LCD 1s equipped with a backlight When the phone is configured properly and in the normal idle state the backlight 1s off Whenever an event occurs the backlight turns
49. nbound and outbound calls INFO level e registration status change INFO level e negotiated codec INFO level e Ethernet link up INFO level e SLIC chip exception WARNING and ERROR levels e memory exception ERROR level The Syslog uses USER facility In addition to standard Syslog payload it contains the following components GS LOG device MAC address error code error message Here is an example May 19 02 40 38 192 168 1 14 GS LOG 00 0b 82 00 a1 be 000 Ethernet link is up DHCP Option 66 is used to identify a TFTP server when the sname field in the DHCP header has been used for DHCP options If you choose yes GXP2000 will use the TFTP server resolved from DHCP instead of the one you specified in the TFTP Server option above DHCP Option 42 specifies a list of IP addresses for Network Time Protocol NTP servers available to the client If you choose yes GXP2000 will use the NTP servers resolved from DHCP instead of the one you specified in the NTP Server option above 42 Four independent SIP accounts each has its own configuration page Their configurations are identical The following is a screen shot of SIP Account 1 settings Account Active Account Name SIP Server Outbound Proxy SIP User ID Authenticate ID Authenticate Password Name Use DNS SRV User ID is phone number SIP Registration Unregister On Reboot Register Expiration local SIP port SIP TI Timeout S
50. nd GXP2000 through the PC port to do the configuration On the other hand it exposes the GXP2000 to others and may cause some security issues for users Default is No If set to Yes The GXP2000 will respond to the PING command from other computers for testing but it also is vulnerable to the DOS attack Default is No Allow the user to set a specific MAC address Set in Hex format Sets the LAN subnet mask Default value is 255 255 255 0 Base IP for the LAN port which function as a Gateway for the subnet Default value is 192 168 2 1 Value is set in units of hours Default value is 120hr 5 Days The time IP address is assigned to the LAN clients Forward all WAN IP traffic to a specific IP address if no matching port is used by GXP 2000 itself or in the defined port forwarding Allow the user to forward a matching TCP UDP port to a specific LAN IP address with a specific TCP UDP port DHCP Option 2 specifies the offset of the client s subnet in seconds from Coordinated Universal Time UTC The offset is expressed as a two s complement 32 bit integer A positive offset indicates a location east of the zero meridian and a negative offset indicates a location west of the zero meridian If you choose yes GX P2000 will use the time offset resolved from DHCP instead of the one you specified in the Time Zone option above 33 In addition to the Basic Settings configuration page end user also has access to the d
51. nday 3rd last Tuesday etc The saving is in the unit of minutes The saving time may also be preceded by a negative sign if subtraction is desired instead of addition The default value for Automatic Daylight Saving Time Rule shall be set to 04 01 7 02 00 10 1 7 02 00 60 which is the rule for US Examples US Canada where daylight saving time is applicable 04 01 7 02 00 10 1 7 02 00 60 This means the daylight saving time starts from the first Sunday of April at 2AM and ends the last Sunday of October at 2AM The saving is 60 minutes 1hour LCD Backlight Always Allow user to keep the LCD backlight on all the time Default is On No Time Display Format LCD time display in 12 hour or 24 hour format 32 Date Display Format Display Clock instead of Date Device Mode WAN side http access Reply to ICMP on WAN port Cloned WAN MAC Addr LAN Subnet Mask LAN DHCP Base IP DHCP IP Lease Time DMZ IP Port Forwarding Allow DHCP Option 2 to override Time Zone setting Allow user to choose among the following three formats Year Month Day Month Day Y ear Day Month Y ear LCD displays clock if set to Yes Default is No This parameter controls whether the device is working in NAT router mode or Bridge mode Need save the setting and reboot the device before the setting start to work If set to Yes user can access the configuration page through the WAN port instead of connecting PC a
52. on DIGEST using MD5 and MD5 sess Support for Layer 2 802 1Q VLAN 802 1p and Layer 3 QoS ToS DiffServ MPLS Support automated NAT traversal without manual manipulation of firewall NAT Provide easy configuration through manual operation phone keypad Web interface or automated provisioning by downloading encrypted configuration file via HTTP TFTP for mass deployment Support firmware upgrade via TFTP or HTTP Support customized configuration firmware file name by attaching prefix and post fix Support GXP 2000 Extension board for multi purpose functionality Support phonebook downloading via HTTP TFTP Support customizable idle screen by downloading XML format file via HTTP TFTP Support SIP MESSAGE method RFC 3428 stores up to 100 incoming IM messages Support SIP PUBLISH method RFC 3903 SIP Presence package RFC 3856 3863 for use of 7 MFKs and GXP 2000EXT SIP Dialog package RFC 4235 Support Power over Ethernet PoE IEEE standard 802 3af Support Headset which will auto switch to Headset when plugged in Support 10 100 Full Half Duplex Ethernet Switch with LAN and PC port Ethernet polarity can be auto detected thus either straight through or twist cable can be used 12 3 2 Hardware Specification The table below describes the hardware specification of GXP 2000 Model LAN interface Power over Ethernet Headset Jack LED Universal Switching Power Adaptor Dimension Weight Temperature Humidity Co
53. or Caller Request Timer Callee Request Timer and Force Timer As a Caller select UAC to use the phone as the refresher or UAS to use the Callee or proxy server as the refresher As a Callee select UAC to use caller or proxy server as the refresher or UAS to use the phone as the refresher Session Timer can be refreshed using INVITE method or UPDATE method Select Yes to use INVITE method to refresh the session timer The use of the PRACK Provisional Acknowledgment method enables reliability to be offered to SIP provisional responses 1xx series This is very important if PSTN internetworking is to be supported A user s wish to use reliable provisional responses is invoked by the 100rel tag which is appended to the value of the required header of initial signalling messages There are 4 different ring tone that are defined e System Ring Tone when selected all calls will ring with system ring tone e Customer Ring Tone 1 to 3 when selected GXP 2000 will ONLY play this ring tone for all the incoming calls for this account If this parameter is set to Yes the From header in outgoing INVITE message will be set to anonymous essentially blocking the Caller ID from displaying 48 Auto Answer When set to Yes GXP 2000 will automatically switch to speaker when there is an incoming call Allow Auto Answer Default is No If set to Yes auto answer depends on the Call Info in the SIP by Call Info message Th
54. pgrade may take as long as 5 to 20 minutes over Internet or just 2 minutes 1f it is performed on a LAN It 1s recommended to conduct firmware upgrade in a controlled LAN environment if possible For users who do not have a local firmware upgrade server Grandstream provides a NAT friendly TFTP server on the public Internet for firmware upgrade Please check the Services section of Grandstream s Web site at http www grandstream com y firmware htm to obtain our public TFTP server s IP address Alternatively user can download a free TFTP or HTTP server and conduct local firmware upgrade A free windows version TFTP server is available for download from http support solarwinds net updates New customerFree cfm Our latest official release can be downloaded from http www grandstream com y firmware htm Unzip the file and put all of them under the root directory of the TFTP server Put the PC running the TFTP server and the GXP2000 in the same LAN segment Please go to File gt Configure gt Security to change the TFTP server s default setting from Receive Only to Transmit Only for the firmware upgrade Start the TFTP server in the GXP2000 s web configuration page configure the Firmware Server Path with the IP address of the PC update the change and reboot the unit 22 6 2 Configuration File Download Grandstream SIP Device can be configured via Web Interface as well as via Configuration File through TFTP or HTTP Config Server P
55. pper menu Display Factory Functions Press Menu to display the factory function items including Ethernet Loopback or Audio Loopback or Diagnostic Mode or Enable Diag Port or Back Press or T to toggle the selection Press to return to the upper menu Display Reboot Press Menu button to reboot the device Display Exit Press Menu button to exit the menu 25 Menu Functions Display Ring Volume Press Menu button to hear the selected ring volume press or gt to hear and adjust the ring tone volume Press Menu button to select and exit take effect immediately Display Ethernet Loopback Press Menu button to enter this mode A cross Ethernet cable 1s needed for the test Before you do the test plug one end of the cable in the PC port and the other end in the LAN port You will see the test result on the screen This is a feature that is useful for factory as well as for tech team if you need to diagnose if the RJ45 jacks are still good in terms of hardware Note Running the Ethernet Loopback mode under normal connection will cause IP lost Press Menu button to exit the diagnostic mode Display Audio Loopback Press Menu button to enter this mode Tap the keypad to check if the speaker plays the sound caused by your tapping If yes the audio part of your phone works fine Or you can pick up the handset and say something to the mic of the handset If you c
56. r e Both phones can be connected through a router using public or private IP addresses To make a direct IP calling disable Use Random Port option at advanced web configuration page and then press Menu button and then select Direct IP Call submenu to enter the direct IP call interface and then enter the 12 digit target IP address and then press Menu button twice to make the call 19 From 1 1 0 13 firmware build GXP2000 begins to offer Quick IP call feature first make Handset SPEAKER Headset off hook and then press key and enter the last 3 digits of the target IP address and then press the SEND key or key To use this feature you need to enable Quick IP call mode in the advanced web configuration page 4 4 5 Receiving Calls There are two states when GXP 2000 receives a call l When receiving an initial call Besides ringing with selected Ring Tone the corresponding account LINE will flash in red taking Handset SPEAKER Headset off hook will enable user to hear the calling party in Handset SPEAK ER Headset 2 When receiving second or more incoming calls besides playing stutter Call Waiting tone GXP 2000 will pick up the corresponding account LINE or the next available LINE as described in section 4 4 2 4 4 6 Call Hold While in conversation pressing the HOLD button will put the other party on hold User can resume the conversation by pressing the corresponding LINE User will also automatically put t
57. ric NAT is detected STUN will not work and ONLY outbound proxy can provide solution for it 45 SIP User ID Authenticate ID Authenticate Password Name Use DNS SRV User ID is Phone Number SIP Registration Unregister on Reboot Register Expiration Local SIP port SIP T1 Timeout SIP T2 Interval User account information provided by VoIP service provider ITSP usually has the form of digit similar to phone number or actually a phone number SIP service subscriber s Authenticate ID used for authentication Can be identical to or different from SIP User ID SIP service subscriber s account password for GXP 2000 to register to SIP servers of ITSP SIP service subscriber s name which will be used for Caller ID display Default is No If set to Yes the client will use DNS SRV to look up server If the GXP 2000 has an assigned PSTN telephone number this field should be set to Yes Otherwise set it to No If Yes is set a user phone parameter will be attached to the From header in SIP request This parameter controls whether the GXP 2000 needs to send REGISTER messages to the proxy server The default setting 1s Yes Default is No If set to yes the SIP user s registration information will be cleared on reboot This parameter allows user to specify the time frequency in minutes that GXP 2000 refreshes its registration with the specified registrar The default interval
58. s 20 seconds NAT IP address used in SIP SDP message Default is blank IP address or Domain name of the STUN server 38 Firmware Upgrade and provisioning Via TFTP Server Via HTTP Server Automatic Upgrade Authenticate Conf File This radio button will enable disable GXP 2000 to download firmware or configuration file through either TFTP or HTTP This is the IP address of the configured TFTP server If selected and it 1s non zero or not blank the GXP 2000 will attempt to retrieve new configuration file or new code image from the specified TFTP server at boot time It will make up to 3 attempts before timeout and then it will start the boot process using the existing code image in the Flash memory If a TFTP server is configured and a new code image is retrieved the new downloaded image will be verified and then saved into the Flash memory Note Please do NOT interrupt the TFTP upgrade process especially the power supply as this will damage the device Depending on the network environment this process can take up to 15 or 20 minutes The URL for the HTTP server used for firmware upgrade and configuration via HTTP For example http provisioning mycompany com 6688 Grandstream 1 0 5 16 Here 6688 is the specific TCP port that the HTTP server is listening at it can be omitted if using default port 80 Note If Auto Upgrade is set to No GXP 2000 will only do HTTP download once at boot up Choose Yes to
59. s on the right 7 Multi functional keys with LED that can be configured to use for speed dial Asterisk BLF presence watcher Scroll up Menu item when phone is in MENU mode Or increase handset speakerphone volume when phone is ACTIVE Or access the missed calls menu when phone is in IDLE mode Scroll down Menu item when phone is in MENU mode Or reduce handset speakerphone volume when phone is ACTIVE Or access the Phone Book when phone is in IDLE mode Shift cursor to left Shift cursor to right Enter MENU mode when phone is in IDLE mode It is also the ENTER key once entering MENU Transfer an ACTIVE call to another number Bring Calling Called party into conference Enter to retrieve voice mails or other messages Mute an ACTIVE call or Delete a key entry call log voice mail and etc Or use of MUTE DEL key during incoming call ringing state to reject call using SIP 486 message Or act as toggle key to turn DND on and off during idle Temporarily hold an ACTIVE call Enter hands free mode Dial a new number or Redial the last number dialed After entering the phone number pressing this key would force a call to go out immediately before timeout 12 standard Digit and keys are usually used to make phone calls 17 4 3GUI Menu Chart Please see the Appendix II 4 4Making and Answering Phone Calls 4 4 1 Handset Speakerphone and Headset Mode The regular Handset mode can be switched with either the Speaker mode Hand free or
60. set to Yes with no specified STUN server the GXP 2000 will periodically every 20 seconds or so send a blank UDP packet with no payload data to the SIP server to keep the hole on the NAT open Default is No When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically GXP2000 supports SIP PUBLISH for Presence RFC 3903 If SIP PUBLISH for Presence on GXP2000 1s enabled the status of GXP2000 can be monitored by other SIP devices Please refer to Presence Watcher function on Multi Purpose Keys The status of GXP2000 can be toggled by enabling disabling Do Not Disturb via the GUI Menu The default status of GXP2000 is Do Not Disturb disabled The LED of the multi purpose key on the watcher side will be solid on Once Do not disturb 1s enable the device will send out a PUBLISH message with status update and the LED on the Watcher side will be turn off SIP Extension to notify SIP server that the unit is behind the NAT Firewall When configured user will be able to dial voice mail server by pressing MSG button This parameter specifies the mechanism to transmit DTMF digit There are 3 modes supported in audio which means DTMF is combined in audio signal not very reliable with low bit rate codec via RTP RFC2833 or via SIP INFO Default 1s No Use only if proxy supports 484 response Sets the prefix added to each dialed number Default is No If set to Yes Cal
61. the resolution of the screen XML should be within this range The following 2 options are added to the Preference LCD GUI submenu Download SCR XML Erase Custom SCR User will have to choose to Downl oad SCR XML to start the download process Once the XML 1s successfully downloaded it will be effective right away The file will be saved and loaded automatically after reboot Example XML file of gs screen xml lt xml version 1 0 gt lt Screen gt dl eScreen gt lt ShowStatusLine gt false lt ShowStatusLine gt Di spl ayBi t map Bitmap Put your customized screen file with bitmap format here lt Bit map gt lt X gt 0 lt X cns Di spl ayBi t map DisplayString font f8 halign Right gt lt DisplayStr gt Doraemon lt DisplayStr gt lt X gt 130 lt X lt Y gt 0 lt Y gt lt DisplayString gt DisplayString font f10 halign Left valign Bottom gt lt DisplayStr gt Call me lt DisplayStr gt EX hai NS lt Y gt 54 lt Y Di spl ayString DisplayString font f8 halign Left valignz Bottom Di splayStr XQ V Di spl ayStr EKNE o lt Y gt 64 lt Y gt lt DisplayString gt lt dl eScreen gt Screen Note The feature requires some expertise on XML For more technical details please contact Grandstream This parameter sets the payload type for DTMF using RFC2833 This parameter defines the URI or IP address of the NTP Network Time Protocol server which is used
62. the Headset mode however whenever the Headset 1s plugged in Speaker mode will be switched to the Headset mode automatically To Switch between Handset and Speaker Headset simply press the Hook Flash in the Handset cradle or the Speaker button 4 4 2 Multiple SIP Accounts and Lines GXP 2000 can support up to 4 independent SIP accounts Each account 1s capable of independent SIP Server user and NAT settings among others GXP 2000 supports up to 11 concurrent audio channels arbitrarily assigned to these SIP accounts they can be used in any combination as long as the server allows it Speed dial numbers configured must be associated to a specific SIP account Each of the 4 LINE buttons LINEI LINEA is virtually mapped to each SIP account In off hook state when user chooses an idle line the name of the account as configured in the web interface will be displayed in the LCD while a dial tone is being played out For example if the 4 SIP accounts are named FWD SIPPHONE BROADVOICE and PBX respectively and they are all active and registered When LINE is pressed user will hear dial tone and see FWD When LINE2 is pressed user will hear dial tone and see SIPPHONE When LINE3 is pressed user will hear dial tone and see BROADVOICE When LINEZA is pressed user will hear dial tone and see PBX For outgoing calls GXP 2000 will pick up the LINE pressed which will be lit up in solid red color User can switch t
63. tion 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 is a software application that accepts a combination of voice telephone input and touch tone keypad selection and provides appropriate responses in the form of voice fax callback e mail and perhaps other media A Maximum Transmission Unit MTU 1s 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 Network Address Translation 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 58 Outbound Proxy or another name Session Border Controller A device used in VoIP networks OB

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