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GXW400x User Manual - Grandstream Networks, Inc

1. FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 48 of 60 andstream Innovative IP Voice amp Video Dial Plan Dial Plan Rules 1 Accept Digits 1 2 3 4 5 6 7 8 9 0 A a B b C c D d 2 Grammar x any digit from 0 9 a Xx atleast 2 digits number b xx at least 2 digits number c exclude d 3 5 any digit of 3 4 or 5 e 147 any digit 1 4 or 7 f lt 2 011 gt replace digit 2 with 011 when dialing g lt 1 gt adda leading 1 to all numbers dialed vice versa will remove a 1 from the number dialed h or e Example 1 369 11 1617xxxxxxx Allow 311 611 911 and any 10 digit numbers of leading digits 1617 e Example 2 1900x lt 1617 gt Xxxxxxx Block any number of leading digits 1900 and add prefix 1617 for any dialed 7 digit numbers e Example 3 1xxx 2 9 xxxxxx lt 2 011 gt x Allow any length of number with leading digit 2 and 10 digit numbers of leading digit 1 and leading exchange number between 2 and 9 If leading digit is 2 replace leading digit 2 with 011 before dialing 3 Default Outgoing x Example of a simple dial plan used in a Home Office in the US 41900x lt 1617 gt 2 9 xxxxxx 1 2 9 xx 2 9 xxxxxx 011 2 9 x 3469 11 Explanation of example rule reading from left to right e 41900x prevents dialing any number started with 1900 e lt 1617 gt 2 9 xxxxxx allows dialing to local area code 617 numbers by dial
2. 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 It really depends on your network settings so set this parameter to Yes or No whichever works Generally if you have multiple IP devices under the same network it should be set to Yes 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 DTMF METHODS DTMF Settings are in Profile pages e DTMF in audio e DTMF via RTP RFC2833 e DTMF via SIP INFO You can enable set priority of DTMF methods according to your preference from Priority 1 to 3 This setting should be based on your server DTMF setting PREFERRED VOCODER CODEC The GXW400X supports a broad range of voice codecs Under Profile web pages choose your preferred order of different codecs e PCMU A or G711p a e G729 A B E e G723 FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 29 of 60 andstream Innovative IP Voice amp Video e G726 16 24 32 40 e iLBC e AAL2 all G726 DEFINITIONS This section will describe the options in the Web configuration user interface As mentioned a user can log in as an administrator or end user Functions availa
3. 1 0 14 1 GXW400X USER MANUAL Page 4 of 60 andstream Innovative IP Voice amp Video CHANGE LOG This section documents significant changes from previous versions of GXW400X user manuals Only major new features or major document updates are listed here Minor updates for corrections or editing are not documented here CHANGES FROM 1 0 13 3 USER MANUAL e Added feature Hold Target Before ReferHold Target Before ReferHold Target Before ReferHold Target Before ReferHold Target Before ReferHold Target Before ReferHold Target Before ReferHold Target Before ReferHold Target Before Refer in profile settings which allows user to hold or not hold the phone call before refer e Added feature Crypto Life Time in profile settings which allows user to enable or disable Crypto life time when using SRTP e Added feature Play busy reorder tone before Loop Current Disconnect in profile settings which allows user to configure if it will play busy reorder tone before loop current disconnect upon call fail CHANGES FROM 1 0 12 4 USER MANUAL e Added option SIP Timer D to configure RFC 3261 timer D in Advanced Settings e Changed option name from Allow DHCP Option 66 to override server to Allow DHCP Option 66 or 160 to override server Now option160 will be accepted by unit along with option 66 when enabled e Added feature Download Device XML Configuration in advanced settings which allows user to download device configuration to local dir
4. OPERATIONS ese es ee ee ee aana aa anaana 17 UNDERSTANDING GXW VOICE PROMPTS esse ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee Ge ER Re ee ee ee ee ee ee Re ee ee ee ee 17 PLACINGA PHONE CALL wists ccs OR N N OD N RE OE N 18 PHONE OR EXTENSION NUMDERG ese ee ee ee ese se ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee Ee ee ee ee ee ee 18 DIREC TIP CALES ES EE Senses teen ee 19 GALEHOLED EE 20 GAL WALING oa ee 20 GALLTRANSEER it ee Eege 20 BLIND rd ds AE EE N OE N RE OE EERE 20 nia eland EE N EE leds 20 3 WAY CONFERENCING E 21 al Elle iese dIE N EE EE N N 21 INTER PORT GANG eeng N N RE N EE N 23 PSTN PASS THROUGB LIFE UNE iste s esse ee EER GEE EERS SN Rg be GEE EE GED NRS Ee BERE EE Gee Re ESE EE EE Gee ee RE EE SEE Ee 24 FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 1 of 60 andstream Innovative IP Voice amp Video SENDING AND RECEIVING FAX ed osse eer eed ese Feeds een Gees retenant ratent eg 24 SUPPORT RADIUS PROTOCOL OE EE EE 24 CALL FEATURES sisissossiasssstsnsssssssnaccacisnsestsassiccnasnascanasuncsassssnsisassasaceuassses 25 CONFIGURATION GUIDE ass ss ss sies ds ge N GEAG EG NR SKA GN NR GN Ee ee 27 CONFIGURING GXW400X VIA VOICE PROMPT EEN 27 CONFIGURING GXW400X WITH WEB BROWSER 27 ACCESS THE WEB CONFIGURATION MENU tegen Eeer eene 28 IMPORTANT SE TINGS ee 29 NEEN 29 HERE 29 e 29 SAVING THE CONFIGURATION CHANGES eeneg 53 REBOOTING FROM REMOT RE EE N NE EN 53 CONFIGURATION
5. Postfix allows device to download the configuration file with the matching Prefix and Postfix Thus multiple configuration files for the same device can be stored in one directory In addition when the field Check New Firmware only when F W pre suffix changes is selected the device will only issue firmware upgrade request if there are changes in the firmware Prefix or Postfix MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD When Automatic Upgrade is set Yes every the auto check will be done in the minute specified in this field If set to daily at hour 0 23 Service Provider can use P193 Auto Check Interval to have the devices do a daily check at the hour set in this field with either Firmware Server or Config Server If set to weekly on day 0 6 the auto check will be done in the day specified in this field This allows the device periodically check if there are any new changes need to be taken on a scheduled time By defining different intervals in 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 Automatic Upgrade D 10080 l 2 No Yes every minutes 60 5256000 1 1 Yes daily at hour 0 23 C Yes weekly on day 0 6 FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 57 of 60 andstream Innovative IP Voice amp Video This page is intentionally left
6. bank FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 58 of 60 andstream Innovative IP Voice amp Video 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 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 VolP service provider FACTORY RESET There are two 2 methods for resetting your unit RESET BUTTON Reset default factory settings following 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 3 Insert a pin in this hole and press for about 7 seconds 4 Take out the pin All unit settings are restored to factory settings IVR COMMAND Reset default factory settings using the IVR Prompt Table 5 1 Dial for voice prompt Enter 99 and wait for reset voice prompt Enter the encoded MAC address Look below on how to encode MAC address BP oO DP Wait 15 seconds and device will automatically reboot and restore factory settings Encode the MAC Address 1 Locate the MAC address of the device It is the 12 digit HEX number on the bottom of the unit 2 Key in the MAC address Use the following mapping FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Pag
7. best use of its many upgraded features including simple and quick installation multi party 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 VolP solution for the small medium business or enterprise GATEWAY GXW400X OVERVIEW The new GXW400X 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 GXW400X 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 for administration In addition it supports the option of 3 SIP Server profiles caller ID for various countries regions T 38 fax flexible dialing plans security protection SIPS TLS comprehensive voice codec including G 711 a u law G 723 1 G 726 16 24 32 48 bit rates G 729A B E and iLBC SAFETY COMPLIANCE The GXW400X is compliant with various safety standards including FCC CE Its power adapter is compliant with UL standard Warning use only the power adapter included in the GXW400X package Use of alternative power adapter may permanently damage the unit FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 7 of 60 andstream WARRANTY Grandstream has a reseller agreement w
8. device or Enter MAC address to restore factory default setting See Restore Factory Default Setting section Invalid Entry Automatically returns to Main Menu Five Success Tips when using the Voice Prompt 1 BEI 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 cannot be deleted but the phone may prompt error once it is detected PLACING A PHONE CALL PHONE OR EXTENSION NUMBERS 1 or 2 Dial the number directly and wait for 4 seconds To change the default value modify the following setting No Key Entry Timeout Dial the number directly and press Use as dial key must be configured in web configuration FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 18 of 60 andstream Innovative IP Voice amp Video Examples 1 Dial a number e g 626 666 7890 first enter the prefix number usually 1 or international code followed by the phone number Press or wait for 4 seconds Check with your VolP service provider for further details on prefix numbers DIRECT IP CALLS Direct IP calling allows two parties that is a FXS Port
9. 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 Default is 2 from 1 4 for G711 G726 G729 Defines the encoding rate for G 723 vocoder By default 6 3kbps rate is chosen Sets the iLBC frame size in 20ms or 30ms Default value is 97 Defines payload type for iLBC The valid range is between 96 and 127 Default value is 100 Range is from 96 to 127 Default value is 99 Range is from 96 to 127 Default value is 104 Range is from 96 to 127 Default value is 103 Range is from 96 to 127 Default value is 102 Range is from 96 to 127 FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 51 of 60 VAD Symmetric RTP Fax Mode Fax Tone Detection Mode Jitter Buffer Type Jitter Buffer Length SRTP Mode Crypto Life Time SLIC Setting Caller ID Scheme Polarity Reversal Loop Current Disconnect Play busy reorder tone before Loop Current Disconnect andstream Innovative IP Voice amp Video Default is No VAD allows detecting the absence of audio and conserve bandwidth by preventing the transmission of silent packets over the network Default is No When set to Yes the device will change the destination to send RTP packets to the source IP address and port of the inbound RTP packet last received by the device T 38 Auto Detect FolP by default or Pa
10. the unique MAC address of each device GAPS provision the devices with redirection settings so that they will be redirected to customer s TFTP or HTTP HTTPS server for further provisioning Grandstream also provides configuration tools Windows and Linux Unix version to facilitate the task of generating device configuration files The Grandstream configuration tools are free to end users The configuration tools and configuration templates are available for download from http www grandstream com support tools FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 54 of 60 andstream Innovative IP Voice amp Video SOFTWARE UPGRADE Software upgrade can be done via either TFTP or HTTP HTTPS The corresponding configuration settings are in the ADVANCED SETTINGS configuration page FIRMWARE UPGRADE THROUGH TFTP HTTP HTTPS To upgrade via TFTP or HTTP HTTPS the Firmware Upgrade and Provisioning upgrade via field needs to be set to TFTP HTTP or HTTPS 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 e g firmware mycompany com 6688 Grandstream 1 0 9 1 e g firmware grandstream com NOTES Firmware upgrade server in IP address format can be configured via IVR Please refer to the CONFIGURATION GUIDE section for instructions If the server is in FQDN format it must be set via
11. with an analog phone and another VolP Device to talk to each other in an ad hoc fashion without a SIP proxy Elements necessary to completing a Direct IP Call 1 Both GXW400X and other VolP Device have public IP addresses or 2 Both GXW400X and other VoIP Device are on the same LAN using private IP addresses or 3 Both GXW400X and other VoIP Device can be connected through a router using public or private IP addresses with necessary port forwarding or DMZ GXW400X supports two ways to make Direct IP Calling Using IVR 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 the IP address using format ex 192 168 0 160 after the dial tone Using Star Code 1 Pick up the analog phone then dial 47 2 Enter the target IP address using same format as above Note NO dial tone will be played between step 1 and 2 Destination ports can be specified by using encoding for followed by the port number Examples a If the target IP address is 192 168 0 160 the dialing convention is 47 or Voice Prompt with option 47 then 192 168 0 160 Followed by pressing the key if it is configured as a send key or wait 4 seconds In this case the default destination port 5060 is used if no port is specified FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 19 of 60 andstream Innovative IP Voice amp Video b If
12. 00X package EQUIPMENT PACKAGING Unpack and check all accessories Equipment includes e one device unit e one RJ45 Ethernet cable e one 12V universal power adapter CONNECT THE GXW400X Follow these four 4 steps to connect your GXW400X gateway to the Internet and access the unit s configuration pages 1 Connect standard touch tone analog phones to the FXS ports 2 Insert an RJ45 Ethernet cable into the WAN port of GXW400X and connect the other end to an uplink port a router or a modem etc Note If the gateway is used in router mode connect a PC to the LAN port of GXW400X for initial configuration 3 Plug the power adapter into the GXW400X gateway into a power outlet Follow the instructions from the topic Configuring GXW 400X with Web Browser for initial configuration 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 products gxw_series gxw400x documents gxw400x_gui zip FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 9 of 60 tream Innovative IP Voice amp Video GXW 400x Reset Power Console SE FXS LAN WAN Supply Connection Ports RJ 45 Ethernet Ports connection FIGURE 1 DIAGRAM OF GXW4004 8 BACK PANEL TABLE 1 DEFINITIONS OF THE GXW CONNECTORS LAN Connect the LAN port with an Ethernet cable to your PC WAN Connect to the internal LAN network or router RESET Factory Re
13. 6 SIP UserID and Authenticate ID left blank Hunting Group set to 4 FXS Port 7 SIP UserID and Authenticate ID entered Hunting group set to Active FXS Port 8 SIP UserlD and Authenticate ID left blank Hunting Group set to 7 Hunting Group 1 contains ports 1 2 3 Hunting Group 4 contains ports 4 5 6 Hunting Group 7 contains ports 7 8 Request URI Routing ID If configured device will route the incoming call to designated port by request URI user ID in SIP INVITE Enable Port If set to No FXS port will become inactive Default is set to Yes Port FXS Port Number FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 42 of 60 Offhook Auto dial Offhook Auto Dial Delay Map to FXO Port Map to FXO Gateway IP and Port Profile Active Primary SIP Server Failover SIP Server Prefer Primary SIP Server Outbound Proxy SIP transport andstream Innovative IP Voice amp Video This feature allows you to automatically dial the number specified in this field as soon as the port is offhooked i e when the receiver on the phone connected to Port is picked up Configure the delay time for offhook auto dial function Range is 0 60 seconds default is 0 This is used only when peering with a Grandstream GXW410x Default is 1 Supported values are 1 8 meaning line 1 to line 8 of the GXW410x device where the port will be mapped to This is used when peering with an FXO gateway of any brand You have to spec
14. ACS Password Periodic Inform Enable Periodic Inform Interval Connection Request Username Connection Request Password Connection Request Port CPE SSL Certificate CPE SSL Private Key andstream Innovative IP Voice amp Video Choose Yes to enable automatic upgrade and provisioning When set to No GXW400X will only do upgrade once at boot up When Check every day or Check every week is checked user can specify Hour of the day 0 23 or Day of the week 0 6 Default time is Monday 1AM There are three options to choose from Always check for New Firmware at Boot up Check New Firmware only when F W pre suffix changes and Always Skip the Firmware Check Device will not challenge NOTIFY with 401 when set to Yes If set to Yes configuration file is authenticated before being accepted This protects the configuration from unauthorized modifications For firmware encryption It should be 32 digit in Hexadecimal Representation End user should keep it blank The GXW400X series supports SIP over TLS It has built in private key and SSL certificate The user specified SSL certificate used for SIP over TLS is in X 509 format You may also customize the SSL Private Key The user specified SSL private key used for SIP over TLS is in X 509 format Enter SSL Private Key password here User specify the Auto Configuration Server s URL TR 069 protocol User specify the ACS Us
15. FTWARE FEATURES GXW4004 GXW4008 Telephone Interfaces 4 FXS ports 8 FXS ports SIP Provisioning 4 SIP accounts 2 profiles 8 SIP accounts 2 profiles Network Interface Two 2 10M 100 Mbps RJ 45 Two 2 10M 100 Mbps RJ 45 Number of Concurrent Calls 4 Concurrent Calls 8 Concurrent Calls except when using SRTP Voice over Packet Voice Activity Detection VAD with Voice Activity Detection VAD with CNG Capabilities CNG comfort noise generation and comfort noise generation and PLC FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 13 of 60 PSTN Fail over Voice Compression DHCP Server Client Fax over IP QoS Transport Protocol DTMF Method IP Signaling Provisioning Control Management Dial Plan PLC packet loss concealment AEC with NLP Packetized Voice Protocol Unit supports RTP and AAL2 protocol G168 compliant Echo Cancellation Dynamic Jitter Buffer Modem detection amp auto switch to G 711 PSTN failover port on power failure G 711 Annex PLC Annex Il VAD CNG format encoder and decoder G723 1A G726 ADPCM with 16 24 32 40 bit G 729A B E iLBC G 726 provides proprietary VAD CNG and signal power estimation Voice rates Play Out unit reordering fixed and buffer synchronization AGC automatic gain adaptive jitter clock control Status output Decoder controlling via voice packet header Yes NAT Router or Switched Mode T 38 compliant Group 3 Fax Relay up to 14 4
16. I format then this option needs to be selected If set to Yes device will use Request URI Routing ID defined in FXS ports settings to replace From and Contact headers for outgoing calls This parameter controls whether the GXW400X 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 FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 44 of 60 Outgoing Call without Registration Register Expiration Reregister before Expiration Local SIP port Local RTP Port Use random port Hold Target Before Refer Refer to Use Target Contact Transfer on Conference Hang up Disable Bellcore Style 3 Way Conference Remove OBP from Route Header andstream Innovative IP Voice amp Video Default is No If set to Yes user can place outgoing calls even when not registered if allowed by Internet Telephone Service Provider but is unable to receive incoming calls Any port member of a Hunting Group that is not registered with a SIP account will be able to place outbound calls using the SIP credentials of the primary Hunting Group port For example Port 1 3 and 5 are members of the same Hunting Group Port 1 is registered with a SIP account Ports 3 and 5 are not registered Ports 3 and 5 will be able to place outbound calls using the SIP account of port 1 even if Outgoing Call wit
17. ING GXW400X WITH WEB BROWSER The GXW400X series gateway has an embedded Web server that allows users to configure the GXW400X through a web browser FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 27 of 60 andstream Innovative IP Voice amp Video ACCESS THE WEB CONFIGURATION MENU The GXW400X HTML configuration menu can be accessed via LAN or WAN port From the LAN port For GXW4004 8 only 1 Directly connect a computer to the LAN port 2 In Windows open a Command Prompt window 3 Type in ipconfig release the IP address etc becomes 0 4 Type in ipconfig renew the computer gets an IP address in 192 168 2 x segment by default 5 Open a web browser type in the default gateway IP address http 192 168 2 1 You will see the login page of the device From the WAN port For GXW4004 8 only The WAN port HTML configuration option is disabled by default from factory To access the HTML configuration menu from the WAN port 1 Follow table 4 to find the WAN side IP address 2 Open a web browser type in the WAN side IP address for example http GXW400X_ IP Address the GXW400X IP Address is the WAN IP address for the GXW400X NOTE e WAN side HTTP access is disabled by default for security reasons You can enable HTTP access on the configuration page by setting WAN side HTTP access to be YES e Initial access to the configuration pages is always from the LAN port The instructions are li
18. RE VERSION 1 0 14 1 GXW400X USER MANUAL Page 55 of 60 andstream Innovative IP Voice amp Video e Grandstream s latest firmware is available http www grandstream com support firmware Oversea users are strongly recommended to download the binary files and upgrade firmware locally in a controlled LAN environment e 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 INSTRUCTIONS FOR LOCAL FIRMWARE UPGRADE 1 Unzip the file and put all of them under the root directory of the TFTP server 2 Putthe PC running the TFTP server and the GXW400X device in the same LAN segment 3 TFTP server s security settings should be changed from Receive Only to Transmit Only for the firmware upgrade 4 Configure the Firmware Server Path with the IP address of the PC 5 Update the change and reboot the unit CONFIGURATION FILE DOWNLOAD Grandstream SIP Device can be configured via Web Interface as well as via Configuration File through TFTP or HTTP HTTPS Config Server Path is the TFTP or HTTP HTTPS 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 S
19. THROUGH A CENTRAL SEAVER ei siese sies getrennt 53 GAR a ET 55 FIRMWARE UPGRADE THROUGH TETP HTTP HTTPS see sesse see ees ee esse seke ees Re ee 55 INSTRUCTIONS FOR LOCAL FIRMWARE UPGRADE us see esels gese sees Gelee Spes een beee 56 CONFIGURATION FILE DOWNLOAD RE EE erstarrt enste rnar rater nert 56 FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX esse sesse ees ee 57 MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD 57 RESTORE FACTORY DEFAULT SETTING ees ee ee ee esse se se se ee ee ee ee ee ee ee EE 59 FACTORY ESE KS SG ee n Ge ei se ER or De oe Re ei 59 RESET BUTTON AN OE EE 59 CRS OI NII IE ET EE IE EI 59 FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 2 of 60 andstream Innovative IP Voice amp Video TABLE OF FIGURES GXW400X User Manual FIGURE 1 DIAGRAM OF GXW4004 8 BACK PANEL A 10 FIGURE 2 DIAGRAM OF GXW4004 8 DISPLAY PANEL AAA 11 TABLE OF TABLES GXW400X User Manual TABLE 1 DEFINITIONS OF THE GXW CONNECTORS ees see ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee eek ee 10 TABLE 2 DEFINITIONS OF THE GXW DISPLAY PANEIL esse see see ee ee ee ee se ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee 11 TABLE 3 GXW400X SOFTWARE FEATURES ee ese ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ke eek ee 13 TABLE A HARDWARE SPECIFICATION OF GXW400X SERIES GATEWAYS A 15 TABLE 5 DEFINITIONS OF THE GXW VOICE PROMPTS ee ee see se ee ee se ee ee ee ee ee ee ee ee
20. VERSION 1 0 14 1 GXW400X USER MANUAL Page 15 of 60 andstream Innovative IP Voice amp Video This page is intentionally left bank FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 16 of 60 andstream Innovative IP Voice amp Video BASIC OPERATIONS UNDERSTANDING GXW VOICE PROMPTS GXW400X has a built in voice prompt menu for simple device configuration To enter the voice prompt menu press on the standard analog phone connected to any FXS port Main Menu 01 02 03 04 05 07 10 12 13 14 TABLE 5 DEFINITIONS OF THE GXW VOICE PROMPTS Menu Voice Prompt User s Options Enter a Menu Option DHCP Mode PPPoE 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 Enter Enter Enter Enter for the next menu option to return to the main menu 01 05 07 10 17 47 86 or 99 Menu option 9 to toggle the selection If user selects Static IP Mode user need configure all the IP address information through menu 02 to 05 If user selects Dynamic IP Mode the device will retrieve all IP address information from DHCP server automatically when user reboots the device The current WAN IP address is announced En
21. Wink EMC Yes Bellcore Type 1 amp 2 ETSI BT NTT and DTMF based CID Yes EN55022 EN55024 and FCC parti5 Class B HARDWARE SPECIFICATION The hardware specifications of the GXW FXS series are detailed in Table 4 andstream Innovative IP Voice amp Video Yes Bellcore Type 1 amp 2 ETSI BT NTT and DTMF based CID Yes EN55022 EN55024 and FCC parti5 Class B TABLE 4 HARDWARE SPECIFICATION OF GXW400X SERIES GATEWAYS Ports Network interface PSTN Port Console Universal Switching Power Adaptor Short Haul Loop Mounting Dimension Weight Temperature Humidity Safety Compliance GXW4004 4 FXS Ports 1 Rap Full Duplex for LAN 10 100 Base TX 1 RJ45 Full Duplex for WAN 10 100Base TX PSTN fail over port DB9 Serial port Input 100 240V AC 50 60Hz 0 3A Max Output 12V DC 1 25A UL certified 2REN Up to 1km on 24 AWG wire Wall mount Desktop 225mm L x 135mm W x 35mm H 0 29 Ibs 3 5 oz 32 104 F 0 40 C 10 90 non condensing UL FCC CE GXW4008 8 FXS Ports 1 RJ45 for LAN 10 100 Base TX Full Duplex 1 RJ45 for WAN 10 100Base TX Full Duplex PSTN fail over port DB9 Serial port Input 100 240V AC 50 60Hz 0 3A Max Output 12V DC 1 25A UL certified 2REN Up to 1km on 24 AWG wire Wall mount Desktop 225mm L x 135mm W x 35mm H 0 29 Ibs 3 5 oz 32 104 F 0 40 C 10 90 non condensing UL FCC CE FIRMWARE
22. able SRTP Block CallerlD for all config change Send CallerlD for all config change Block Caller per call Send CallerlD per call Direct IP Calling Dial 47 IP address No dial tone will be played in the middle Detail see Direct IP Calling section on page 12 Disable Call Waiting for all config change Enable Call Waiting for all config change Call Return Service Dial 69 and the phone will dial the last incoming phone number received Disable Call Waiting Per Call Enable Call Waiting Per Call Unconditional Call Forward Dial 72 and then the forwarding number followed by Wait for dial tone and hang up dial tone indicates successful forward Cancel Unconditional Call Forward Dial 73 and wait for dial tone then hang up Enable Paging Call Dial 74 and then the destination phone number you want to activate in Paging mode Enable Do Not Disturb DND When enabled all incoming calls will be rejected Disable Do Not Disturb DND When disabled incoming calls will be accepted FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 25 of 60 andstream Innovative IP Voice amp Video 87 Blind Transfer 90 Busy Call Forward Dial 90 and then the forwarding number followed by 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 and then the forwardin
23. ain phones GXW400X has this feature enabled by default However certain phones rare that do not support it may mistakenly treat this CID signal as an incoming call A configuration option is needed to turn on MWI in this case Do Not Escape If set to Yes device will use instead of 23 in the send URI as 23 in SIP URI Disable Multiple m line Default is No If set to Yes device will send only one m line in SDP regardless how in SDP many m field in the incoming SDP Ring Timeout Incoming call will stop ringing when not picked up given a specific period of time FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 47 of 60 Hunting Group Ring Timeout Hunting Group Type Delayed Call Forward Wait Timeout No Key Entry Timeout Early Dial Dial Plan Prefix Use as Dial Key andstream Innovative IP Voice amp Video Default is 20 seconds If call is not answered within this designated time period the call will be forwarded to the next member of a Hunt Group Linear and Circular Linear style will sort the call to the lowest numbered available line this is also called serial hunting Circular style will distribute the calls round robin If a call is assigned to line 1 the next call goes to 2 and the next to 3 The succession throughout each of the lines continues even if one of the previous lines becomes available When the end of the hunt group is reached the hunting starts over at the first
24. automatically configured from a central provisioning system When GXW400X boots up it will send TFTP or HTTP HTTPS requests to download configuration files cfg000b82xxxxxx and cfg00082xxxxxx xml where 000b82xxxxxx is the LAN MAC address of the GXW400X If the download of cfgxxxxxxxxxxxx xml is not successful the provision program will issue FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 53 of 60 andstream Innovative IP Voice amp Video request a generic configuration file cfg xmI Configuration file name should be in lower case letters The configuration data can be downloaded via TFTP or HTTP HTTPS from the central server A service provider or an enterprise with large deployment of GXW400X can easily manage the configuration and service provisioning of individual devices remotely from a central server Grandstream provides a central provisioning system GAPS Grandstream Automated Provisioning System to support automated configuration of Grandstream devices GAPS uses enhanced NAT friendly TFTP or HTTP thus no NAT issues and other communication protocols to communicate with each individual Grandstream device for firmware upgrade remote reboot etc Grandstream provides GAPS service to VoIP service providers Use GAPS for either simple redirection or with certain special provisioning settings At boot up Grandstream devices by default point to Grandstream provisioning server GAPS based on
25. ber of times the device will try to authenticate itself with preconfigured RADIUS server during initialization process This setting allows user to download a text file containing all the P values of each setting as configured on the unit Note For Security Reasons any Password will not be Downloaded Allows user to download and save a XML file containing all the P values of each setting as configured at that point on the unit Note For Security Reasons all Passwords won t be Downloaded Allows user to upload firmware file from local directory Allows user to upload configuration file from local directory TABLE 10 FXS PORTS SETTINGS FXS Port Number User account information provided by VolP service provider ITSP Usually in the form of digit similar to phone number or actually a phone number FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 40 of 60 An Innovative IP Voice amp Video 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 GXW400X to register to SIP servers of ITSP Name Any name to identify this specific user Profile ID Select the corresponding Profile ID 1 2 FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 41 of 60 andstream Innovative IP Voice amp Video Hunting Group This feature enables the gateway to register all existing FXS por
26. ble for the end user are e STATUS Displays the network status account status software version and MAC address of the phone e BASIC SETTINGS Basic preferences such as date and time settings multi purpose keys and LCD settings can be set here Additional functions available to administrators are e ADVANCED SETTINGS To set advanced network settings codec settings and XML configuration settings e PROFILE X To configure each of the SIP accounts e FXS PORTS To configure each of the FXS ports and Hunting Groups etc End User Password Web Port Telnet Server IP Address TABLE 7 BASIC SETTINGS 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 Default is Yes There are two modes to operate the GXW400X DHCP mode The GXW400X acquires its IP address from the first DHCP server it discovers from the LAN It is on PPPoE Set the PPPoE account settings The GXW400X 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 FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 30 of 60 DHCP hostname DHCP vendor class ID PPPoE account ID PPPoE password PPPoE Service Name Time Zone S
27. den Innovative IP Voice amp Video Grandstream Networks Inc Analog IP Gateway GXW400X 4 or 8 FXS Ports GXW400X USER MANUAL This page is intentionally left bank andstream Innovative IP Voice amp Video GXW400X USER MANUAL INDEX GNU GPL INFORMATION ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee 4 CHANGE aaa ei n Ee ee ee oe 5 CHANGES FROM 1 0 13 3 USER MANUALE 5 CHANGES FROM 1 0 12 4 USER MANUALE 5 CHANGES FROM 1 0 12 1 USER MANUALE 5 CHANGES FROM 1 0 11 3 USER MANUALE 5 CHANGES FROM 1 0 10 9 USER MANUALE 6 CHANGES FROM 1 0 7 6 USER MANUALE 6 CHANGES FROM 1 0 6 8 USER MANUALE 6 WELCOME ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee 7 GATEWAY GXW400X OVERVIEW ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee Ee ee ee ee ee ee Ee Re Re ee ee ee ee ee ee 7 SAFETY KENE Ee OR A GO EG Ge EE Oe ORE ee ses 7 Mtb NOR EE RR EE RE RE EE 8 CONNECT YOUR GXW400X GATEWAY EE 9 EQUIPMENT PACKAGING ee ee ee ee RR RR ee ee ee ee 9 CONNECT THE GXWAOOX ee ee ee ee ee ee se se ee ee ee ee ee ee ee Ee ee Re ee ee ee Ee Re Re Ee ee EE Ee ee ee AA Ee Ee Re ee ee AE ee Re ee 9 GXWAOOX FEATURES ENEE 13 SOFTWARE FEATURES OVERVIEW e ese ee esse ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee Ee ee ee ee 13 HARDWARE GPECIEICATION ee ese ee ee ee ees ee ee ee ee ee ee ee ee ee ee ee ee ee ee Ee ee ee ee ee Ge ee ee ee ee ee ee Re ee ee ee ee 15 BASIC
28. e 59 of 60 andstream Innovative IP Voice amp Video 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 333 3333 mI op For example if the MAC address is 000b8200e395 it should be keyed in as 0002228200333395 NOTE 1 Factory Reset will be disabled if the Lock keypad update is set to Yes 2 Please be aware by default the GXW400X WAN side HTTP access is disabled After a factory reset the device s web configuration page can be accessed only from its LAN port FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 60 of 60
29. ectory in xml format e Added feature Upload firmware in advanced settings which allows user to upload firmware file from local directory e Added feature Upload configuration in advanced settings which allows user to upload configuration file from local directory CHANGES FROM 1 0 12 1 USER MANUAL e Added the options to enable disable Always send HTTP Basic Authentication Information e Added the options to restrict the m field sent in SDP Disable Multiple m line in SDPDisable Multiple m line in SDP CHANGES FROM 1 0 11 3 USER MANUAL FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 5 of 60 andstream Innovative IP Voice amp Video e Added the options to enable disable Do Not Escape as 23 in SIP URI e Added network whist black list function on WAN port White list for WAN side Black list for WAN side CHANGES FROM 1 0 10 9 USER MANUAL e Added the options to enable disable Use P Preferred Identity Header and Use Privacy Header e Added the option to enable disable Add Auth Header On Initial REGISTER CHANGES FROM 1 0 7 6 USER MANUAL e Added option to enable disable SIP NOTIFY Authentication Disable SIP NOTIFY Authentication e Added option Use Configured IP in DNS mode Added configurable parameter Primary IP Backup IP1 Backup IP2 e Added option to set Reregister before Expiration e Added option Use Request Routing ID in SIP INVITE Header to allow user to replace From and Contact headers
30. ee ee ee ee ee ee ee ee ee ee 17 TABLE 6 CALL FEATURES TABLE STAR COD I ees ee ee ee ee een ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee de ee ee ee ke eek ee 25 TABLE 7 ee oo RE EE une 30 TABLE 8 STATUS EE 33 TABLE 9 ADVANCED SET TINGS tee susse seg seed eu iaeia ais aeia a ees ee Be eeu Gegee ek dek Eege 35 TABLE IO EXS PORTS IER Hl TEE A0 TABLE 1 PROFILE SETTINGS see osse giee ss gees ges ke Dee eed Eeue sege bees Dee be ge Ge Ge Ge eeue ee De ee ie gees De 43 CONFIGURATION GUI INTERFACE EXAMPLES GXW400X User Manual http www grandstream com products gxw_series gxw400x documents gxw400x_qui zip 1 SCREENSHOT OF ADVANCED SETTINGS CONFIGURATION PAGE SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE SCREENSHOT OF FXS PORTS CONFIGURATION LOGIN PAGE SCREENSHOT OF PROFILE 1 CONFIGURATION PAGE SCREENSHOT OF PROFILE 2 CONFIGURATION PAGE o a fF oO LV SCREENSHOT OF STATUS PAGE FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 3 of 60 andstream Innovative IP Voice amp Video GNU GPL INFORMATION GXWAOOX firmware contains third party software licensed under the GNU General Public License GPL Grandstream uses software under the specific terms of the GPL Please see the GNU General Public License GPL for the exact terms and conditions of the license Grandstream GNU GPL related source code can be downloaded from Grandstream web site from hittp www grandstream com support faq gnu_gpl FIRMWARE VERSION
31. elf Defined Time Zone Language Device Mode NAT maximum ports NAT TCP timeout NAT UDP timeout Uplink Bandwidth andstream Innovative IP Voice amp Video This option specifies the name of the client This field is optional but may be required by some Internet Service Providers Default is blank Used by clients and servers to exchange vendor specific information Default is HT500 PPPoE username Necessary if ISP requires you to use a PPPoE Point to Point Protocol over Ethernet connection PPPoE account password This field is optional If your ISP uses a service name for the PPPoE connection enter the service name here Default is blank Controls how the date time is displayed according to the specified time zone The syntax is std offset dst offset start time end time Default is set to MTZ 6MDT 5 M3 2 0 M11 1 0 MTZ 6MDT 5 It indicates a time zone with 6 hours offset with 1 hour ahead which is the US central time It is positive if the local time zone is west of the Prime Meridian and negative if itis east Prime Meridian nternational or Greenwich Meridian M3 2 0 M11 1 0 The 1st number indicates Month 1 2 3 12 for Jan Feb Dec The 2nd number indicates the nth iteration of the weekday 1st Sunday 3rd Tuesday etc The 3rd number indicates Weekday 0 1 2 6 for Sun Mon Tue Sat Therefore this example is the DST which starts from the second Sunday of March to t
32. ername User specify the ACS password Default is No If set to YES device will send inform packets to the ACS Frequency that the inform packets will be sent out to the ACS Set a user name for the ACS to connect to this device Set a password for the ACS to connect to this device Set a port number for the ACS to connect to this device default is 7547 Configure the SSL authentication of Customer premises equipment Configure the SSL Private Key of Customer premises equipment FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 37 of 60 System Ring Cadence Call Progress Tones Prompt Tone Access Code Lock Keypad Update Disable Voice Prompt Disable Direct IP Call Lifeline Mode andstream Innovative IP Voice amp Video Configuration option for all FXS ports ring cadence for all incoming calls Syntax c on1 off1 on2 off2 on3 off3 Default is set to c 2000 4000 US standards Using these settings user can configure tone frequencies according to user preference By default the tones are set to North American frequencies Frequencies should be configured with known values to avoid uncomfortable high pitch sounds ONis the period of ringing ON time in ms while OFF is the period of silence In order to set a continuous ring OFF should be zero Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern e Dial tone e Ringback tone e Busy Re order tone e Confirmation
33. erver 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 in the future digit numeric numbers i 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 a Grandstream device boots up or reboots it will issue a request for a configuration file C JXXXXXXXXXXXX where XXXXXXXXXXXX is the MAC address of the device Le cfg000b820102ab In addition device will also requests a XML configuration file cfgxxxxxxxxxxxx xml If the download of CEQXXXXXXXXXXXX xMI is not successful the provision program will issue a request for a generic configuration file cfg xml Configuration file name should be in lower case letters FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 56 of 60 andstream Innovative IP Voice amp Video For more details on Grandstream Device provisioning please refer to http www grandstream com general gs provisioning guide public pdf FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX Firmware Prefix and Postfix allows device to download the firmware name with the matching Prefix and Postfix This makes it possible to store ALL of the firmwares with different version in one single directory Similarly Config File Prefix and
34. et access If black list exists and white list is empty then ONLY these IP addresses are NOT ALLOWED to web and telnet access Case 2 WAN side telnet HTTP access is not allowed All addresses are NOT ALLOWED http and telnet access SIP RTP If both white list and black list are empty the behaviour is same as before If white list exists then ONLY these IP addresses are ALLOWED SIP RTP access If black list exists and white list is empty then ONLY these IP addresses are NOT ALLOWED SIP RTP access List the IP address or IP range in the White list The same rules as white list This allows you to change set the MAC address on the WAN interface Default is Yes 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 When the device detects WAN IP is conflicting with LAN IP the LAN base IP address will be changed based on the network mask the effective subnet will be increased by 1 For example 192 168 2 1 will be changed to 192 168 3 1 if net mask is 255 255 255 0 Then the device will reboot FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 32 of 60 LAN DHCP Start IP LAN DHCP End IP DHCP IP Lease Time DMZ IP Port Forwarding andstream Innovative IP Voice amp Video Default is 100 Default is 199 Value is set in units of hours Default value is 120 hrs 5 Days The time IP address is ass
35. evice has not registered with any SIP Service provider FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 11 of 60 andstream Innovative IP Voice amp Video This page is intentionally left bank FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 12 of 60 andstream Innovative IP Voice amp Video GXW400X FEATURES The GXW4000 series is a next generation IP voice gateway that is interoperable and compatible with leading IP PBXs Softswitches and SIP platforms The GXW4000 series FXS gateway is auto configurable remotely manageable and scalable The GXW4000 series gateways come in two models the GXW 4004 and GXW 4008 each offering superb voice quality traditional telephony functionality easy deployment and 4 and 8 FXS ports respectively Each model features flexible dialing plans integrated call routing to support a pure IP network call and an external power supply SOFTWARE FEATURES OVERVIEW e 4or8FXS ports e Two RJ 45 ports switched or routed e Multiple SIP accounts amp profiles 4 8 configurable through 2 profiles e Supports Voice Codecs G711 a u Annex amp Il G723 1A G726 ADPCM with 16 24 32 40 bit rates G729 A B E iLBC e T 38 Fax e Comprehensive Dial Plan support for Outgoing calls e G168 Echo Cancellation e Voice Activation Detection VAD Comfort Noise Generation CNG and Packet Loss Concealment PLC e Supports PSTN PBX analog telephone sets or analog trunks TABLE 3 GXW400X SO
36. figure the SIP account from your VoIP Service Provider on FXS port 1 under FXS Ports webpage 2 Select Active under the Hunting Group drop box for FXS port 1 3 For the remaining ports say 2 3 and 4 select 1 for Hunting Group Ports 2 3 and 4 are now active members of the hunting group associated with port 1 This configuration will route all calls directed to FXS port 1 to ports 2 3 and or 4 in round robin fashion respectively f port 1 is busy or times out You can configure the ring timeout on the Profile page Example configuration of a multiple Hunting Group FXS Port 1 SIP UserlD and Authenticate ID entered Hunting group set to Active FXS Port 2 SIP UserlD and Authenticate ID left blank Hunting Group set to 1 FXS Port 3 SIP UserID and Authenticate ID left blank Hunting Group set to 1 FXS Port 4 SIP UserlD and Authenticate ID entered Hunting group set to Active FXS Port 5 SIP UserlD and Authenticate ID left blank Hunting Group set to 4 FXS Port 6 SIP UserlD and Authenticate ID left blank Hunting Group set to 4 FXS Port 7 SIP UserlD and Authenticate ID entered Hunting group set to Active FXS Port 8 SIP UserlD and Authenticate ID left blank Hunting Group set to 7 Hunting Group 1 contains ports 1 2 3 Hunting Group 4 contains ports 4 5 6 Hunting Group 7 contains ports 7 8 Please be aware the choice of 1 for ports 2 and 3 the choice of 4 for ports 5 and 6 the choice 7 for port 8 is requ
37. for outgoing calls by Request URI Routing ID e Added field Request URI Routing ID to allow device to route the calls to individual fxs ports based on the Request URI user ID in the incoming INVITE CHANGES FROM 1 0 6 8 USER MANUAL e Added the option to enable disable hook flash Enable Hook Flash e Added two CPE SSL configuration CPE SSL Certificate CPE SSL Private Key e Added a configuration parameter to set the Connection Request Port e Removed DHCP Domain from Web UI e Removed Enable Ring transfer from Web UI e Added a configuration parameter to set the Offhook Auto Dial Delay e Changed the SSL Web UI description to SIP TLS Certificate SIP TLS Private Key and SIP TLS Private Key Password FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 6 of 60 andstream Innovative IP Voice amp Video WELCOME Thank you for purchasing the Grandstream GXW400X Analog FXS IP Gateway The GXW400X offers an easy to manage easy to configure IP communications solution for any business with virtual and or branch locations The GXW400X supports popular voice codecs and is designed for full SIP compatibility and interoperability with third party SIP providers thus enabling you to fully leverage the benefits of VolP technology integrate a traditional phone system into a VolP network and efficiently manage communication costs This manual will help you learn how to operate and manage your GXW FXS Analog IP Gateway and make the
38. g number followed by 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 Pressing pound sign will serve as Re Dial key FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 26 of 60 andstream Innovative IP Voice amp Video CONFIGURATION GUIDE CONFIGURING GXW400X VIA VOICE PROMPT DHCP MODE Select voice menu option 01 to enable GXW400X to use DHCP STATIC IP MODE Select voice menu option 01 to enable GXW400X to use STATIC IP mode then use option 02 03 04 05 to set up IP address Subnet Mask Gateway and DNS server respectively FIRMWARE SERVER IP ADDRESS Select voice menu option 13 to configure the IP address of the firmware server CONFIGURATION SERVER IP ADDRESS Select voice menu option 14 to configure the IP address of the configuration server UPGRADE PROTOCOL Select voice menu option 15 to choose firmware and configuration upgrade protocol User can choose between TFTP and HTTP FIRMWARE UPGRADE MODE Select voice menu option 17 to choose firmware upgrade mode among the following three options 1 always check 2 check when pre suffix changes and 3 never upgrade WAN Port WEB ACCESS Select voice menu option 12 to enable WAN Port Wed Access of the device configuration pages CONFIGUR
39. he 1st Sunday of November Default is English Languages supported with voice prompt 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 Maximum NAT ports typically one connection per port NAT TCP timeout in seconds NAT UDP timeout in seconds ISP specified WAN uplink bandwidth FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 31 of 60 Downlink Bandwidth Enable UPnP support Reply to ICMP on WAN port WAN side HTTP Telnet Access White list for WAN side Black list for WAN side Cloned WAN MAC Address Enable LAN DHCP LAN Subnet Mask LAN DHCP Base IP andstream Innovative IP Voice amp Video ISP specified WAN downlink bandwidth If set to Yes the GXW400X would act as an UPnP gateway for your UPnP enable applications UPnP Universal Plug n Play If set to Yes the GXW400X will respond to the PING Default is No Yes setting may make the gateway vulnerable to a Denial of Service attack If 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 List the IP address or IP range in the White list Note for list Rules HTTP telnet Case 1 WAN side telnet HTTP access enabled If white list exists then ONLY these IP addresses are ALLOWED to web and teln
40. he session timer extension enables SIP sessions to be periodically refreshed via a SIP request UPDATE or re INVITE When 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 if 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 for 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 time
41. hind a NAT firewall or a router to set up calls via VolP STUN server is often provided by the VolP service provider or a public STUN server can be used For a sample list of public free STUN servers please refer to http www voip info org wiki STUN This parameter specifies the number of seconds after which a new blank UDP packet will be sent out to the proxy server port in order to have the port stay open and the device remain reachable Default is 20 seconds Use STUN keep alive to detect WAN side network problems If keep alive request does not yield any response for configured number of times the device will restart the TCP IP stack If the STUN server does not respond when the device boots up the feature is disabled Enables GXW400X to download firmware or configuration file using either the TFTP or HTTP S protocols This is the IP address of the configured TFTP server If selected and it is non zero or not blank the GXW400X 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 Do NOT interrupt the firmware upgrade process especially the power supply as this will damage the device Depending on the network environment
42. hout Registration is set to No Allows the user to specify the time frequency in minutes for the GXW400X 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 This parameter allows the user to specify the reregistration time before expiration Defines the local SIP port the GXW400X will listen and transmit The default value for Profile 1 is 5060 and 6060 for Profile 2 Defines the local RTP 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 channel 1 will use port_value 2 for RTP The default value for Profile 1 is 5004 and 6004 for Profile 2 Default is No If set to Yes the device will pick randomly generated SIP and RTP ports This is usually necessary when multiple GXW400X HT50X are behind the same NAT Default is Yes Allows user to hold the phone call before refer it If set to No the call will not be hold before referred Default is No If set to YES then for Attended Transfer the Refer To header uses the transferred target s Contact header information Default is No In which case if conference originator hangs up the conference will be terminated When option YES is chosen originator will transfer other parties to each other so that B and C can choose either to continue the conversation or hang up Default is No
43. ifically mention the IP and sip port where the call will be sent to Sip port that will be annexed to the IP address above TABLE 11 PROFILE SETTINGS When set to Yes the SIP Profile is activated Primary SIP Server s IP address or Domain name provided by VolP service provider Failover SIP Server s IP address or Domain name provided by VoIP Service provider This server will be used if the Primary SIP server becomes unavailable Default is no If set to yes it will register to Primary Server if registration with Failover server expires IP address or Domain name of Outbound Proxy or Media Gateway or Session Border Controller Used by GXW400X 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 Please make sure you re SIP Server or network environment supports SIP over the selected transport method Default is UDP FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 43 of 60 NAT Traversal DNS Mode Primary IP Backup IP1 Backup IP2 Tel URI Use Request Routing ID in SIP INVITE Header SIP Registration Unregister on Reboot andstream Innovative IP Voice amp Video This parameter defines whether the GXW400X NAT traversal mechanism is activated or not If activated by choosing Yes and a STUN server is also specified then the GXW400X performs accord
44. igned 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 end users also have access to the Device Status page MAC Address WAN IP Address Product Model Software Version System Up Time PPPoE Link Up NAT TABLE 8 STATUS PAGE The device ID in hexadecimal format This is needed for Internet Service Provider troubleshooting Note there are separate MAC addresses for the WAN side and the LAN side The LAN MAC address will be used for provisioning and can be found on the label on original box and on the label located on the bottom panel of the device Shows WAN IP address of GXW400X Contains the product model info Program This is the main software release Boot and Loader are not changed often Shows system uptime since the last reboot Shows whether the PPPoE connection is running if connected to DSL modem Shows type of NAT the GXW400X is connected to via its WAN port It is based on STUN protocol FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 33 of 60 andstream Innovative IP Voice amp Video Port Status Displays relevant information regarding the individual FXS ports Example Port Hook Registration DND Forward Busy Delayed Forward Forward FXS1 On Hoo
45. ing 7 numbers and 1617 area code will be added automatically e 1 2 9 xx 2 9 xxxxxx allows dialing to any US Canada Number with 11 digits length e 011 2 9 x allows international calls starting with 011 3469 11 allow dialing special and emergency numbers 311 411 611 and 911 Note In some cases user wishes to dial strings such as 123 to activate voice mail or other application provided by service provider In this case should be predefined inside dial plan feature and the Dial Plan will be x FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 49 of 60 Subscribe for MWI Send Anonymous Anonymous Call Rejection Special Feature Session Expiration Min SE Caller Request Timer Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher Force INVITE andstream Innovative IP Voice amp Video Default is No When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically If this parameter is set to Yes the From header along with Privacy and P_Asserted_Identity headers in outgoing INVITE message will be set to anonymous blocking Caller ID Default is No If set to Yes incoming calls with anonymous Caller ID will be rejected with 486 Busy message Default is Standard Choose the selection to meet some special requirements from Softswitch vendors Example of vendors CBCOM RNK Grandstream implemented SIP Session Timer T
46. ing 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 GXW400X 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 GXW400X will periodically every 20 seconds send a blank UDP packet with no payload data to the SIP server to keep the hole on the NAT open One from the 3 modes are available for DNS Mode configuration A Record for resolving IP Address of target according to domain name SRV DNS SRV resource records indicates how to find services for various protocols NAPTR SRV Naming Authority Pointer according to RFC 2915 Use Configured IP Use the three configured IP address instead of any DNS query One mode can be chosen for the client to look up server The default value is A Record Configure the primary IP for DNS Mode Use Configured IP Configure the first backup IP for DNS Mode Use Configured IP Configure the second backup IP for DNS Mode Use Configured IP The default setting is Disabled If the phone has an assigned PSTN Number this field should be set to User Phone then a User Phone parameter will be attached to the From header in the SIP request to indicate the E 164 number If server supports TEL UR
47. ired to indicate that the SIP account tied to port marked as Active will be used for all members of the same Hunting group Needless to say those members of the same Hunting group may not be sequential ports In following example ports 3 5 and 7 tied to SIP Account configured in Port 1 marked as Active and ports 4 6 8 tied to SIP Account configured in Port 2 marked as Active as well FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 22 of 60 andstream Innovative IP Voice amp Video Example of not sequential configuration of a multiple Hunting Group FXS Port 1 SIP UserID and Authenticate ID entered Hunting group set to Active FXS Port 2 SIP UserID and Authenticate ID entered Hunting Group set to Active FXS Port 3 SIP UserID and Authenticate ID left blank Hunting Group set to 1 FXS Port 4 SIP UserID and Authenticate ID left blank Hunting group set to 2 FXS Port 5 SIP UserID and Authenticate ID left blank Hunting Group set to 1 FXS Port 6 SIP UserID and Authenticate ID left blank Hunting Group set to 2 FXS Port 7 SIP UserID and Authenticate ID left blank Hunting group set to 1 FXS Port 8 SIP UserID and Authenticate ID left blank Hunting Group set to 2 Note A single call directed to the SIP account will NOT result in all ports ringing at the same time They will ring in the hunting group only This feature is applicable to incoming calls only There are two types of hunting gro
48. ith our reseller customers End users should contact the company from whom the product was purchased for replacement repair or refund If you purchased the product directly from Grandstream contact your Grandstream Sales and Service Representative for an RMA Return Materials Authorization number Grandstream reserves the right to change the warranty policy without prior notification 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 download these examples http www grandstream com products gxw_series gxw400x documents qxw400x dui zip for your reference e This document is subject to change without notice The latest electronic version of this user manual is located at http www grandstream com products gxw_series gxw400x documents gxw400x usermanual englis h pdf 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 FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 8 of 60 andstream Innovative IP Voice amp Video CONNECT YOUR GXW400X GATEWAY Connecting the GXW400X gateway is easy Before you begin please verify the contents of the GXW4
49. ive IP Voice amp Video PSTN PASS THROUGH LIFE LINE The RJ 11 line jack on the GXW400X side connected to a legacy PSTN line functions as a pass through jack when the GXW400X loses power or becomes unregistered In this case analog phone connected to FXS port 1 will be directly connected to RJ 11 jack marked as LINE The pass through life line mode enables the user to use the analog phone for PSTN calls directly without using an access code SENDING AND RECEIVING FAX GXW400X 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 If you have problems with sending or receiving Fax toggle the Fax Tone Detection Mode setting SUPPORT RADIUS PROTOCOL GXW400X supports RADIUS for authentication authorization and billing purposes Primary and secondary RADIUS server configurations are available to provide redundancy to this feature In case Primary Radius server becomes unusable RADIUS requests will be automatically sent to the secondary server When at least one RADUIS server was configured the device will allow users to make phone calls only after authorization from RADIUS server has been received CDR Call Detail Record is also sent to the RADIUS se
50. k Registered No 613 FXS2 Off Hook Registered No 614 FXS3 On Hook Not No Registered FXS8 On Hook Registered Yes 615 FXS port 8 user has set Do Not Disturb FXS port 1 user has set his calls to be forwarded unconditionally to ext 613 FXS port 2 user has set his calls to be forwarded to 614 when his phone is busy FXS port 3 user is not registered with his SIP Server Advanced User configuration includes not only the end user configuration but also advanced configurations such as SIP configuration Codec selection NAT Traversal Setting and other miscellaneous configuration FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 34 of 60 Admin Password Layer 3 QoS Layer 2 QoS STUN Server Keep alive interval Use STUN to detect network connectivity Firmware Upgrade amp Provisioning Via TFTP Server andstream Innovative IP Voice amp Video TABLE 9 ADVANCED SETTINGS Administrator password Only the administrator can configure the Advanced Settings page Password field is purposely left blank for security reasons after clicking update and saved The maximum password length is 25 characters 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 Value used for layer 2 VLAN tag Default setting is blank IP address or Domain name of the STUN server STUN server assists devices be
51. kpbs and auto switch to G 711 for Fax Pass through Fax Datapump V 17 V 19 V 27ter V 29 for T 38 fax relay Diffserv TOS 802 1 P Q VLAN tagging RTP In audio RFC2833 and or SIP Info SIP RFC 3261 TFTP HTTP HTTPS TLS SIPS Syslog support HTTP and Telnet access Yes andstream Innovative IP Voice amp Video packet loss concealment AEC with NLP Packetized Voice Protocol Unit supports RTP and AAL2 protocol G168 compliant Echo Cancellation Dynamic Jitter Buffer Modem detection A auto switch to G 711 PSTN failover port on power failure G711 VAD CNG decoder G723 1A G726 ADPCM with 16 24 32 40 bit rates G729A B E iLBC G 726 provides proprietary VAD CNG and signal power estimation Voice Play Annex PLC Annex Il format encoder and Out unit reordering fixed and adaptive jitter buffer clock synchronization AGC automatic gain control Status output Decoder controlling via voice packet header Yes NAT Router or Switched Mode T 38 compliant Group 3 Fax Relay up to 14 4kpbs and auto switch to G711 for Fax Pass through Fax Datapump V 17 V 19 V 27ter V 29 for T 38 fax relay Diffserv TOS 802 1 P Q VLAN tagging RTP In audio RFC2833 and or SIP Info SIP RFC 3261 TFTP HTTP HTTPS TLS SIPS Syslog support HTTP and Telnet access Yes FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 14 of 60 UPnP Support Caller ID Polarity Reversal
52. lected but no Caller ID is configured the selected ring tone will be used for all incoming calls Distinctive ring tones can be configured not only for matching whole number but also for matching prefixes In this case symbol star will be used If server supports Alert Info header and standard ring tone set Bellcore or distinctive ring tone 1 10 is specified then the ring tone in the Alert Info header from server will be used For example If configured as 617 Ring Tone 1 will be used in case of call arrived from Massachusetts Any other incoming call will ring using cadence defined in parameter System Ring Cadence located under Advanced Settings Configuration page Disable Call Waiting Default is No If set to YES Call Waiting indication information will not be provided to analog phone connected to this FXS port Disable Call Waiting Default is No If set to YES Call Waiting caller ID will not be provided to analog phone Caller ID connected to this FXS port Disable Call Waiting Default is No This is to disable the stutter Call Waiting Tone when a Call Waiting call Tone arrives The CWCID will still be displayed Disable Reminder Default is No This is to disable the Reminder Ring that is played when a call is waiting Ring for On Hold Call on hold and the analog phone goes on hook Disable Visual MWI Visual message indicator is a special on hook caller ID type message that enables and disables the message waiting light on cert
53. line Lines are skipped if they are still busy on a previous call These two hunting styles can be configured from the Profile_x page Default value is 20 seconds In case this feature activated using codes 92 code the call will be forwarded after this preconfigured amount of time Default is 4 seconds Call will be completed within this time interval if no additional key entry occurs 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 Allows users to configure the key as the Send or Dial key If set to Yes will send the number In this case this key is essentially equivalent to the Dial key If set to No this key can be included as part of number
54. lt is 200 In 100 1 0000 milliseconds range Disconnect duration Proxy Require SIP Extension to notify SIP server that the unit is behind the NAT Firewall Enable Hook Flash Default is Yes If set to No FLASH button could only be used for terminating calls Hook Flash timing Time period when the cradle is pressed Hook Flash to simulate FLASH To prevent unwanted activation of the Flash Hold and automatic phone ring back adjust this time value On Hook Timing On hook timing is the minimum time for an on hook event to be validated Gain Handset volume adjustment RX is for receiving volume direction FXS gt to analog phone TX is for transmission volume Analog phone gt to FXS Default values are 0dB for both parameters Loudest volume 6dB Lowest volume 6dB Ring Tones Configure ring cadences according to preference SAVING THE CONFIGURATION CHANGES After user makes a change to the configuration press the Update button in the Configuration Menu The web browser will then display a message window to confirm saved changes Grandstream recommends reboot or power cycle the IP phone after saving changes REBOOTING FROM REMOTE Press the Reboot button at the bottom of the configuration menu to reboot the phone remotely The web browser will then display a message window to confirm that reboot is underway Wait 30 seconds to log in again CONFIGURATION THROUGH A CENTRAL SERVER Grandstream GXW400X can be
55. nding authentication challenge This field enables user to store different versions of firmware files in one single directory on the firmware server If configured only the firmware file with the matching prefix will be downloaded This field enables user to store different versions of firmware files in one single directory on the firmware server If configured only the firmware file with the matching postfix will be downloaded This field enables user to store different configuration files in one single directory on the configuration server If configured only the configuration file with the matching prefix will be downloaded This field enables user to store different configuration files in one single directory on the configuration server If configured only the configuration file with the matching postfix will be downloaded If set to Yes configuration and upgrade server information can be obtained using DHCP option 66 or option 160 from DHCP server located in customer s environment Note If DHCP Option 66 is enabled the gateway will attempt downloading a configuration file from the server URL provided by DHCP even though Config Server Path is left blank FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 36 of 60 Automatic Upgrade Disable SIP NOTIFY Authentication Authenticate Conf File Firmware Key SIP TLS Certificate SIP TLS Private Key SIP TLS Private Key Password ACS URL ACS Username
56. on Conference Hangup is set to No If the configuration is set to Yes A will transfer B to C so that B and C can continue the conversation HUNTING GROUP This feature allows the user to setup a single SIP account on the gateway and have the ability to use all FXS ports to make receive calls Using this feature all ports active in same Hunting Group will have the same phone number and incoming calls will be distributed in a Linear or Circular manner among the ports active in that Hunting Group The number of hunting groups is limited by the number of ports each GXW model has i e each port can be its own Hunting Group The most practical and efficient way to use Hunting Groups is to assign 2 or 3 ports to separate Hunting Groups One additional and popular way to use the Hunting Group feature is called multiplexed analog lines In this configuration a legacy PBX system with 8 FXO trunks can be connected to 8 GXW 4008 ports FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 21 of 60 andstream Innovative IP Voice amp Video configured as Hunting Group The GXW can be registered to a SIP server provider using only one phone number If the SIP service provider allows multiple calls to the same number the GXW will allow 8 concurrent calls to the same SIP number All office members can be reached remotely using the same phone number in a round robin fashion Example Configuration of a typical Hunting Group 1 Con
57. on page 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 Athen dials Caller C s number followed by or wait for 4 seconds FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 20 of 60 andstream Innovative IP Voice amp Video 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 GXW400X 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 3 WAY CONFERENCING The GXW400X supports Bellcore style 3 way Conference Instructions for 3 way conference Assuming that call party A and B are in conversation A GXW400X wants to bring C in a conference 1 A presses FLASH on the analog phone or Hook Flash for old model phones to get a dial tone A dials C s number then or wait for 4 seconds If C answers the call then A presses FLASH to bring B C in the conference If C does not answer the call Acan press FLASH back to talk to B If A presses FLASH during conference C will be dropped out o o fF wo LV If A hangs up the conference will be terminated for all three parties when configuration Transfer
58. or message Example May 19 02 40 38 192 168 1 14 GS LOG 00 0b 82 00 a1 be 000 Ethernet link is up If Syslog is enabled and Send SIP Log is set to YES then SIP messages will also be delivered via Syslog Default is set to NO FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 39 of 60 Primary RADIUS Server Primary RADIUS Auth Port Primary RADIUS Acct Port Primary Radius Server Secret Secondary RADIUS Server Secondary RADIUS Auth Port Secondary RADIUS Acct Port Secondary Radius Server Secret RADIUS Timeout RADIUS Retry Download Device Configuration Download Device XML Configuration Upload firmware Upload configuration FXS Port SIP User ID andstream Innovative IP Voice amp Video IP Address or FQDN of the primary RADIUS Server Primary Radius server authentication port Default value is 1812 Primary Radius server accounting port Default value is 1813 Special secret string should be preconfigured according to RADIUS Server configuration IP Address or FQDN of the secondary RADIUS Server Secondary Radius server authentication port Default value is 1812 Secondary Radius server accounting port Default value is 1813 Special secret string should be preconfigured according to RADIUS Server configuration Default value is 2 seconds The time between retries the GXW will send Access Request message to RADIUS server in purpose to authenticate it Default value is 3 times Num
59. r FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 50 of 60 Send Re INVITE After Fax Enable Silence Detection for Fax Disconnect Enable 100rel Add Auth Header On Initial REGISTER Use First Matching Vocoder in 2000K SDP Preferred Vocoder Voice Frames per TX G723 Rate iLBC Frame Size iLBC Payload type AAL2 G726 16 Payload type AAL2 G726 24 Payload type AAL2 G726 32 Payload type AAL2 G726 40 Payload type G729E payload type andstream Innovative IP Voice amp Video Default is No If set to Yes device will send an INVITE with audio vocoders upon completition of Fax to continue session in audio only For fax machines that do not send a Disconnect when fax is done This option Enables Disables the detection of silence in order to know the fax has finished The silence period is non configurable and fixed to 7 seconds Enables the use of PRACK Provisional Acknowledgment method If set to Yes device will include authorization header in the Register request Default is No If set to Yes device will include only the first match vocoder in its 2000K response otherwise it will include all match vocoders in same order received in INVITE The GXW400X supports up to 5 different Vocoder types including G 711 A U law G 726 Supports bit rates 16 24 32 and 40 G 723 1 G 729A B E and iLBC The user can configure Vocoders in a preference list that will be included with the same
60. rver for billing purposes RAIDUS server can send requests to terminate calls when run out of pre paid credit The GXW400X will be able to work in VoIP billing environment using redundant double server configuration User will be able to configure primary and secondary RADUIS server IP Addresses or FQDNs Once at least one RADUIS server was configured the device will allow users to make phone calls only after permission from RADIUS server has been received In case Primary Radius server becomes unusable secondary will take role of primary and will manage credit recourses in the network Imbedded RADIUS client also supports request generated by Radius server to terminate calls when run out of pre paid credit FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 24 of 60 andstream Innovative IP Voice amp Video CALL FEATURES GXW400X supports the traditional telephony features available in a PBX as well as additional advanced telephony features Key 02 03 16 17 30 31 KH 82 47 50 Di 69 70 71 72 73 74 78 79 TABLE 6 CALL FEATURES TABLE STAR CODE Call Features Forcing a Codec per call 027110 PCMU 027111 PCMA 02723 G723 02729 G729 0272616 G726 r16 0272624 G724 r24 0272632 G726 132 0272640 G726 r40 027201 iLBC Disable LEC pe call Dial 03 number No dial tone is played in the middle Enable SRTP Dis
61. s of an active registration SIP server unreachable of all FXS profiles If set to YES when GXW400X recognizes a loss of registration all outgoing calls will be routed to an FXO gateway The use of this option presumes a configured GXW410x or another FXO gateway with an active PSTN line connection IP Address or URI of the FXO gateway URI or IP address of the NTP Network Time Protocol server Used by the phone to synchronize the date and time An extensive list of public NTP servers can be found at http www ntp org Default is 1440 Updates the Network Time Protocol Values range from 5 1440 minutes The IP address or URL of System log server The server collects system log information from the device Select the GXW400X 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 1 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 Si Sn Sl ge wl Si N 10 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 err
62. set button Press for 7 seconds to reset factory default settings DC 12V Power adapter connection CONSOLE Console connection for a PC LINE When the unit loses power or unit became unregistered FXS port 1 will be able to make receive calls from the PSTN line connected to this port FXS1 FXS8 FXS port to be connected to analog phones fax machines Note Once the GXW400X is turned on and configured the front display panel indicates the status of the unit FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 10 of 60 stream Innovative IP Voice amp Video GXW4008 Display FXS port LEDs status FIGURE 2 DIAGRAM OF GXW4004 8 DISPLAY PANEL 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 Console Turns ON when Console port is being used LAN LED Indicates LAN or WAN port activity WAN LED Indicates PC or LAN port activity LED 1 8 Indicate status of the respective FXS Ports on the back panel Busy ON Solid Green Available OFF Slow blinking FXS LEDs indicates Voice Mail for that port NOTE e All LED lights display green when ON e Slow blinking of READY WAN and LAN LED together indicates a firmware upgrade or provisioning state e LED POWER READY and WAN lights are ON when device is up and running and successfully registered to the SIP Server e Slow blinking of READY LED indicates that d
63. ss Through must use codec PCMU PCMA Default is Callee This decides whether Caller or Callee sends out the re INVITE for T 38 or Fax Pass Through Select either Fixed or Adaptive based on network conditions Select Low Medium or High based on network conditions e High initial 200ms min 40ms max 600ms Note not all vocoders can meet the high requirement e Medium initial 100ms min 20ms max 200ms e Low initial 50ms min 10ms max 100ms Default is Disabled Other options are Enabled but not forced and Enabled and forced It uses SDP Security Description to exchange key Please refer SDES http www apps ietf org rfc ric4568 html SRTP http www apps ietf org rfe rfca711 html Default is Enabled Allows user to enable or disable Crypto life time when using SRTP Depends on standard phone type and location Select the value according to the local Telco standard where the GXW400x is installed Please refer to the pull down list to select Default is No If set to Yes polarity will be reversed upon call establishment and termination Set to Yes if the traditional PBX you are using with GXW400X uses this method for signaling call termination Default is No Default is No If set to Yes it will play busy reorder tone before loop current disconnect upon call fail FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 52 of 60 andstream Innovative IP Voice amp Video Loop Current Defau
64. sted above e The IVR announces 12 digits IP address you need to strip out the leading 0 in the IP address For ex IP address 192 168 001 014 you need to type in http 192 168 1 14 in the web browser Once the HTTP request is entered and sent from a web browser the user will see a log in screen There are two default passwords for the login page User Level Password Web pages allowed End User Level 123 Only Status and Basic Settings Administrator Level Admin Browse all pages The password is case sensitive with maximum length of 25 characters The factory default password for End User and administrator is 123 and admin respectively Only an administrator can access the ADVANCED SETTING Profile 1 Profile 2 and FXS PORTs configuration pages Please reference the GUI pages using the following link http www grandstream com products gxw_series gxw400x documents gxw400x qui Zip FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 28 of 60 andstream Innovative IP Voice amp Video 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 If 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
65. ter 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 e e e e e 9 to go to the next selection in the list PCM U PCMA iLBC G 726 G 723 G 729 Announces the Mac address of the unit Enter 9 to toggle between enabled and disabled 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 FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 17 of 60 15 16 17 47 71 78 86 99 andstream Innovative IP Voice amp Video Upgrade Protocol Upgrade protocol for firmware and configuration update Enter 9 to toggle between TFTP and HTTP Firmware Version Firmware version information Firmware Upgrade Firmware upgrade mode Enter 9 to rotate among the following three options 1 always check 2 check when pre suffix changes 3 never upgrade Direct IP Calling Enter the target IP address to make a direct IP call after dial tone See Make a Direct IP Call Phone calls between Applies to GXW4004 GXW4008 User can make internal calls different ports of the same between different ports inside the same GW even without being GW 400x registered to SIP server By dialing and 7x user can reach colleague connected to port x Voice Mail Number of voice mails RESET Enter 9 to reboot the
66. the target IP address port is 192 168 1 20 5062 then the dialing convention would be 47 or Voice Prompt with option 47 then 192 168 0 160 5062 followed by pressing the key if it is configured as a send key or wait for 4 seconds NOTE When completing direct IP call the Use Random Port should set to NO You can not make direct IP calls between FXS ports since they are using same IP CALL HOLD Place a call on hold by pressing the flash 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 2 short beeps indicates an incoming call if the call waiting feature is enabled Toggle between incoming call and current call by pressing the flash button First call is placed on hold 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 3 Caller Apresses FLASH on the analog phone to hear the dial tone 4 Caller A dials 87 then dials caller C s number and then or wait for 4 seconds 5 Caller A will hear the dial tone Then A can hang up NOTE Enable Call Feature must be set to Yes in web configurati
67. the web configuration interface Grandstream recommends end user use the Grandstream HTTP server Its address can be found at http Awww grandstream com support firmware Currently the HTTP firmware server URL is firmware grandstream com For large companies we recommend to maintain their own TFTP HTTP HTTPS server for upgrade and provisioning procedures 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 GXW will attempt to retrieve the new image files by downloading them into the GXW400x s SRAM During this stage the GXW s LEDs will blink until the checking downloading process is completed Upon verification of checksum the new code image will then be saved into the Flash If TFTP HTTP HTTPS fails for any reason e g TFTP HTTP HTTPS server is not responding there are no code image files available for upgrade or checksum test fails etc the GXW will stop the TFTP HTTP HTTPS process and simply boot using the existing code image in the flash Firmware upgrade may take as long as 15 to 30 minutes over Internet or just 5 minutes if it is performed on a LAN It is 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 FIRMWA
68. this process may take up to 15 or 20 minutes FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 35 of 60 Via HTTP or HTTPS Server Firmware Server Path Config Server Path XML Config File Password HTTP HTTPS User Name HTTP HTTPS Password Always send HTTP Basic Authentication Information Firmware File Prefix Firmware File Postfix Config File Prefix Config File Postfix Allow DHCP Option 66 or 160 to override server andstream Innovative IP Voice amp Video The URL for the HTTP server used for firmware upgrade and configuration via HTTP Note If Auto Upgrade is set to No GXW400X will only do HTTP download once at boot up IP address or domain name of firmware server That URL of the server that hosts the firmware release The default server is fm grandstream com gs IP address or domain name of configuration server The server hosts a copy of the configuration file to be installed on the gateway The default server is fm grandstream com gs The password used for encrypting the XML configuration file using OpenSSL This is required for the phone to decrypt the encrypted XML configuration file The user name needed to authenticate withthe HTTP HTTPS server The password needed to authenticate with the HTTP HTTPS server Default is Send HTTP Basic Authentication Information only when challenged If set to Always device will send configured user name and password within HTTP request without server se
69. tone Please refer the document below to determine your local call progress tones http www itu int ITU T inr forms files tones 0203 pdf Key pattern to get Prompt Access Tone Maximum 20 digits No default If set to Yes the configuration update via keypad is disabled Disables the voice prompt configuration Default is No If set to Yes accessing integrated voice menu will be impossible Disables the Direct IP Call function Default is No If set to Yes direct IP to IP calling will not be supported Life line feature ensures user can place receive a PSTN call in an emergency situation 1 If set to Auto in case of power loss or loss of SIP registration the PSTN line will be seamlessly connected to analog phone connected to FXS port 2 If set to Always Connected the PSTN line will be always connected to the phone connected to FXS port VoIP calls will not be allowed in this configuration 3 If set to Always Disconnected user can only place VoIP calls regardless of any power loss and or SIP registration problems User will be unable to place receive any PSTN calls FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 38 of 60 Failover to FXO Gateway FXO Gateway NTP server NTP Update Interval Syslog Server Syslog Level Send SIP Log andstream Innovative IP Voice amp Video This feature allows users to place an outbound PSTN call in case there is a los
70. ts with the same phone number Each incoming call will be routed to first available port in Linear or Circular mode User may configure all ports as members of the same Hunting Group or it may configure different port combinations for more than one Hunting Group For example Ports 1 3 and 5 are members of the same Hunting Group the rest of the ports may have separate numbers and may be reached independently Any port member of a Hunting Group that is not registered with a SIP account will be able to place outbound calls using the SIP credentials of the primary Hunting Group port For example Port 1 3 and 5 are members of the same Hunting Group Port 1 is registered with a SIP account Ports 3 and 5 are not registered Ports 3 and 5 will be able to place outbound calls using the SIP account of port 1 Select appropriate value for Hunting Group feature The original SIP account should be set to Active while the group members should be set to the port number of the Active Port Example configuration of a multiple Hunting group FXS Port 1 SIP UserID and Authenticate ID entered Hunting group set to Active FXS Port 2 SIP UserlD and Authenticate ID left blank Hunting Group set to 1 FXS Port 3 SIP UserlD and Authenticate ID left blank Hunting Group set to 1 FXS Port 4 SIP UserID and Authenticate ID entered Hunting group set to Active FXS Port 5 SIP UserID and Authenticate ID left blank Hunting Group set to 4 FXS Port
71. ups Linear and Circular Linear style will sort the call to the lowest numbered available line this is also called serial hunting Circular style will distribute the calls round robin If a call is assigned to line 1 the next call goes to 2 and the next to 3 The succession throughout each of the lines continues even if one of the previous lines becomes available When the end of the hunt group is reached the hunting starts over at the first line Lines are skipped if they are still busy on a previous call These two hunting styles can be configured from the Profile_x page INTER PORT CALLING In some cases a user may want to make phone calls between the phones connected to multiple ports of the same gateway when it is used as a stand alone unit without the use of a SIP server This feature will also be applicable when the gateway is used with Hunting Groups and is registered to SIP server only with one master number In such cases users still will be able to make inter port calls by using the IVR feature For example on the GX4004 and GXW4008 the user connected to port 1 can reach the user connected to port 3 by dialing and 73 Digit 7 indicated using inter port calling feature digit 3 indicates the port number which should be reached At the same manner the user connected to port 4 can reach the user connected to port 8 by dialing and 78 FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 23 of 60 andstream Innovat
72. vacy or PPI header when special feature is not Telkom SA or CBCOM If set to Default it will only add Privacy or PPI header when special feature is not Telkom SA or CBCOM T1 is an estimate of the round trip time between the client and server transactions If the network latency is high select larger value for more reliable usage Maximum retransmission interval for non INVITE requests and INVITE responses Set the SIP Timer D Default is 0 Sets the payload type for DTMF using RFC2833 The GXW400X supports up to 3 different DTMF methods including in audio via RTP RFC2833 and via Sip Info The user can configure DTMF method in a priority list Default is No If set to yes use above DTMF order without negotiation Default is No If set to yes flash will be sent as a DTMF event Default is Yes If Yes call features using star codes will be supported locally 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 FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 46 of 60 andstream Innovative IP Voice amp Video Use SIP User Agent Used to replace SIP User Agent Header No Default Header Distinctive Ringtone Custom Ring 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 When se
73. you can make a Conference by pressing Flash key If set to Yes you need to dial 23 second callee number Default is No If set to Yes the Outbound Proxy will be removed from the route header FIRMWARE VERSION 1 0 14 1 GXW400X USER MANUAL Page 45 of 60 Support SIP Instance ID Validate incoming SIP message Check SIP user ID for incoming INVITE Authenticate incoming INVITE Allow Incoming SIP Messages from SIP Proxy Only Use Privacy Header Use P Preferred Identity Header SIP T1 Timeout SIP T2 Interval SIP Timer D DTMF Payload Type Preferred DTMF method in listed order Disable DTMF Negotiation Send Hook Flash Event Enable Call Features Proxy Require Use NAT IP andstream Innovative IP Voice amp Video Default is Yes If set to Yes the contact header in REGISTER request will contain SIP Instance ID as defined in IETF SIP Outbound draft Default is No If set to yes all incoming SIP messages will be strictly validated according to RFC rules If message does not pass validation process call will be rejected Default is No Check the SIP User ID in Request URI If they don t match the call will be rejected Default is No If set to Yes device will challenge the incoming INVITE for the Authenticate ID and Password with 401 Unauthorized Default is No If incoming SIP message does not match with SIP Server it will be rejected If set to Default it will only add Pri

GXW400x User Manual - Grandstream Networks, Inc

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