Grandstream Networks GXW400x User's Manual
Contents
1. andstream Innovative IP Voice 8 Video TABLE 3 GXW 400X SOFTWARE FEATURES GXW 400x FXS Analog Gateway Series GXW 4004 4 ports 2 SIP accounts amp profiles GXW 4008 8 ports 2 SIP accounts amp profiles Network Interface Two 2 10M 100 Mbps RJ 45 LED Indicators Power and Line LEDs Voice over Packet Voice Activity Detection VAD with CNG comfort noise generation and Capabilities PLC packet loss concealment Dynamic Jitter Buffer Modem detection amp auto switch to G 711 Packetized Voice Protocol Unit supports RTP RTCP and AAL2 protocol G 168 compliant Echo Cancellation LEC line echo cancellation with NLP PSTN Fail over PSTN failover on power failure Voice Compression G 711 Annex I PLC Annex II VAD CNG format encoder and decoder G 723 1A G 726 ADPCM G 729A B E GSM BC G 726 provides proprietary VAD CNG and signal power estimation Voice Play Out unit reordering fixed and adaptive jitter buffer clock synchronization AGC automatic gain control Status output Decoder controlling via voice packet header DHCP Server Client Yes NAT Router or Switched Mode i FaxoverlP n T 38 compliant Group 3 Fax Relay up to 14 4kpbs and auto switchto G 711 for Fax Pass through pending Fax Datapump V 17 V 19 V 27ter V 29 for T 38 fax relay QS Diffserve TOS 802 1P QVLANtaggng IP Transport sid RTPIRTCP DTMF Method flexible DTMF transmission method User inter2. button on the analog phone if the phone has that button Press the flash button again to release the previously held Caller and resume conversation If no flash button is available use hook flash toggle on off hook quickly You may drop a call using hook flash CALL WAITING Call waiting tone series of quick consecutive dial tones during conversation indicates an incoming call if the call waiting feature is enabled Press the flash button to place the current caller on hold and switch to the other call Press the flash button to toggle between two active calls CALL TRANSFER Blind Transfer Assume that call Caller A and B are in conversation A wants to Blind Transfer B to C 1 Caller A presses FLASH on the analog phone to hear the dial tone 2 Caller A dials 87 then dials caller C s number and then or wait for 4 seconds 3 Caller A can hang up NOTE Enable Call Feature must be set to Yes in web configuration page Caller A can place a call on hold and wait for one of three situations 1 A quick confirmation tone similar to call waiting tone followed by a dial tone This indicates the transfer is successful transferee has received a 200 OK from transfer target At this point Caller A can either hang up or make another call 2 A quick busy tone followed by a restored call on supported platforms only This means the transferee has received a 4xx response for the INVITE and we will tr
3. ndstream Innovative IP Voice amp Video Grandstream Networks Inc Analog IP Gateway GXW 400x 4 or 8 FXS Ports GXW 400x User s Manual www grandstream com Firmware Version 1 0 0 17 support grandstream com andstream Innovative IP Voice amp Video TABLE OF CONTENTS GXW 400X USER MANUAL MELC eck wicca tik ia ka tl e e et i ne wie kn EE 4 CONNECTING YOUR PHONE ei ei s isi kwak ad at n ok at a as a a n ne a n ek e kk A n e a a a a Ka nnn 5 EGUIPME NT PACKAGING EE 5 SAFET COMPLAN ES monia n dean cps tea kota ta te aki tad wade it ii aa a ne v ka EE e L n e kte Gan ka 6 MALAD dn e Si cee ae ay n etat ee E pi ki e a a ki a ah a vase tk a a a e ek AN pk ae a oka 6 CONNECTING ERENNERT kk n kk n a e a l t ek e ba k t 5 APPLICATION DESCRIPTION WEE EE 7 GXW 400X FEATURES 000 ta tata tata t atanta nannan aaa aa aaa aaa aa aa aa aa aa aa aa aa nana nana nana nana na nasasasesesesesesosososososounnnnnnnnn 8 SOFTWARE FEATURES OVERVIEW Edge ki ak ken e kk ka kk l ad et ek a ek a ka kk a e a e e det 8 HARDWARE er Te hg e LEE 10 USING THE WEB CONFIGURATION PAGES ccccececececececececececeeeeeecececeeeneeseeeeeeeeneeeeeeeneeececeeeeeeeeeaeanans 11 CONFIGURING THE GX W 40OX ccccccccecececececececscccccccccccccacacececucscucacacacacacaeaeauauauauauaeceaecucenenecuaucucecuceaeaes 11 IMPORTANT SE TIN aac ee eae sete ees sis aces ate an li Facto wees bce nine Sales ane Se Geers ee cece dace cerns dio mans e an e B
4. When selected but no Caller ID is configured the selected ring tone will be used for all incoming calls Default is No Default is No Incoming call will stop ringing when not picked up given a specific period of time Default is 4 seconds Default is No Use only if proxy supports 484 response This parameter controls whether the phone will send an early INVITE each time a key is pressed when a user dials a number If set to Yes an INVITE is sent using the dial number collected thus far Otherwise no INVITE is sent until the Re Dial button is pressed or after about 5 seconds have elapsed if the user forgets to press the Re Dial button The Yes option should be used ONLY if there is a SIP proxy configured and the proxy server supports 484 Incomplete Address response Otherwise the call will likely be rejected by the proxy with a 404 Not Found error This feature is NOT designed to work with and should NOT be enabled for direct IP to IP calling Sets the prefix added to each dialed number GXW 400x User Manual Firmware 1 0 0 17 Page 23 of 31 Last Updated 12 2006 Use as Dial Key Subscribe for MWI Send Anonymous Special Feature Preferred Vocoder G723 Rate ILBC Frame Size iLBC Payload type G726 16 Payload type G726 24 Payload type G726 32 Payload type G726 40 Payload type G729E payload type VAD Fax Mode Jitter Buffer Type Jitter Buffer Lengt
5. Default is No If set to Yes the client will use DNS SRV to look up server lf the GXW 400x 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 GXW 400x needs to send REGISTER messages to the proxy server The default setting is Yes Default is No If set to Yes the SIP user s registration information is cleared on reboot Default is No If set to Yes user can place outgoing calls even when not registered if allowed by ITSP but is unable to receive incoming calls Allows the user to specify the time frequency in minutes for the GXW 400x to refresh its registration with the specified registrar The default interval is 60 minutes or 1 hour The maximum interval is 65535 minutes about 45 days Defines the local SIP port the GXW 400x 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 Defines the local RTP RTCP port pair the GXW400x 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 Used for call transfer incase s
6. address of GXW 400x Product Model Contains the product model info Software Version Program This is the main software release Boot and Loader are not changed often System Up Time Shows system up time since the last reboot PPPoE Link Up Shows whether the PPPoE connection is running if connected to DSL modem Detected NAT Type Shows type of NAT the GXW 400x is connected to via its LAN port It is based on STUN protocol Port Indicates whether the different FXS Ports are registered to the SIP server s ADVANCED USER SETTINGS ADVANCED USER CONFIGURATION To access the advanced user configuration page follow the directions for Using the Web Configuration Pages Advanced User configuration includes the end user configuration and the advanced configurations including SIP configuration Codec selection NAT Traversal Setting and other miscellaneous configuration FIGURE 3 ADVANCED USER CONFIGURATION SCREENSHOT Password Login Grandstream Networks Inc GXW 400x User Manual Page 19 of 31 Firmware 1 0 0 17 Last Updated 12 2006 GH Innovative IP Voice amp Video TABLE 9 ADVANCED CONFIGURATION PAGE DEFINITIONS Admin Administrator password Only the administrator can configure the Advanced Settings Password page Password field is purposely left blank for security reasons after clicking update and saved The maximum password length is 25 characters Layer 3 QoS This field defines the layer 3 QoS paramete
7. e This document is subject to change without notice The latest electronic version of this user manual is available for download from the following location http www grandstream com user_manuals GXW_ 400x pdf e Reproduction or transmittal of the entire or any part in any form or by any means electronic or print for any purpose without the express written permission of Grandstream Networks Inc is not permitted Grandstream Networks Inc GXW 400x User Manual Page 4 of 31 Firmware 1 0 0 17 Last Updated 12 2006 dstream Innovative IP Voice amp Video CONNECTING YOUR GXW 400x Connecting your GXW 400x is easy Before you begin please verify the contents of the GXW 400x package EQUIPMENT PACKAGING Unpack and check all accessories Equipment included in the package e one 12V universal power adapter e one Ethernet cable e one device unit CONNECT THE GXW 400x Managing the GXW and connecting the unit to the VoIP network is very simple Follow these four 4 steps to connect your GXW to the Internet and access the unit s configuration pages The GUI pages will guide you through the remaining steps to set up your gateway Examples of the GUI Interfaces can be downloaded from http www grandstream com GUI GUI GXW400x rar Connect the Power to the GXW 400x unit Connect an Ethernet cable between the LAN port on GXW 400x to your PC Your PC will be assigned an IP address like 192 168 2 100 Launch web browser and
8. the forwarding number followed by H Wait for dial tone then hang up 91 Cancel Busy Call Forward dial 91 Wait for dial tone Hang up 92 Delayed Call Forward Dial 92 for a dial tone Dial the forwarding number followed by H Wait for dial tone then hang up 93 Cancel Delayed Call Forward Dial 93 for a dial tone then hang up Flash Hook If user hears call waiting beep flash hook will switch to the new incoming call Also used to switch to a new channel for a new call PSTN PASS THROUGH LIFE LINE The RJ 11 line jack on the GXW 400x side will function as a pass through jack when GXW 400x is out of power The pass through life line enables the user to use the analog phone for PSTN calls directly without using an access code T 38 FAX SUPPORT GXW 400x supports fax in two modes 1 T 38 Fax over IP and 2 fax pass through T 38 is the preferred method because it is more reliable and works well in most network conditions If the service provider supports T 38 please use this method by selecting T 38 as fax mode default If the service provider does not support T 38 pass through mode may be used To send or receive faxes in fax pass through mode users must select all the Preferred Codecs to be PCMU PCMA G 711 u a End User Configuration Once this HTTP request is entered and sent from a Web browser the GXW 400x will respond with a login screen There are two default passwords for the login page Us
9. value is 103 Range is from 96 to 127 Default value is 102 Range is from 96 to 127 Default is No T 38 Auto Detect FolP by default or Pass Through must use codec PCMU PCMA Select either Fixed or Adaptive based on network conditions Select Low Medium or High based on network conditions Dependent on standard phone type and location e Bellcore Telcordia e ETSI DTMF e ETSI FSK e SIN 227 BT e NTT Japan GXW 400x User Manual Firmware 1 0 0 17 Page 24 of 31 Last Updated 12 2006 andstream Innovative IP Voice 8 Video Polarity Reversal Default is No If set to Yes polarity will be reversed upon call establishment and termination Hook Flash timing Gain Handset volume adjustment RX is for receiving volume TX is for transmission volume Default values are 0dB for both parameters Loudest volume 6dB Lowest volume 6dB Call Progress Configure ring or tone frequencies according to preference By default tones are Ring Tones set to North American frequencies Frequencies should be configured with known values to avoid uncomfortable high pitch sounds SAVING THE CONFIGURATION CHANGES Once a change is made press the Update button in the Configuration Menu The GXW 400x will display the following screen to confirm that the changes have been saved Reboot or power cycle the GXW 400x after all the changes are made so that those changes can take effect FIGURE4 SAVING CONFIGURATION CHANGE
10. Hz 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 The ability for an outside caller to dial to a PBX extension without going through an attendant or auto attendant DSP 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 Grandstream Networks Inc GXW 400x User Manual Page 28 of 31 Firmware 1 0 0 17 Last Updated 12 2006 andstream Innovative IP Voice 8 Video 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 ww
11. S SCREENSHOT Your configuration changes have been saved They wil take effect on next reboot Reboot Grandstream Networks Inc GXW 400x User Manual Page 25 of 31 Firmware 1 0 0 17 Last Updated 12 2006 andstream Innovative IP Voice amp Video FIRMWARE UPGRADE Our latest official release can be downloaded from hitp www grandstream com y firmware htm To upgrade your unit firmware follow these steps 1 Under Advanced Settings webpage enter your TFTP or HTTP Server IP address or FQDN next to the Firmware Upgrade Upgrade Server field Select via TFTP or HTTP accordingly If you plan to use Automatic Upgrade set it to Yes otherwise No this will make it check for upgrade every time you reboot UPGRADE THROUGH HTTP To upgrade firmware via HTTP the field Firmware Upgrade and Provisioning Upgrade Via needs to be set to HTTP The Firmware Server Path should be set to where the firmware files are located For example the user can use the following URL in the Firmware Server Path firmware mycompany com 6688 Grandstream 1 0 0 29 where firmware mycompany com is the FQDN of the HTTP server It can also be in IP address format 6688 is the TCP port the HTTP server listening to default http server listens to port 80 Grandstream 1 0 0 29 is the RELATIVE directory to the root dir on HTTP web server UPGRADE THROUGH TFTP To upgrade firmware via TF TP set the field Firmware
12. This parameter controls whether the device is working in NAT router mode or Bridge mode Save the setting and reboot prior to configuring the GXW lf set to Yes user can access the configuration page through the WAN port instead of through the PC port Warning this configuration is less secure than default option Default is No If set to Yes the GXW400x will respond to the PING command from other computers but it also is vulnerable to the DOS attack Default is No Enables 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 functions as a Gateway for the subnet Default value is 192 168 2 1 Value is set in units of hours Default value is 120 hrs 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 GXW400x or defined in port forwarding Forwards a matching TCP UDP port to a specific LAN IP address with a specific TCP UDP port In addition to the Basic Settings configuration page the end user also has access to the Device Status page Grandstream Networks Inc GXW 400x User Manual Firmware 1 0 0 17 Page 18 of 31 Last Updated 12 2006 andstream Innovative IP Voice amp Video TABLE 8 STATUS PAGE DEFINITIONS MAC Address The device ID in HEX format This is needed for ISP troubleshooting WAN IP Address Shows LAN IP
13. Upgrade and Provisioning Upgrade Via to TFTP The TFTP server can be configured in either IP address format or FQDN To configure the TFTP server via the Web configuration interface follow these five steps 1 Open your browser to input the IP address of the GXW 400x 2 Enter the admin password to enter the configuration screen 3 Enter the TFTP server address or URL in the Firmware Server Path field near the bottom of the configuration screen 4 Once the Firmware Server Path is set update the change by clicking the Update button 5 Reboot or power cycle the unit If the configured updating server is found and a new code image is available the GXW 400x will retrieve the new image files by downloading them into the GXW 400x s SRAM During this stage the GXW 400x s LED will blink until the checking downloading process is completed Upon verification of checksum the new code image will be saved into the Flash If TFTP fails for any reason e g TFTP server is not responding there are no code image files available for upgrade or checksum test fails etc the GXW 400x will stop the TFTP process and simply boot using the existing code image in the flash A firmware upgrade may take as long as 20 minutes over the Internet or 20 seconds if performed on a LAN Grandstream recommends conducting firmware upgrades in a controlled LAN environment if possible No LOCAL TFTP SERVER For users who do not have a l
14. ages are sent based on the following events 1 3 4 5 6 T 8 9 1 0 product model version on boot up INFO level NAT related info INFO level sent or received SIP message DEBUG level SIP message summary INFO level inbound and outbound calls INFO level registration status change INFO level negotiated codec INFO level Ethernet link up INFO level SLIC chip exception WARNING and ERROR levels 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 Example May 19 02 40 38 192 168 1 14 GS LOG 00 0b 82 00 a1 be 000 Ethernet link is up TABLE 10 FXS PORTS CONFIGURATION DEFINITIONS FXS Port FXS Port Number SIP User ID User account information provided by VoIP service provider ITSP Usually in the form of digit similar to phone number or actually a phone number Authenticate ID SIP service subscriber s Authenticate ID used for authentication Can be identical to or different from SIP User ID Password SIP service subscriber s account password for GXW 400x to register to SIP servers of ITSP Name Name Profile ID Select the corresponding Profile ID 1 2 GXW 400x User Manual Page 21 of 31 Grandstream Networks Inc Firmware 1 0 0 17 Last Updated 12 2006 Am Innovative IP Voice amp Video TABLE 11 PROFILE PAGE DEFINITIONS Profile A
15. call can be completed if 1 Both GXW 400X and other VoIP Device have public IP addresses or 2 Both GXW 400X and other VoIP Device are on the same LAN using private IP addresses or 3 Both GXW 400X 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 call 1 Pick up the analog phone then access the voice menu prompt by dial 2 Dial 47 to access the direct IP call menu 3 Enter a 12 digit target IP address after the dial tone and voice prompt Direct IP Calling Destination ports can be specified by using 4 encoding for followed by the port number NOTE When completing direct IP call the Use Random Port should set to NO You can not make direct IP calls between FXS1 to FXS2 since they are using same IP Grandstream Networks Inc GXW 400x User Manual Page 14 of 31 Firmware 1 0 0 17 Last Updated 12 2006 andstream Innovative IP Voice amp Video EXAMPLES OF DIRECT IP CALLS 1 Ifthe target IP address is 192 168 0 10 the dialing convention is Voice Prompt with option 47 then 192 168 000 010 followed by the key or wait 4 seconds 2 If the target IP address port is 192 168 1 20 5062 then the dialing convention is Voice Prompt with option 47 then 192168001020 45062 followed by the key or wait 4 seconds CALL HOLD Place a call on hold by pressing the flash
16. can be omitted if using default port 80 Note If Auto Upgrade is set to No GXW 400x will only do HTTP download once at boot up Automatic Choose Yes to enable automatic upgrade and provisioning In Check for new Upgrade firmware every field enter the number of days to enable GXW 400x to check the server for firmware upgrade or configuration in the defined period of days When set to No GXW 400x will only do upgrade once at boot up Always check for New Firmware Check New Firmware only when F W pre suffix changes Authenticate lf set to Yes config file is authenticated before acceptance This protects the Conf File configuration from an unauthorized change Firmware Key For firmware encryption It should be 32 digit in Hexadecimal Representation End user should keep it blank Lock Keypad lf set to Yes the configuration update via keypad is disabled Update NTP server URI or IP address of the NTP Network Time Protocol server Used by the phone to synchronize the date and time Grandstream Networks Inc GXW 400x User Manual Page 20 of 31 Firmware 1 0 0 17 Last Updated 12 2006 andstream Innovative IP Voice 8 Video Syslog Server The IP address or URL of System log server This feature is especially useful for the ITSP Internet Telephone Service Provider Syslog Level Select the GXW 400x to report the log level Default is NONE The level is one of DEBUG INFO WARNING or ERROR Syslog mess
17. ctive SIP Server Outbound Proxy SIP transport NAT Traversal Use DNS SRV User ID is Phone Number SIP Registration Unregister on Reboot Outgoing Call w o Registration Register Expiration Local SIP port Local RTP Port Refer to Use Target Contact When set to Yes the SIP Profile is activated SIP Server s IP address or Domain name provided by VoIP service provider IP address or Domain name of Outbound Proxy or Media Gateway or Session Border Controller Used by GXW 400x for firewall or NAT penetration in different network environments If symmetric NAT is detected STUN will not work and ONLY outbound proxy can correct the problem User can select UDP or TCP or TLS This parameter defines whether the GXW 400x NAT traversal mechanism is activated or not If activated by choosing Yes and a STUN server is also specified then the GXW 400x performs according to the STUN client specification Under this mode the embedded STUN client will detect if and what type of firewall NAT is being used If the detected NAT is a Full Cone Restricted Cone or a Port Restricted Cone the GXW 400x will use its mapped public IP address and port in all of its SIP and SDP messages If the NAT Traversal field is set to Yes with no specified STUN server the GXW 400x 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
18. e 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 touch tone keypad selection and provides appropriate responses in the form of voice fax callback e mail and perhaps other media Grandstream Networks Inc GXW 400x User Manual Page 29 of 31 Firmware 1 0 0 17 Last Updated 12 2006 andstream Innovative IP Voice 8 Video 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 VolP networks OBP SBCs are put into the signaling and media path between calling and called Caller The OBP SBC acts as if it was the called VoIP phone and plac
19. e selection Static IP Mode configure all the IP address information through menu options 02 to 05 Dynamic IP Mode IP address information is retrieved from the DHCP server automatically at reboot The current WAN IP address is announced Enter 12 digit new IP address if in Static IP Mode Same as Menu option 02 Same as Menu option 02 Same as Menu option 02 Enter 9 to go to the next selection in the list e PCMU PCMA e G 723 e G 729 e iLBC e G 726 32 e G 28 e G 29E Announces the Mac address of the unit Enter 9 to toggle between enable and disable Announces current Firmware Server IP address Enter 12 digit new IP address Announces current Config Server Path IP address Enter 12 digit new IP address Upgrade protocol for firmware and configuration update Enter 9 to toggle between TFTP and HTTP Firmware version information Firmware upgrade mode Enter 9 to rotate among the following three options 1 always check 2 check when pre suffix changes 3 never upgrade User will be prompted with a dial tone to dial a 12 digit IP address to make a direct IP call For details see Make a Direct IP Call Enter 9 to reboot the device or Enter MAC address to restore factory default setting Automatically returns to Main Menu Page 13 of 31 Firmware 1 0 0 17 Last Updated 12 2006 andstream Innovative IP Voice 8 Video Five Success Tips when using the Voice Pro
20. e successfully registered You can now use your standard phones connected to ports FXS1 to FXS8 to make calls Ext 5 Profile1 Ext 6 Profile1 i 604 604 Ext4 Profile1 i i i IMPORTANT SETTINGS The end user must configure the following settings according to the local environment NOTE Most settings on the web configuration pages are set to the default values NAT SETTINGS lf you plan to keep the gateway within a private network behind a firewall we recommend using STUN server The following three 3 settings are useful in the STUN Server scenario 1 STUN Server under Advanced Settings webpage Enter a STUN Server IP or FQDN that you may have or look up a free public STUN Server on the internet and enter it on this field If using Public IP keep this field blank 2 Use Random Ports under Advanced Settings webpage This is a tricky setting It really depends on your network settings so set this parameter to Yes or No whichever works If using a Public IP address set this parameter to No 3 NAT Traversal under the Profile web pages Set this to Yes when gateway is behind firewall on a private network Grandstream Networks Inc GXW 400x User Manual Page 11 of 31 Firmware 1 0 0 17 Last Updated 12 2006 andstream Innovative IP Voice 8 Video DTMF METHODS DTMF Settings are in Profile pages e DTMF in audio e DTMF via RTP RFC2833 e DTMF via SIP INFO Enable one or more DIMF methods based on your PBX sys
21. el 11 RE RRE 11 ETAN MET EE 12 FREFERRED VOGODE CODEC E 12 BASIC OPERATIONS eege dis klas t l e ka ik a dE aj 13 UNDERSTANDING GXW VOICE bpOoMpTs nran oraraa Daaraan annaa a nanne annarra nann nne 13 PLACING PHONE CALLS EE 14 CALLING PHONE OR ESTENSIONNUMBERS 14 DIRECTII Ee 14 EXAMPLES OF DIRECT IP Ca cts cs l a ea e a etn a A cee 15 CAL T e EE 15 CP MWA TIN EE 15 CA TEE 15 3 WAY CONFERENCING it ase tate ad to oe n tn a la e kk a en al an ks ka no an an a a e pi ente ski ae 16 GABI FE ATORE EE E EE 17 PSTN PASS THROUGH LIFE UNE 17 T 38 FAX GUppPORT A OALA DAAO EA REA DEAL EACEACAADAEADA ADER DEEDAR EAR EE REAREA nanan 17 END USER CONFIGURATION ccccccececececccecececececececeeneeeneceneaeeeaeaeneeeneeeneeneneneneneneaeneeeaeaeaeaeeceneneneaeasans 17 ADVANCED USER SETTINGS ccccccccececececececececececeeeeeeeececeneaeeeeeeeeeeeneeeeeeneneneneneneaeaeeeaeaeaeaeaenececeeeeeeeanans 19 ADVANCED USER CONFIGURATION EE 19 SAVING THE CONFIGURATION CHANGE GS kk tk kk kk e ea ea ea ee aa aa aa ae aa ae aaa aa aaeaeaaaaaaesesesesesesesesesasasaoaoaoaosoooooooaoenn 25 FIRMWARE UPGRADE EE 26 UPGRADE THROUGH HIT 26 UPGRADE THROUGH IEIR 26 NOLOCAL RE 26 RESTORE FACTORY DEFAULT SETTING Q ccccccececececececececececeeeeeeeecececeaeeeeeeeeeeeneeeeeeeneceeeeececeeeeeeeneasens 27 FAGTORY EFE DE EE 27 GLOSSARI OF eu LEE 28 Grandstream Networks Inc GXW 400x User Manual Page 2 of 31 Firmware 1 0 0 17 Last Updated 12 2006 ands
22. er Level Password Web pages allowed End User Level Only Status and Basic Settings Administrator Level All pages can be browsed Grandstream Networks Inc GXW 400x User Manual Page 17 of 31 Firmware 1 0 0 17 Last Updated 12 2006 Am Innovative IP Voice amp Video TABLE 7 BASIC SETTINGS PAGE DEFINITIONS End User Password Web Port IP Address Time Zone Self Defined Time Zone 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 Password to access the Web Configuration Menu This field is case sensitive with a maximum length of 25 characters By default HTTP uses port 80 This field is for customizable web port There are two modes to operate the GXW 400x DHCP mode all the field values for the Static IP mode are not used even though they are still saved in the Flash memory The GXW 400x acquires its IP address from the first DHCP server it discovers from the LAN it is connected Using the PPPoE feature set the PPPoE account settings The GXW 400x will establish a PPPoE session if any of the PPPoE fields is set Static IP mode configure the IP address Subnet Mask Default Router IP address DNS Server 1 primary DNS Server 2 Secondary fields These fields are set to zero by default Controls how the date time is displayed according to the specified time zone TBD
23. erver requires this setting GXW 400x User Manual Grandstream Networks Inc Page 22 of 31 Firmware 1 0 0 17 Last Updated 12 2006 DTMF Payload Type DTMF in audio DTMF via RTP DTMF via SIP INFO Send Flash Event Enable Call Features Off Hook Auto Dial Proxy Require Use NAT IP Distinctive Ringtone Disable Call Waiting Disable Call Waiting Tone Ring Timeout No Key Entry Timeout Early Dial Dial Plan Prefix Grandstream Networks Inc GH Innovative IP Voice amp Video Sets the payload type for DIMF using RFC2833 send DTMF as inband in audio send DTMF via RTP According to RFC 2833 send DTMF via SIP INFO message Default is NO If set to yes flash will be sent as DTMF event Default is No lf set to Yes Call Forwarding 8 Do Not Disturb are supported locally Allows the user to configure a User ID or extension number to be automatically dialed upon off hook Only the user part of a SIP address needs to be entered here The phone will automatically append the Q and the host portion of the corresponding SIP address SIP Extension to notify SIP server that the unit is behind the NAT Firewall NAT IP address used in SIP SDP message Default is blank CustomRing Tone 1 to 3 with associate Caller ID when selected if Caller ID is configured then the device will ONLY uses this ring tone when the incoming call is from the Caller ID System Ring Tone is used for all other calls
24. es a second call to the called Caller 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 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 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 publi
25. face of In audio RFC2833 and or SIP Info IP Signaling SIP RFC 3261 Provisioning TFTP HTTP HTTPS pending Control TLS SIPS Management Syslog support HTTPS and telnet pending remote management using Web browser Power t S Output 12VDC Input 100 240 VACIS0 60HZ ss S Environmental Operational 32 104 F or 0 40 C Be Storage 10 130 F Humidity 10 90 Non condensing t Dimensions HxWxD 35x 135x225mm Mounting si Rack mount Wall mount Desktop i isstsi lt is s Short and long haul REN3 Up to150 ft on 24 AWG line CallerlD sid Bellcore Type 1 8 2 ETSI BT NTT and DTMF based CID iw Polarity Reversal Wink YES On EMC EN55022 EN55024 and FCC part15 Class B Seen e Grandstream Networks Inc GXW 400x User Manual Page 9 of 31 Firmware 1 0 0 17 Last Updated 12 2006 andstream Innovative IP Voice 8 Video HARDWARE FEATURES e 4or8FXS ports e Two RJ 45 switched or routed e Auto detect Ethernet polarity e PC Console connection e PSTN fail over port TABLE 4 HARDWARE SPECIFICATION OF GXW 400x LAN interface 2 x RJ45 10 100Mbps LED 8 LEDs GREEN Universal Switching Input 100 240V AC 50 60Hz 0 5A Max Power Adaptor Output 12V DC 1 25A UL certified Dimension 225mm L x 135mm W x 35mm H Weight 0 29 Ibs 3 5 oz Temperature 32 104 F 0 40 C Humidity 10 90 non condensing Compliance FCC CE Grandstream Networks Inc GXW 400x User Manual Page 10
26. ffice connecting analog lines directly to the gateway Grandstream Networks Inc GXW 400x User Manual Page 7 of 31 Firmware 1 0 0 17 Last Updated 12 2006 andstream Innovative IP Voice 8 Video SCENARIO THREE GXW 410x 8 GXW 400x SCENARIO TOLL FREE CALLING BETWEEN LOCATIONS Branch A Boston MA 4 employees FXO Lines Branch B Denver CO GXW 400x FEATURES The GXW 400x is feature rich and is a next generation IP voice gateway that is interoperable and compatible with leading IP PBXs SoftSwitches and SIP platforms The GXW 400x FXS Gateway series is auto configurable remotely manageable and scalable There are two FXS models the GXW 4004 and GXW 4008 each offering superb voice quality traditional telephony functionality easy deployment and 4 or 8 FXS ports respectively Each model features flexible dialing plans PSTN failover integrated call routing to support a pure IP network call and an external power supply SOFTWARE FEATURES OVERVIEW e Two SIP accounts amp profiles e Multi party conferencing e Supports Voice Codecs G711 a p Annex amp II G723 1A G726 ADPCM G728 G729 A B E iLBC and GSM e 1 38 Fax e G 168 Echo Cancellation e Voice Activity Detection VAD with CNG comfort noise generation and PLC packet loss concealment e Supports PSTN PBX analog telephone sets or analog trunks Grandstream Networks Inc GXW 400x User Manual Page 8 of 31 Firmware 1 0 0 17 Last Updated 12 2006
27. h SLIC Setting Caller ID Scheme Grandstream Networks Inc Am Innovative IP Voice amp Video Allows users to configure the key as the Send or Dial key If set to Yes H will send the number In this case this key is essentially equivalent to the Real key If set to No this key can be included as part of number Default is No When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically lf this parameter is set to Yes the From header in outgoing INVITE message will be set to anonymous blocking Caller ID Default is Standard Choose the selection to meet some special requirements from Softswitch vendors The GXW 400x supports up to 5 different Vocoder types including G 711 A U law GSM G 723 1 G 729A B The user can configure Vocoders in a preference list that will be included with the same preference order in SDP message The first Vocoder is entered by choosing the appropriate option in Choice 1 The last Vocoder is entered by choosing the appropriate option in Choice 8 Defines the encoding rate for G 723 vocoder By default 6 3kbps rate is chosen Sets the iLBC frame size in 20ms or 30ms Defines payload type for iLBC Default value is 97 The valid range is between 96 and 127 Default value is 98 Range is from 96 to 127 Default value is 99 Range is from 96 to 127 Default value is 100 Range is from 96 to 127 Default
28. 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 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 interchangeabl
29. log IP Gateway and make the best use of its many upgraded features including simple and quick installation 3 way conferencing and direct IP IP Calling This IP Analog Gateway is very easy to manage and scalable specifically designed to be an easy to use and affordable VoIP solution for the small medium business or enterprise Gateway GXW 400x Overview The new GXW 400x series has a compact and quiet design no fans and offers superb audio quality rich feature functionality strong security protection and good manageability It is auto configurable remotely manageable and scalable The GXW 400x features 4 or 8 port FXS interface for analog telephones dual 10M 100Mbps network ports with integrated router PSTN life line in case of power failure and an RS232 serial port for administration In addition it supports 2 SIP account profiles caller ID for various countries regions T 38 fax 1 or 2 stage dialing flexible dialing plans security protection SIPS TLS comprehensive voice codecs including G 711 a u law G 723 1 G 726 G 728 G 729A B E and iLBC 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 e This document is contains links to Grandstream GUI Interfaces Please remember to download these examples http www grandstream com GUI GUI GXW 400x for your reference
30. mpt 1 shifts down to the next menu option 2 returns to the main menu 3 9 functions as the ENTER key in many cases to confirm an option 4 All entered digit sequences have known lengths 2 digits for menu option and 12 digits for IP address For IP address add 0 before the digits if the digits are less than 3 i e 192 168 0 26 should be key in like 192168000026 No decimal is needed 5 Key entry can not be deleted but the phone may prompt error once it is detected PLACING PHONE CALLS CALLING PHONE OR EXTENSION NUMBERS There are currently two methods to make an extension number call 1 Dial the numbers directly and wait for 4 default seconds 2 Dial the numbers directly and press assuming that use as dial key is selected in web configuration Examples e Todial another extension on the same proxy such as 1008 simply pick up the attached phone dial 1008 and then press the or wait for 4 seconds e TodialaPSTN number such as 6266667890 you may need a prefix number followed by the phone number Please check with your VoIP service provider for correct information If your phone is assigned a PSTN like number such as 62655567839 it is likely that you dial 1 prior to the number 16266667890 DIRECT IP CALLS Direct IP calling allows two parties that is a FXS Port with an analog phone and another VoIP Device to talk to each other in an ad hoc fashion without a SIP proxy A direct IP
31. ocal TFTP server Grandstream provides a NAT friendly TFTP server on the public Internet for users to download the latest firmware upgrade automatically Please check the Services section of Grandstream s Web site to obtain this TFTP server IP address Alternatively user can download and install a free TF TP or HTTP server in his LAN for a firmware upgrade A free Windows version TFTP server can be downloaded from http support solarwinds net updates New customerFree cfm Grandstream Networks Inc GXW 400x User Manual Page 26 of 31 Firmware 1 0 0 17 Last Updated 12 2006 andstream Innovative IP Voice amp Video Directions For Downloading TFTP Server 1 Unzip the file and put all of the files under the root directory of the TFTP server 2 Put the PC running the TFTP server and the GXW 400x in the same LAN segment 3 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 4 Start the TFTP server in the phone s web configuration page Configure the Firmware Server Path with the IP address of the PC 6 Update the change and reboot the unit on You can also download the free HTTP server from hitp httpd apache orq or just use Microsoft IIS web RESTORE FACTORY DEFAULT SETTING WARNING Restoring the Factory Default Setting will DELETE all configuration information of the phone Please BACKUP or PRINT out all the settings bef
32. of 31 Firmware 1 0 0 17 Last Updated 12 2006 andstream Innovative IP Voice 8 Video USING THE WEB CONFIGURATION PAGES Connect the Power to the GXW 400x unit Connect an Ethernet cable between the LAN port on GXW 400x to your PC Your PC will be assigned an IP address like 192 168 2 100 Launch web browser and type hitp 192 168 2 1 at Address of web browser This will let you to connect to build in web server of GXW 400x 5 Login to the GUI using the password admin for the Advanced Configuration Page or 123 for the Basic Configuration Page SC E Se CONFIGURING THE GXW 400x 1 There are six different tabs Status Basic Settings Advanced Settings Profile 1 Profile 2 and FXS Ports on the top of the screen after login To open each page click on the tab 2 Click on Profile 1 to enter your SIP Server SIP Proxy Registrar information Enter the IP Address or FQDN of the Server under SIP Server and or Outbound Proxy 3 Click on FXS ports to enter the extensions or account information You will need to fill in the following information for each extension Once the extensions are configured you are finished 601 601 602 602 603 603 Ext 1 Profile1 Ext 2 Profile1 Ext 3 Profile1 605 605 6 607 607 Ext 7 Profile 8 608 608 Ext 8 Profile 4 Click Update after changing any setting and then Re boot to confirm changes 5 After reboot check the Status Page to confirm the extensions ar
33. ore you approach to following steps Grandstream will not take any responsibility if you lose all the parameters of setting and cannot connect to your VoIP service provider FACTORY RESET The only way to reset default factory settings is to follows these four 4 steps 1 Unplug the Ethernet cable 2 Locate a needle sized hole on the back panel of the gateway unit next to the power connection Insert a pin in this hole and press for about 7 seconds Take out the pin All unit settings are restored to factory settings Grandstream Networks Inc GXW 400x User Manual Page 27 of 31 Firmware 1 0 0 17 Last Updated 12 2006 andstream Innovative IP Voice 8 Video 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
34. panel Busy ON Solid Green Available OFF NOTE All LEDs display green when ON SAFETY COMPLIANCES The GXW 400x is compliant with various safety standards including FCC CE Its power adaptor is compliant with UL standard WARRANTY Grandstream has a reseller agreement with our reseller customer End users 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 fora RMA Return Materials Authorization number Grandstream reserves the right to remedy warranty policy without prior notification Warning Please do not use a different power adaptor with the GXW 400x as it may cause damage to the products and void the manufacturer warranty Grandstream Networks Inc GXW 400x User Manual Page 6 of 31 Firmware 1 0 0 17 Last Updated 12 2006 andstream Innovative IP Voice 8 Video APPLICATION DESCRIPTION There are three scenarios where the GXW 400x series can be effectively used to enable any business to leverage the benefits of VolP and the Internet SCENARIO ONE GXW 400x FXS GATEWAY CONFIGURATION PBX SCENARIO VoIP Anywhere in the world FXS Trunks Analog 4 or 8 Ports nee SCENARIO TWO GXW 400x FXS GATEWAY CONFIGURATION NO PBX SCENARIO VOIP Anywhere in the world PSTN Analog 4 or 8 Ports Lines GXW FXS Series FXS Gateway acts as a virtual o
35. r which can be the value used for IP Precedence or Diff Serv or MPLS Default value is 48 Layer 2 QoS Value used for layer 2 VLAN tag Default setting is blank STUN Server IP address or Domain name of the STUN server Keep alive This parameter specifies how often the GXW 400x sends a blank UDP packet to the interval SIP server to keep the hole on the NAT open Default is 20 seconds Firmware Enables GXW 400x to download firmware or configuration file through either the TFTP Upgrade amp or HTTP server Provisioning Via TFTP Server This is the IP address of the configured TFTP server If selected and it is non zero or not blank the GXW 400x retrieves the new configuration file or new code image from the specified TFTP server at boot time After 5 attempts the system will 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 is saved into the Flash memory Note Please do NOT interrupt the TF TP 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 Via HTTP Server The URL for the HTTP server used for firmware upgrade and configuration via HTTP For example ttp provisioning mycompany com 6688 Grandstream 1 0 0 36 6688 is the specific TCP port where the HTTP server is listening it
36. s call then Caller A presses flash to bring Parties B amp C into conference If Caller C does not answer the call Caller A can press flash to resume call with Caller B oe Men Bellcore Style 3 way Conference The GXW 400x also supports Bellcore style 3 way Conference Ensure to enable Use Bellcore Style 3 way Conference under Profile Page Follow directions for Star Code Style Grandstream Networks Inc GXW 400x User Manual Page 16 of 31 Firmware 1 0 0 17 Last Updated 12 2006 andstream Innovative IP Voice amp Video CALL FEATURES The GXW 400x supports traditional and advanced telephony features These features are easy to use with any phone TABLE 6 CALL FEATURES TABLE STAR CODE COT Call Features 23 3 way Conferencing 30 Block CalleriD for all config change 31 Send CalleriD for all config change 67 Block CalleriD per call 82 Send CalleriD per call 50 Disable Call Waiting for all config change 51 Enable Call Waiting for all config change 70 Disable Call Waiting Per Call 71 Enable Call Waiting Per Call 72 Unconditional Call Forward Dial 72 for a dial tone Dial the forwarding number followed by Wait for dial tone and hang up dial tone indicates successful forward 73 Cancel Unconditional Call Forward Dial 73 and wait for dial tone then hang up 87 Blind Transfer 90 Busy Call Forward Dial 90 for a dial tone Dial
37. shed 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 TFTP Trivial File Transfer Protocol is 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 Grandstream Networks Inc GXW 400
38. tem PREFERRED VOCODER CODEC The GXW 400x supports a broad range of voice codecs Under Profile web pages choose your preferred order of different codecs e PCMU or G711yu e PCMA or G711a e G729 A B E e G723 e G726 32 e iLBC e G728 e GSM Grandstream Networks Inc GXW 400xUserManual Page 12of31 Firmware 1 0 0 17 Last Updated 12 2006 BASIC OPERATIONS andstream Innovative IP Voice 8 Video UNDERSTANDING GXW VOICE PROMPTS GXW 400x has a stored voice prompt menu for quick browsing and simple configuration Enter on the standard analog phone connected to any FXS port to use the voice prompt menu TABLE 5 DEFINITIONS OF THE GXW VOICE PROMPTS Menu Voice Prompt User s Options Main Menu 01 02 03 04 05 07 10 12 13 14 15 16 17 47 99 Enter a Menu Option DHCP Mode or Static IP Mode IP Address IP address Subnet IP address Gateway IP address DNS Server IP address Preferred Vocoder MAC Address WAN Port Web Access Firmware Server IP Address Configuration Server IP Address Upgrade Protocol Firmware Version Firmware Upgrade Direct IP Calling RESET Invalid Entry Grandstream Networks Inc GXW 400x User Manual Enter for the next menu option Enter to return to the main menu Enter 01 06 47 86 or 99 Menu option Enter 9 to toggle th
39. tream Innovative IP Voice 8 Video TABLE OF FIGURES GXW 400X USER MANUAL FIGURE 1 DIAGRAM OF GXW 400X BACK PANEL cccccccceecceeceeeceueceuuceueceueeeueeeueeeeueeeueeeueeaueeeeueaeeeuueeueeeaeeeuaes 5 FIGURE 2 DIAGRAM OF GXW 400X DISPLAY DANEL rinrin rre rre renni 6 SCENARIO ONE GXW 400x FXS GATEWAY CONFIGURATION PBX SCENARIO VOIP ccecceceeceeceeeeeeeeeeeess 7 SCENARIO TWO GXW 400x FXS GATEWAY CONFIGURATION No PBX SCENARIO MO 7 SCENARIO THREE GXW 400x amp GXW 400X SCENARIO TOLL FREE CALLING BETWEEN LOCATIONS as 8 FIGURE 3 ADVANCED USER CONFIGURATION DCREENSHOT ninna nnn rreren nne 19 FIGURE4 SAVING CONFIGURATION CHANGES SCREENSHOT see NNN 25 TABLE OF TABLES GXW 400X USER MANUAL TABLE 1 DEFINITIONS OF THE GXW CONNECTORS tieri trke tnan nunn AnA AREER EERE Annne Ennn Ennan nennen 5 TABLE 2 DEFINITIONS OF THE GXW DISPLAY DANEL 6 TABLE 3 GXW 400xX SOFTWARE FEATURES a es otila aie ske ia ass Gn li n l is ki l k e bn e ekip ao b deans 9 TABLE 4 HARDWARE SPECIFICATION OF GXW 40OX 0eatteereooeooouooouoooeeooueoooeoooeoooeoouooonoooneoooeoooooooeooooeoneonne 10 TABLE 5 DEFINITIONS OF THE GXW VOICE pop 13 TABLE 6 CALL FEATURES TABLE STAR CODE 17 TABLE 7 BASIC SETTINGS PAGE DEFINITIONS oisicseeseiseisesnsrieriurinrinnieninni erken an nu nn nnna EARE EREEREER EEn n Ennan Enana 18 TABLE 8 STATUS PAGE DEFINITIONS cccccccccecceecceecececeuceueeeuuceueceueueuuueueueuueueeuuueusue
40. type hitp 192 168 2 1 at the address of web browser This will give you access to the GXW 400x web server ae E FIGURE 1 DIAGRAM OF GXW 400x BACK PANEL GXW 400x MN Fes Fxs2 FESI Fes FISS 8x56 FHG FESE PSTN Power FXS LAN WAN Console Line Supply Connection Ports RJ 45 Ethernet PC Ports connection TABLE 1 DEFINITIONS OF THE GXW CONNECTORS LAN or PC Connect your PC to the LAN to find IP address from your Router DHCP Server WAN or LAN Connect to the internal LAN network or router VIDEO IN Connection for Analog based Video Surveillance Camera RCA RESET Factory Reset button Press for 7 seconds to reset factory default settings POWER IN Power adapter connection CONSOLE Console connection for a PC FXS1 FXS8 FXS ports to be connected to physical PSTN lines Grandstream Networks Inc GXW 400x User Manual Page 5 of 31 Firmware 1 0 0 17 Last Updated 12 2006 Innovative IP Voice amp Video Once the GXW is turned on and configured the front display panel indicates the status of the unit FIGURE 2 DIAGRAM OF GXW 400x DISPLAY PANEL GXW 400x Display LEDs FXS port status TABLE 2 DEFINITIONS OF THE GXW DISPLAY PANEL Power LED Indicates Power Remains ON when Power is connected and turned ON Ready LED Remains ON after boot up LAN LED Indicates LAN or WAN port activity PC LED Indicates PC or LAN port activity LEDs 1 8 Indicate status of the respective FXS Ports on the back
41. ueueueeuueueeeueeeueueueeeuuas 19 TABLE 9 ADVANCED CONFIGURATION PAGE DEFINITIONS ccccccccecsceeseueseeueeeseueeuueeuueeeuueeueeeueeaueeaueeueeenees 20 TABLE 10 FXS PORTS CONFIGURATION DEFINITIONS 0e tteerteeoeeoooeoooueooeeooeeoooeoooeoooeoonooonooonooooeooneoooeeooeeooeonnre 21 TABLE 11 PROFILE PAGE DEFINITIONS acs iiseaces sess cette akite lf enn l eat av E 22 GUI INTERFACES GXW 400X USER MANUAL http www grandstream com GUI GUI GXW400x rar SCREENSHOT OF ADVANCED SETTINGS CONFIGURATION PAGE SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE SCREENSHOT OF FXS PORTS CONFIGURATION PAGE SCREENSHOT OF PROFILE 1CONFIGURATION SCREENSHOT OF PROFILE 2 CONFIGURATION SCREENSHOT OF STATUS CONFIGURATION LOGIN PAGE So So e Grandstream Networks Inc GXW 400x User Manual Page 3 of 31 Firmware 1 0 0 17 Last Updated 12 2006 andstream Innovative IP Voice 8 Video WELCOME Thank you for purchasing the Grandstream GXW 400x IP Analog FXS Gateway The GXW 400x is a cost effective easy to use and easy to configure IP communications solution for any business The GXW 400x supports popular voice codecs and is designed for full SIP compatibility and interoperability with 3 party SIP providers thus enabling you to fully leverage the benefits of VoIP technology integrate a traditional phone system into a VoIP network and efficiently manage communication costs This manual will help you learn how to operate and manage your GXW FXS Ana
42. 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 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 sub sampling 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 standard is a multi carrier time division multiple access time division duplex MC TDMA TDD radio transmission technique using ten radio frequency channels from 1880 to 1930 M
43. w is the host Grandstream is the second level domain and and com is the top level domain FXS Foreign eXchange Office An FXS 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 e An FXS interface will accept calls from FXS or PSTN interfaces All countries and regions have their own standards e FXS is complimentary to FXS and the PSTN FXS Foreign eXchange Station An FXS device has hardware to generate the ring signal to the FXS extension usually an analog phone e An FXS device will allow any FXS device to operate as if it were connected to the phone company This makes your PBX the POTS PSTN for the phone e The FXS Interface connects to FXS devices by an FXS 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
44. x User Manual Page 30 of 31 Firmware 1 0 0 17 Last Updated 12 2006 andstream Innovative IP Voice 8 Video 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 is usually refer to the IEEE 802 1Q tagging protocol VoIP Voice over the Internet VolP encompasses many protocols All the protocols do some form of signaling of call capabilities and transport of voice data from one point to another e g SIP H 323 etc Grandstream Networks Inc GXW 400x User Manual Page 31 of 31 Firmware 1 0 0 17 Last Updated 12 2006
45. y to recover the call The busy tone is just to indicate to the transferor that the transfer has failed 3 Continuous busy tone The phone has timed out Note continuous busy tone does not indicate the transfer has been successful nor does it indicate the transfer has failed It often means there was a failure to receive second NOTIFY check firmware for most recent release Attended Transfer Assume that Caller A and B are in conversation Caller A wants to Attend Transfer B to C 1 Caller A presses FLASH on the analog phone for dial tone 2 Caller A then dials Caller C s number followed by or wait for 4 seconds 3 If Caller C answers the call Caller A and Caller C are in conversation Then A can hang up to complete transfer 4 If Caller C does not answer the call Caller A can press flash to resume call with Caller B NOTE When Attended Transfer fails and A hangs up the GXW will ring back user A to remind A that B is still on the call A can pick up the phone to resume conversation with B Grandstream Networks Inc GXW 400x User Manual Page 15 of 31 Firmware 1 0 0 17 Last Updated 12 2006 andstream Innovative IP Voice amp Video 3 WAY CONFERENCING Star Code Style 3 way Conference Caller A wants to bring third Caller C into conference Caller A presses FLASH to receive a dial tone Hook Flash for older model phones Caller A dials Caller C s number then or wait for 4 seconds If Caller C answer
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