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gxw4248 features - e4 Technologies

1. 51 INSTRUCTIONS FOR LOCAL FIRMWARE UPGRADE sse 52 CONFIGURATION FILE 52 FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX 53 MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD 53 RESTORE FACTORY DEFAULT SETTING 54 FACTORY RESET eos csc 54 RESET BUTTON 54 IVR COMMAND tans 54 FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 3 OF 55 andstream Innovative IP Voice amp Video TABLE OF FIGURES GXW4248 User Manual Figure 1 Diagram of GXW4248 10 Figure 22 50 Pin a oed 11 Figure 3 ECD 16 5 GXW4248 User Manual Table 1 Definitions of the GXW 10 Table 2 Definitions of the GXW Display 10 Table 3 GXW4248 Software 13 Table 4 Hardware Sp
2. BLIND THANSFER iR REEL Edo ATTENDED TRANSFER te ete obs 3 WAY CONFERENCING Lu cece eese rennen nre HUNTING GROUP ae ee INTER PORT CALLING iaa SENDING AND RECEIVING SUPPORT RADIUS FIRMWARE 1 0 3 4 GXW42XX USER MANUAL andstream Innovative IP Voice amp Video PAGE 2 OF 55 andstream Innovative IP Voice amp Video CALL FEATURES om emer an 25 CONFIGURATION GUIDE isis sessstssseciassssrnsaisdeosseriseranscesininsaiceracenstemieneiens 27 CONFIGURING GXW4248 VIA VOICE PROMPT 27 CONFIGURING GXW4248 WITH WEB BROWSER 27 ACCESS THE WEB CONFIGURATION MENU 28 IMPORTANT SETTINGS sss ttti ttt ttti ttt tenni 28 28 DTMF co T MTM 29 PREFERRED VOCODER CODEQ ttt ttti 29 SAVING THE CONFIGURATION 49 REBOOTING FROM REMOTE Rt ctis Quince dei 49 CONFIGURATION THROUGH A CENTRAL 50 SOFTWARE E Saa CR e CERE 51 FIRMWARE UPGRADE THROUGH
3. G723 Rate G726 32 Packing Mode 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 Use First Matching Vocoder in 2000K SDP FIRMWARE 1 0 3 4 andstream Innovative IP Voice amp Video The GXW4248 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 Sets the payload type for DTMF using RFC2833 The GXW4248 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 iLBC The user can configure Vocoders in a preference list that will be included with the same preference order in SDP message The first Vocoder is entered by choosing the appropriate option in Choice 1 The last Vocoder is entered by choosing the appropriate option in Choice 8 Number of the frame size when it transmits 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 Choose the packing mode for G726 32 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 10
4. When set to No GXW4248 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 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 Web Port Disable Telnet User Password Admin Password By default HTTP uses port 80 This field is for customizable web port If set to Yes telnet access will be disabled Default is No Set new password for web GUI access as User This field is case sensitive with a maximum length of 30 characters Set new password for web GUI access as Admin This field is case sensitive with a maximum length of 30 characters TR 069 ACS URL ACS Username ACS Password Periodic Inform Enable Periodic Inform Interval Connection Request Username FIRMWARE 1 0 3 4 URL of the TR 069 Auto Configuration Servers e g http acs mycompany com or IP address User specify the ACS Username 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 The user name for the TR 069 Auto Configuration Server to connect to the phone GXW42XX USER MANUA
5. amp Video ATTENDED TRANSFER Assume that Caller A and B are in conversation Caller A wants to Attend Transfer B to C 1 Caller A presses FLASH on the analog phone for dial tone 2 Caller A then dials Caller C s number followed by or wait for 4 seconds 3 If Caller C answers the call Caller A and Caller C are in conversation Then A can hang up to complete transfer 4 If Caller C does not answer the call Caller A can press flash to resume call with Caller B NOTE When Attended Transfer fails and A hangs up the GXW4248 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 GXW4248 supports Bellcore style 3 way Conference Instructions for 3 way conference Assuming that call party and are in conversation GXW4248 wants to bring in a conference 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 in the conference If C does not answer the call A can press FLASH back to talk to B If A presses FLASH during conference C will be dropped out eo ON x If A hangs up the conference will be terminated or transfer to if Transfer on Conference Hangup set to yes HUNTING GROUP This feature allows the user to setup a single SIP account on the gatew
6. 099 and wait for reset voice prompt Enter the encoded MAC address Look below on how to encode MAC address Boo 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 Keyinthe MAC address Use the following mapping 0 9 0 9 FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 54 OF 55 andstream Innovative IP Voice amp Video 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 gt For example if the MAC address is 000582006395 it should be keyed in as 0002228200333395 NOTE 1 Factory Reset will be disabled if the Lock keypad update is set to Yes 2 f still have difficulties to get access to the device pushing the reset button during booting up will trigger the RECOVERY MODE and device will use static IP 192 168 1 234 24 You may need a direct connection to access the configuration page FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 55 OF 55
7. Number of Concurrent Calls except when using SRTP Voice over Packet Capabilities FIRMWARE 1 0 3 4 TABLE 3 GXW4248 SOFTWARE FEATURES GXW4248 48 FXS ports 48 SIP accounts 4 profiles 10 100 1000 Mbps RJ 45 48 Concurrent Calls Voice Activity Detection VAD with CNG comfort noise generation and PLC packet loss concealment LEC with NLP Packetized Voice Protocol Unit supports GXW42XX USER MANUAL PAGE 13 OF 55 Voice Compression DHCP Server Client Fax over IP QoS Transport Protocol DTMF Method IP Signaling Provisioning Security Management Dial Plan 3 Way Conference Caller ID Polarity Reversal Wink Network Connectivity andstream Innovative IP Voice amp Video RTP RTCP and AAL2 protocol 168 compliant Echo Cancellation Dynamic Jitter Buffer Modem detection amp auto switch to G 711 G 711 Annex PLC Annex VAD CNG format encoder and decoder 722 pending G 723 1A G 726 ADPCM with 16 24 32 40 bit rates G 729 iLBC G 726 provides proprietary VAD CNG and signal power estimation Voice Play Out unit reordering fixed and adaptive jitter buffer clock synchronization AGC automatic gain control Status output Decoder controlling via voice packet header DHCP Client only T 38 compliant Group 3 Fax Relay up to 14 4kpbs and auto switch to G 711 for Fax Pass through Fax Datapump V 17 V 21 V 27ter V 29 for T 38 fax relay DiffServ TOS 802 1P Q VLAN tag
8. 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 GXW42XX USER MANUAL PAGE 47 OF 55 Innovative IP Voice amp Video Ring Tones Configure ring cadences according to preference FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 48 OF 55 Port Settings Advanced Port Settings FXO Mapping SIP User ID Authenticate ID Password Name Profile Enable FXS TR 069 Offhook Auto dial Hunting Group Map to FXO Port Map to FXO Gateway IP And Port andstream Innovative IP Voice amp Video TABLE 11 FXS PORTS User account information provided by VolP service provider ITSP Usually in the form of digit similar to phone number or actually a phone number SIP service subscribers Authenticate ID used for authentication Can be identical to or different from SIP User ID SIP service subscriber s account password for GXW4248 to register to SIP servers of ITSP Any name to identify this specific user Select the corresponding Profile ID 1 2 3 4 Enable or
9. gt xxxxxxx Block any number of leading digits 1900 and add prefix 1617 for any dialed 7 digit numbers 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 1900x 216172 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 1900x prevents dialing any number started with 1900 e lt 1617 gt 2 9 xxxxxx allows dialing to local area code 617 numbers by dialing 7 numbers and 1617 area code will be added automatically 1 2 9 xx 2 9 xxxxxx allows dialing to any US Canada Number with 11 digits length 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 1 0 3 4 GXW42XX USER MANUAL PAGE 45 OF 55 Use as Dial Key No Key Entry Timeout Off Hook Auto Dial Delay Enable Call Features Disable Call Waiting Disable Call Waiting Call
10. of the GXW4248 If the download of cfgxxxxxxxxxxxx xml is not successful the provision program will issue request a generic configuration file cfg xml 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 GXW4248 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 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 tool
11. 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 GXW42XX USER MANUAL PAGE 44 OF 55 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 at least 2 digits number xx at least 2 digits number a b c exclude d 3 5 any digit of 3 4 or 5 e 147 any digit 1 4 or 7 f 22011 replace digit 2 with 011 when dialing 9 lt 1 gt leading 1 to all numbers dialed vice versa will remove a 1 from the number dialed h or Example 1 369 11 1617 Allow 311 611 911 and any 10 digit numbers of leading digits 1617 Example 2 1900 lt 1617
12. 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 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 GXW4248 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 Server Path A configuration parameter is associated with each particular field in the web configuration page A parameter consists of a Capital letter P and 1 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 r
13. the analog phone goes on hook Visual message indicator is a special on hook caller ID type message that enables and disables the message waiting light on certain phones GXW4248 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 This is the type of signal sent to the analog phone to make it turn the lamp ON upon receiving a Voice mail Check the phone s manual to find out what signal is supported FSK default or Neon Defines whether the call is transferred to the other party if the conference initiator hangs up Disable Bellcore style 3 Way Conference GXW42XX USER MANUAL PAGE 46 OF 55 Send Hook Flash Event Ring Timeout Delayed Call Forward Wait Timeout Send Anonymous Anonymous Call Rejection Hunting Group Type Hunting Group Ring Timeout Special Feature andstream Innovative IP Voice amp Video Default is No If set to yes flash will be sent as a DTMF event Incoming call will stop ringing when not picked up given a specific period of time 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 If this parameter is set to Yes the From header along with Privacy and P Asserted ldentity headers in outgoing INVITE message will be set to
14. 00 0b 82 00 a1 be 000 Ethernet link is up Print SIP in Enable or disable printing of full SIP messages in Syslog Syslog TABLE 9 ADVANCED SETTINGS System Ring Configuration option for all FXS ports ring cadence for all incoming calls Cadence Syntax c on1 off1 on2 off2 on3 off3 Default is set to c 2000 4000 US standards FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 36 OF 55 Innovative IP Voice amp Video Call Progress Using these settings user can configure tone frequencies according to user preference By Tones 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 Dial tone Ringback tone Busy tone Reorder tone Confirmation tone Call Waiting 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 FXO Failover Failover to FXO Enable or disable the Failover FXO Gateway Gateway FXO Gateway IP IP Address or URI of the FXO gateway System Features Disable Direct IP Disables the Direct IP Call function Default is No If set to Yes direct IP to IP calling will Call not be suppo
15. 002 003 004 005 to set up IP address Subnet Mask Gateway and DNS server respectively PPPoE MODE Select voice menu option 001 to enable GXW4248 to use PPPoE mode FIRMWARE SERVER IP ADDRESS Select voice menu option 013 to configure the IP address of the firmware server CONFIGURATION SERVER IP ADDRESS Select voice menu option 014 to configure the IP address of the configuration server UPGRADE PROTOCOL Select voice menu option 015 to choose firmware and configuration upgrade protocol User can choose between TFTP HTTP and HTTPS FIRMWARE UPGRADE MODE Select voice menu option 017 to choose firmware upgrade mode among the following three options 1 always check 2 check when pre suffix changes and 3 never upgrade CONFIGURING GXW4248 WITH WEB BROWSER The GXW4248 gateway has an embedded Web server that allows users to configure the GXW4248 through a web browser FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 27 OF 55 andstream Innovative IP Voice amp Video ACCESS THE WEB CONFIGURATION MENU The GXW4248 HTML configuration menu can be accessed via Ethernet port To access the HTML configuration menu from the Ethernet port 1 Follow table 4 to find the Ethernet port IP address 2 Open a web browser type in the IP address for example http GXW4248 IP Address the GXW 4248 IP Address is the Ethernet IP address for the GXW4248 NOTE e The IVR announces 12 digits IP address you need to strip out the lea
16. 4 Range is from 96 to 127 Default value is 103 Range is from 96 to 127 Default is No If set to Yes device will include only the first match vocoder in its 200 response otherwise it will include all match vocoders in same order received in INVITE GXW42XX USER MANUAL PAGE 42 OF 55 SRTP Mode Silence Suppression VAD Jitter Buffer Type Jitter Buffer Length SLIC Setting Caller ID Scheme Polarity Reversal Loop Current Disconnect Loop Current Disconnect duration Enable Hook Flash Hook Flash timing Minimum Maximum On Hook Timing FIRMWARE 1 0 3 4 Andstream Innovative IP Voice amp Video 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 ric ric4568 html SRTP http www apps ietf org r c r c3711 html Default is No VAD allows detecting the absence of audio and conserve bandwidth by preventing the transmission of silent packets over the network 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 initial 50ms min 10ms 100ms Depends on standard phone type and location Select the value according to the local
17. ACT LED blinking in following status indicate different network speed 10M LINK and ACT all blink ACT blinks faster then LINK 100M LINK and ACT all blink ACT blinks the same fast as LINK 1000M LINK does not blink only ACT blink e Slow blinking of FXS LED together indicates a firmware upgrade or provisioning state FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 12 OF 55 andstream Innovative IP Voice amp Video GXW4248 FEATURES The GXW4248 is a next generation IP voice gateway that is interoperable and compatible with leading IP PBXs Soft switches and SIP platforms The GXW4248 FXS gateway is auto configurable remotely manageable and scalable The GXW4248 offers superb voice quality traditional telephony functionality easy deployment and 48 FXS ports 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 48 FXS ports RJ 45 Ethernet ports 4configurable SIP profiles e Supports Voice Codecs G711 a u Annex amp 11 G723 1A G726 ADPCM with 16 24 32 40 bit rates G729 iLBC T 38 Fax e Comprehensive Dial Plan support for Outgoing calls 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 Telephone Interfaces SIP Provisioning Network Interface
18. L PAGE 33 OF 55 Request Password Andstream Innovative IP Voice amp Video The password for the TR 069 Auto Configuration Server to connect to the phone Security Settings SIP TLS Certificate SIP TLS Private Key SIP TLS Private Key Password Primary RADIUS Server Primary RADIUS Authentication Port Primary RADIUS Account Port Primary RADIUS Server Secret Secondary RADIUS Server Secondary RADIUS Authentication Port FIRMWARE 1 0 3 4 The GXW4248 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 SIP TLS Private Key The user specified SIP TLS private key used for SIP over TLS is in X 509 format SSL Private key password used for SIP Transport in TLS TCP Remote Authentication Dial In User Service RADIUS GXW4248 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 server for billing purposes RAIDUS server can send
19. P FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 38 OF 55 Proxy Require Andstream Innovative IP Voice amp Video ASIP Extension to notify the SIP server that the phone is behind a NAT Firewall SIP Settings Basic Settings SIP transport SIP Registration Unregister on Reboot Outgoing Calls Without Registration Register Expiration SIP Registration Failure Retry Wait Time Reregister Before Expiration Local SIP Port Use Random SIP Port SIP T1 Timeout SIP T2 Timeout FIRMWARE 1 0 3 4 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 This parameter controls whether the GXW4248 needs to send REGISTER messages to the proxy server The default setting is Yes Default is No If set to Yes the SIP user s registration information is cleared on reboot Default is No If set to Yes user can place outgoing calls even when not registered if allowed by 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 ou
20. SIP User ID the call will be rejected Direct IP calling will also be disabled If set to Yes SIP User ID will be checked in the Request URI of the incoming INVITE If it doesn t match the phone s SIP User ID the call will be rejected Direct IP calling will also be disabled If set to Yes the phone will challenge the incoming INVITE for authentication with SIP 401 Unauthorized response Fax Mode Fax Tone Detection Mode Send Re INVITE After Fax Completion Send Re INVITE After Fax Tone Enable Silence Detection for Fax Disconnect Audio Settings T 38 Auto Detect by default or Pass Through must use codec PCMU PCMA Default is Callee This decides whether Caller or Callee sends out the re INVITE for 38 or Fax Pass Through 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 If set to Yes device will send an Re INVITE after Fax tone is detected disabling will only work under Broadsoft feature 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 FIRMWARE 1 0 3 4 GXWA2XX USER MANUAL PAGE 41 OF 55 Preferred method in listed order Disable DTMF Negotiation DTMF Payload Type Preferred Vocoder Voice Frames per TX
21. Telco standard where the 4248 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 GXW4248 uses this method for signaling call termination Default is No Default is 200 In 100 1 0000 milliseconds range If no flash will be treated as an on hook event Sets the minimum maximum time the phone is on hook before being detected as a hook flash The range is 40 to 2000 milliseconds Sets the time required to detect that the phone is on hook The range is 40 to 2000 milliseconds Default is 400 milliseconds GXW42XX USER MANUAL PAGE 43 OF 55 Disable Line Echo Canceller LEC SLIC Setting Caller ID Scheme andstream Innovative IP Voice amp Video 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 OdB for both parameters Loudest volume 6dB Lowest volume 6dB Default is No If set to Yes device will not use LEC to remove echo from a voice communication Depends on standard phone type and location Select the value according to the local Telco standard where the GXW4248 is installed Please refer to the pull down list to select Settings Early Dial Dial Plan Prefix FIRMWARE 1 0 3 4 Default is No Use
22. UL FCC CE C Tick The GXW4248 gateway includes a small LCD screen for the display of basic information The LCD has a display area of 128x32 pixels which will allow for 2 lines of text with a 16px height limitation per line The LCD menu is showed as Figure 3 LCD Menu Menu is navigated by the Down arrow and OK button FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 15 OF 55 IL GHW4232 V 23 1 192 168 254 37 Critical Events 2 3 Other Events 3 10 Back Main Menu PIN 966000 000B8227F539 Uptime 5d 13h Mode DHCP IP 192 168 254 37 0 TESK 255 255 0 0 Language Test D Test Pattern an venu FIGURE 3 LCD MENU FIRMWARE 1 0 3 4 GXW42XX USER MANUAL Auto Default On Back Factory ndstream Innovative IP Voice amp Video 2012 09 28 12 12PM HW Fan Failure 2012 09 28 12 34 VoIP Running 2012 09 28 12 30PM IP Change etc mx Back Factory PAGE 16 OF 55 UNDERSTANDING GXW VOICE PROMPTS andstream Innovative IP Voice amp Video BASIC OPERATIONS GXW4248 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 001 002 003 004 005 007 010 013 014 015 TABLE 5 DEFINITIONS OF THE GXW VOICE PROMPTS Menu Voice Prompt User s Options Enter fo
23. UN PAGE 31 OF 55 andstream Innovative IP Voice amp Video Number of STUN The Number of STUN response misses allowed before Response Misses restarting DHCP The minimum is 3 misses Allowed Keep Alive Interval Specifies in seconds how often the phone sends a blank UDP packet to the SIP server in order to keep the ping hole on the NAT router to open The default is 20 seconds Upgrade and Provisioning Lock Keypad Update Firmware Upgrade and Provisioning XML Config File Password HTTP HTTPS User Name HTTP HTTPS Password Upgrade via Firmware Server Path Config Server Path Firmware File Prefix Firmware File Postfix Config File Prefix Config File Postfix FIRMWARE 1 0 3 4 If set to Yes the configuration update via keypad is disabled Specifies how firmware upgrading and provisioning request to be sent There are three options to choose from Always Check for New Firmware Check New Firmware only when F W pre suffix changes and Always Skip the Firmware Check 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 Allows users to choose the firmware upgrade method via TFTP HTTP or HTTPS IP address or domain name of firmware serve
24. XTENSION NUMBERS 1 or Dial the number directly and wait for 4 seconds To change the default value modify the following setting No Key Entry Timeout 2 Dialthe number directly and press Use as dial key must be configured in web configuration FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 18 OF 55 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 VoIP service provider for further details on prefix numbers DIRECT IP CALLS Direct IP calling allows two parties that is a FXS Port with an analog phone and another VoIP Device to talk to each other in an ad hoc fashion without a SIP proxy Elements necessary to completing a Direct IP Call 1 Both GXW4248 and other VoIP Device have public IP addresses 2 Both GXW4248 and other VoIP Device are on the same LAN using private IP addresses or 3 Both GXW4248 and other VoIP Device can be connected through a router using public or private IP addresses with necessary port forwarding or DMZ GXWA248 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 047 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
25. a free public STUN Server on the internet and enter it on this field If using Public IP keep this field blank 2 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 GXW4248 supports a broad range of voice codecs Under Profile web pages choose your preferred order of different codecs PCMU A or 27 1 0729 G723 e 3726 16 24 32 40 AAL2 all G726 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 available for the end user are e STATUS Displays the network status account status software version and MAC address of the phone e MAINTENANCE Basic settings such as basic network date and time settings and web telnet FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 29 OF 55 ream Innovative IP Voice amp Video access settings can be set here e PROFILE AUDIO SETTINGS DTMF Vocoder and Analog Line settings can be configured here for each port Additional functions available to administrators are e MAINTENANCE Full settings f
26. anonymous blocking Caller ID Default is No If set to Yes incoming calls with anonymous Caller ID will be rejected with 486 Busy message 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 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 Default is Standard Choose the selection to meet some special requirements from Softswitch vendors Example of vendors CBCOM RNK Distinctive Ringtone FIRMWARE 1 0 3 4 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 selected 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
27. ay 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 FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 21 OF 55 andstream Innovative IP Voice amp Video One additional and popular way to use the Hunting Group feature is called multiplexed analog lines this configuration a legacy PBX system with 8 FXO trunks can be connected to 8 GXW 42xx ports configured as a 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 Configure the SIP account from your VolP 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
28. ays relevant information regarding the individual FXS ports Example NAT Traversal Port Hook SIP Registration DND Forward Busy Forward Delayed Forward FXS1 On Hook Registered No 613 FXS2 Off Hook Registered No 614 FXS3 On Hook Not Registered No FXS4 On Hook Registered Yes 615 FXS TABLE 8 MAINTENANCE Network Settings IP Address Mode Choose how the IP address obtained on the phone Preferred DNS Server Enter the preferred DNS server that should be used for DHCP and PPPoE DHCP Settings Host name Option 12 Specifies the name of the client This field is optional but may be required by Internet Service Providers DHCP Domain Specifies the DHCP Domain This value is optional but may be required by Internet Service Providers Vendor Class ID Option 60 Used by clients and servers to exchange vendor class ID Static IP Settings Configure IP Address Subnet Mask Gateway DNS Server 1 and DNS Server 2 QoS Settings Layer 3 QoS Defines the Layer 3 QoS parameter This value is used for IP Precedence Diff Serv or MPLS Layer 2 QoS Value used for layer 2 VLAN tag Default setting is blank 802 1Q VLAN Tag Layer 2 QoS 802 1p Assigns the priority value of the Layer 2 QoS packets Valid range is Priority Value 0 to 7 STUN Settings Use STUN Yes or No Default is No STUN server The IP address or Domain name of the STUN server Only non FIRMWARE 1 0 3 4 GXWA2XX USER MANUAL symmetric NAT routers work with ST
29. ding 0 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 Only Status and Basic Settings Administrator Level 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 174 and FXS PORTS configuration pages Please reference the GUI pages using following link http www grandstream com products ip voice telephony enterprise analog gateways GXW4248 GUI zip 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 FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 28 OF 55 andstream Innovative IP Voice amp Video Server The following three 3 settings are useful in the STUN Server scenario 1 STUN Server under Advanced Settings webpage Enter a STUN Server IP or FQDN that you may have or look up
30. disable the FXS port Configure an auto dial number when offhook Assign this port to a certain hunting group This is used only when peering with a Grandstream GXW410x Default 1 Supported values are 1 8 meaning line 1 to 8 of the GXW410x device where the will be mapped to This is used when peering with an FXO gateway of any brand You have to specifically mention the IP and sip port where call will be sent to Sip port that will be annexed to the IP address above 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 gateway after saving changes REBOOTING FROM REMOTE Press the Reboot button at the bottom of the configuration menu to reboot the phone remotely The web FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 49 OF 55 andstream Innovative IP Voice amp Video 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 GXW4248 can be automatically configured from a central provisioning system When GXW4248 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
31. e 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 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 1 FXS Port 8 SIP UserID and Authenticate ID left blank Hunting Group set to 2 FXS Port 24 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 groups 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 cal
32. ecification of GXW4248 14 Table 5 Definitions of the GXW Voice 17 Table 6 Call Features Table Star 25 30 Table 8 31 Table 9 Advanced settings eter ete 36 TADE EE Edi 37 Table EEA 49 CONFIGURATION GUI INTERFACE EXAMPLES GXW4248 User Manual http www grandstream com products gxw_series GXW42xx documents GXW42xx_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 o FF U N SCREENSHOT OF STATUS PAGE FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 4 OF 55 andstream Innovative IP Voice amp Video GNU GPL INFORMATION GXW4248 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 ca
33. ed by the phone to synchronize the date and time An extensive list of public NTP servers can be found at http www ntp org Defines the update interval in minutes to obtain the date and time from the server Configures the date time used on the phone according to the specified time zone This parameter allows the users to define their own time zone For syntax and examples please refer to user manual Syslog Server FIRMWARE 1 0 3 4 The IP address or URL of System log server The server collects system log information from the device GXWA2XX USER MANUAL PAGE 35 OF 55 andstream Innovative IP Voice amp Video Syslog Level Select the GXW4248 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 ml e gu go G9 09 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 10 memory exception ERROR level The Syslog uses USER facility addition to standard Syslog payload it contains the following components GS LOG device MAC address error code error message Example May 19 02 40 38 192 168 1 14 GS LOG
34. elease configuration template When a Grandstream device boots up or reboots it will issue a request for a configuration file where is the MAC address of the device i e cfg000b820102ab In addition device will also requests a XML configuration file cfgxxxxxxxxxxxx xml If the download of 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 1 0 3 4 GXW42XX USER MANUAL PAGE 52 OF 55 andstream Innovative IP Voice amp Video 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 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 to Yes Se
35. enterprise GATEWAY GXW4248 OVERVIEW The new GXW4248 has a compact and quiet design and offers superb audio quality rich feature functionality strong security protection and good manageability It is auto configurable remotely manageable and scalable The GXW4248 features 48 port FXS interface for analog telephones dual 10 100 1000Mbps network ports and RJ21 analog port In addition it supports the option of 4 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 40 bit rates iLBC and G 729 SAFETY COMPLIANCE The GXW4248 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 GXW4248 package Use of alternative power adapter may permanently damage the unit WARRANTY Grandstream has a reseller agreement with our reseller customers End users should contact the company from whom the product was purchased for replacement repair or refund FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 7 OF 55 andstream Innovative IP Voice amp Video 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 no
36. eps to set up your gateway FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 9 OF 55 CC sun Innovative IP Voice amp Video GXW4248 LCD light LCD screen Ethernet port Analog port FIGURE 1 DIAGRAM OF GXW4248 PANEL TABLE 1 DEFINITIONS OF THE GXW CONNECTORS Ethernet port Connect to the internal LAN network or router RESET Factory Reset button Press and hold for a while to reset factory default settings DC IN Power adapter connection Analog port Connect to analog phones fax machines with an RJ21 to RJ11 cable FXS ports FXS port to be connected to analog phones fax machines Note Once the GXW4248 is turned on and configured the front display panel indicates the status of the unit TABLE 2 DEFINITIONS OF THE GXW DISPLAY PANEL Act LED Remains ON if plug the network cable LINK LED Indicates Ethernet port activity FXS LED 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 All FXS LEDs slow blinking indicates provisioning FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 10 OF 55 tream Innovative IP Voice amp Video FIGURE 2 50 PIN TELCO CONNECTOR NOTE e last pair of pin 25 and 50 on each RJ21 is not connected e All LED lights display green when ON FIRMWARE 1 0 3 4 GXWA2XX USER MANUAL PAGE 11 OF 55 andstream Innovative IP Voice amp Video e LINK LED and
37. er ID Disable Call Waiting Tone Disable Receiver Offhook Tone Disable Reminder Ring for On Hold Call Disable Visual MWI Visual MWI Type Transfer on Conference Hangup Disable Bellcore style 3 Way Conference FIRMWARE 1 0 3 4 Innovative IP Voice amp Video 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 this key can be included as part of number Default is 4 seconds Call will be completed within this time interval if no additional key entry occurs Determines how many seconds after off hook to wait before autodialing the extension set under Advanced Port Settings The range is 0 to 60 seconds Default is Yes If Yes call features using star codes will be supported locally Default is No If set to YES Call Waiting indication information will not be provided to analog phone connected to this FXS port Default is No If set to YES Call Waiting caller ID will not be provided to analog phone connected to this FXS port Default is No This is to disable the stutter Call Waiting Tone when a Call Waiting call arrives The CWCID will still be displayed Default is No If YES ROH tone will not be played after off hook for 60 seconds Default is No This is to disable the Reminder Ring that is played when a call is waiting on hold and
38. g 086 Voice Mail 099 RESET 700 748 Phone calls between different ports of the same GW4248 Invalid Entry andstream Innovative IP Voice amp Video 9 to toggle between TFTP and HTTP Firmware version information Firmware upgrade mode Enter 9 to rotate among the following three options 1 always check 2 check when pre suffix changes 3 never upgrade Enter the target IP address to make a direct IP call after dial tone See Make a Direct IP Number of voice mails Enter 9 to reboot the device or Enter MAC address to restore factory default setting See Restore Factory Default Setting section GXW4248 support inter port calling from voice menu for easy test verification in factory 700 Ring all ports and connect to first port pick up the call 701 748 Call individual port Automatically returns to Main Menu Five Success Tips when using the Voice Prompt 1 shifts down to the next menu option 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 2 returns to the main menu 3 9 functions as the ENTER key in many cases to confirm an option 4 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 E
39. ging RTP RTCP In audio RFC2833 and or SIP Info SIP RFC 3261 TFTP HTTP HTTPS SRTP IPSEC TLS SIPS HTTPS 802 1x Syslog support HTTP and Telnet access Yes 3 Way conference with local mixing Bellcore Type 1 amp 2 ETSI BT NTT FSK and DTMF based CID Yes IPv4 IPv6 TCP UDP RTP RTCP HTTP HTTPS ARP RARP ICMP DNS DHCP NTP TFTP TELNET PPPoE STUN HARDWARE SPECIFICATION The hardware specifications of the GXW4248 are detailed in Table 4 Telephone Interface FXS LEDs FIRMWARE 1 0 3 4 TABLE 4 HARDWARE SPECIFICATION OF GXW4248 GATEWAY GXW4248 2 RJ21 Ports only 48 GXW42XX USER MANUAL PAGE 14 OF 55 Network interface Power Input LCD screen Telco connector RJ 11 connectors NAND Flash DRAM Function Buttons Environmental Mounting Onhook Voltage Ring Voltage Ring Frequency Short Haul Loop Outdoor Protection Signaling EMC Safety Compliance GXW4248 LCD MENU LAN Single 10 100 1000 BASE TX RJ45 Input 100 240VAC 50 60Hz Output 24V DC 6 25A 128x32 pixel 2 RJ21 50 pins No 64MB 128MB separate 8MB if with a slave processor 1 button for Reset Factory Reset Operation 0 C to 45 C Storage 20 C to 60 C Humidity 10 to 90 Non condensing Desktop and Rack mount Fixed 48V 50Vrms balanced ringing 20 50Hz 2REN Up to 1km on 24 AWG wire Over voltage Protection and surge immunity FXS Loop start EN55022 EN55024 and FCC part15 Class B
40. irmware 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 GXW420x 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 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 LAN is recommended to conduct firmware upgrade 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 FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 51 OF 55 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
41. ll is placed on hold 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 A presses 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 confirm tone Then Acan hang up NOTE Enable Call Feature must be set to Yes in web configuration page Caller A can place a call on hold and wait for one of three situations Press A quick confirmation tone similar to call waiting tone followed by a dial tone This indicates the transfer is successful transferee has received 200 OK from transfer target At this point Caller Acan either hang up or make another call A quick busy tone followed by a restored call on supported platforms only This means the transferee has received a 4xx response for the INVITE and we will try to recover the call The busy tone is just to indicate to the transferor that the transfer has failed Continuous busy tone The phone has timed out Note continuous busy tone does not indicate the transfer has been successful nor does it indicate the transfer has failed It often means there was a failure to receive second NOTIFY check firmware for most recent release FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 20 OF 55 andstream Innovative IP Voice
42. ls 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 standalone 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 FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 23 OF 55 andstream Innovative IP Voice amp Video and 7 plus two extra digits kk For example on the GXW4248 inter port calling is achieved by dialing corresponding to the port number For example the user connected to port 1 can be reached by dialing and 701 the user connected to port 24 can be reached by dialing 724 SENDING AND RECEIVING FAX GXW4248 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 pro
43. makes outbound calls if remote party Timer supports session timer Callee Request If selecting Yes the phone will use session timer when it receives inbound calls with session Timer timer request Force Timer 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 FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 40 OF 55 UAC Specify Refresher UAS Specify Refresher Force INVITE ream Innovative IP Voice amp Video 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 select UAC to use caller or proxy server as the refresher UAS to use the phone as the refresher Session Timer can be refreshed using INVITE method or UPDATE method Select Yes to use INVITE method to refresh the session timer SIP Settings Security Settings Validate Incoming Messages Check SIP User ID for Incoming INVITE Accept Incoming SIP from Proxy Only Authenticate Incoming INVITE Fax Seitings Defines whether the incoming messages will be validated If set to Yes SIP User ID will be checked in the Request URI of the incoming INVITE If it doesn t match the phone s
44. n be downloaded from Grandstream web site from http www grandstream com support faqg qnu_ FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 5 OF 55 andstream Innovative IP Voice amp Video CHANGE LOG This section documents significant changes from previous versions of GXW4248 user manuals Only major new features or major document updates are listed here Minor updates for corrections or editing are not documented here FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 6 OF 55 andstream Innovative IP Voice amp Video WELCOME Thank you for purchasing the Grandstream GXW4248 Analog FXS IP Gateway The GXW4248 offers an easy to manage easy to configure IP communications solution for any business with virtual and or branch locations The GXW4248 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 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 VoIP solution for the small medium business or
45. ndstream Innovative IP Voice amp Video Grandstream Networks Inc Analog IP Gateway GXW4248 GXW42XX User Manual www grandstream com Firmware Version 1 0 0 49 support grandstream com GXW4248 USER MANUAL INDEX GNU CHANGE LOG WELCOME GATEWAY GXW4248 SAFETY COMPLIANCE li ipi ab CONNECT YOUR GXW4248 GATEWAY EQUIPMENT PAGCKAGING CONNECT THE 4248 GXW4248 FEATURES SOFTWARE FEATURES OVERVIEW eese emen HARDWARE SPECIFICATION essent GXW4248 LCD MENU iere cetero esie o ne eei FIGURE 3 LCD MENUBASIC OPERATIONS BASIC OPERATIONS UNDERSTANDING GXW VOICE PLACING A PHONE CALL sse ennemi PHONE OR EXTENSION NUMBERS sse DIRECT IP CALLS Ete ene er OUD E CALC Llc TBANSEE E teca e aded Hou
46. 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 if 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 UserID and Authenticate ID entered Hunting group set to Active FXS Port 2 SIP UserID 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 UserID and Authenticate ID left blank Hunting Group set to 4 FXS Port 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 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 FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 22 OF 55 andstream Innovative IP Voice amp Video 8 is required 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 thos
47. ontain SIP Instance Instance ID ID as defined in IETF SIP Outbound draft Refer To Use Default is No If set to YES then for Attended Transfer the Refer To header uses the Target Contact transferred target s Contact header information SUBSCRIBE for Default is No When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent MWI periodically Enable 100rel Enables the use of PRACK Provisional Acknowledgment method TEL URI The default setting is Disabled If the phone has an assigned PSTN Number this field should be set to UserzPhone then a User Phone parameter will be attached to the From header in SIP request to indicate the E 164 number If server supports TEL URI format then this option needs to be selected SIP Settings Session Timer Session Grandstream implemented SIP Session Timer The session timer extension enables SIP Expiration 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 Min SE The minimum session expiration in seconds The default value is 90 seconds Caller Request If selecting Yes the phone will use session timer when it
48. or network upgrade provisioning TR 069 Security and Syslog e ADVANCED SETTINGS To set advanced Ring Tongs FXO Failover and System Features e PROFILE X To configure each of the SIP accounts FXS PORTS To configure each of the FXS ports and Hunting Groups etc TABLE 7 STATUS System Info Product Model Contains the product model info Part Number Product Part Number Software Version Program This is the main software release This number is always used for firmware upgrade current version is 1 0 3 4 Boot current version is 1 0 3 2 Core current version 1 0 3 3 Base current version is 1 0 3 4 System Up Time Shows system uptime since the last reboot System Time The time according to NTP server Service Status Shows the status of the VOIP applications MAC Address The device ID in hexadecimal format This is needed for Internet Service Provider troubleshooting The 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 IP Address Mode Shows the current IP mode IP Address Shows IP address of GXW4248 Subnet Mask Shows Subnet Mask of GXW4248 Gateway Shows Default Gateway of GXW4248 DNS Server Shows DNS Server of GXW4248 FIRMWARE 1 0 3 4 GXWA2XX USER MANUAL PAGE 30 OF 55 ream Innovative IP Voice amp Video Shows type of NAT the GXW4248 is connected to via its WAN port It is based on STUN protocol Displ
49. 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 Ifthe target IP address is 192 168 0 160 the dialing convention is 47 or Voice Prompt with option 047 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 b If the target IP address port is 192 168 1 20 5062 then the dialing convention would be FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 19 OF 55 andstream Innovative IP Voice amp Video 47 or Voice Prompt with option 047 then 192 168 0 160 5062 followed by pressing the key if it is configured as a send key or wait for 4 seconds 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 ca
50. r 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 Default is blank If configured GXW4248 will request firmware file with the prefix This setting is useful for Internet Telephone Service Providers End users should keep it blank Default is blank This setting is useful for Internet Telephone Service Providers End users should keep it blank Default is blank This setting is useful for Internet Telephone Service Providers End users should keep it blank Default is blank This setting is useful for Internet Telephone Service Providers End users should keep it blank GXW42XX USER MANUAL PAGE 32 OF 55 Option 43 and Option 66 to Override Server Automatic Upgrade Authenticate Conf File Firmware Key Web Telnet Access andstream Innovative IP Voice amp Video If set to Yes configuration and upgrade server s information can be obtained using DHCP option 66 from DHCP server This option specifies the URL of the tftp server 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 Choose Yes to enable automatic upgrade and provisioning
51. r the next menu option Enter a Menu Option DHCP Mode PPPoE Mode or Static IP Mode Address IP address Subnet IP address Gateway IP address DNS Server IP address Preferred Vocoder MAC Address Firmware Server IP Address Configuration Server Address Upgrade Protocol FIRMWARE 1 0 3 4 Enter to return to the main menu Enter 001 005 007 010 013 017 047 086 099 Menu option Enter 9 for confirming a option Enter 9 to toggle the selection If user selects Static IP Mode user need configure all the IP address information through menu 002 to 005 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 Enter 12 digit new IP address if in Static IP Mode Same as Menu option 002 Same as Menu option 002 Same as Menu option 002 Enter 9 to go to the next selection in the list PCMU PCMA iLBC G 726 G 723 G 729 Announces the Mac address of the unit Announces current Firmware Server IP address Enter 12 digit new IP address Announces current Config Server Path IP address Enter 12 digit new IP address Upgrade protocol for firmware and configuration update Enter GXW42XX USER MANUAL PAGE 17 OF 55 016 Firmware Version 017 Firmware Upgrade 047 Direct IP Callin
52. requests to terminate calls when run out of pre paid credit Default is 1812 Specifies the port to be used for the Primary RADIUS Authentication Default is 1813 Specifies the port to be used for the Primary RADIUS Account Specifies the secret string to be used to authenticate the RADIUS connection to the Primary Server It should match RADIUS configuration Set the IP or FQDN of the Secondary RADIUS Server In case Primary Radius server becomes unusable secondary will take role of primary and will manage credit recourses in the network Default is 1812 Specifies the port to be used for the Secondary RADIUS Authentication GXW42XX USER MANUAL PAGE 34 OF 55 RADIUS Account Port Secondary RADIUS Sever Secret RADIUS Timeout RADIUS Retry Download Device Configuration ream Innovative IP Voice amp Video Default is 1813 Specifies the port to be used for the Secondary RADIUS Account Specifies the secret string to be used to authenticate the RADIUS connection to the Secondary Server It should match RADIUS configuration Default is 2 Default is 3 Allows user to download and save a text 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 Date and Time NTP Server NTP Update Interval Time Zone Self Defined Time Zone URI or IP address of the NTP Network Time Protocol server Us
53. rted Disable SIP Disables challenging SIP NOTIFY reboot and resync messages NOTIFY Authentication Disable Voice Disables the voice prompt configuration Default is No If set to Yes accessing integrated Prompt voice menu will be impossible IVR Language Choose English Chinese or Spanish TABLE 10 PROFILES General Settings Profile Active When set to Yes the SIP Profile is activated SIP Server SIP Server s IP address or Domain name provided by VoIP service provider FIRMWARE 1 0 3 4 GXWA2XX USER MANUAL PAGE 37 OF 55 Innovative IP Voice amp Video Failover SIP Failover SIP Server s IP address or Domain name provided by VoIP Service provider This Server server will be used if the Primary SIP server becomes unavailable Prefer Primary SIP Default is No If set to yes it will register to Primary Server if registration with Failover server Server expires Outbound Proxy IP address or Domain name of Outbound Proxy or Media Gateway or Session Border Controller Used by GXW4248 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 Network Settings DNS Mode One from the 3 modes available for DNS Mode configuration A Record for resolving IP Address of target according to domain name SRV DNS resource records indicates how to find services for various protocols NAPTR SRV Naming Authori
54. rvice Provider can use P193 to have the devices periodically check with either Firmware Server or Config Server whenever they are defined This allows the device periodically check whether there is any new changes need to be taken similar to the AntiVirus Software to upgrade the Virus Definition files Screenshot is below Automatic Upgrade 10080 Yes every minutes 60 5256000 1 1 Yes daily at hour 0 23 Yes weekly on day 0 6 FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 53 OF 55 andstream Innovative IP Voice amp Video RESTORE FACTORY DEFAULT SETTING FACTORY RESET 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 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 more than 4 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
55. s are free to end users The configuration tools configuration templates available for download from http www grandstream com support tools FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 50 OF 55 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 7 6 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 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 F
56. t 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 Blind Transfer GXW42XX USER MANUAL PAGE 25 OF 55 andstream Innovative IP Voice amp Video 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 forwarding number followed 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 1 0 3 4 GXW42XX USER MANUAL PAGE 26 OF 55 andstream Innovative IP Voice amp Video CONFIGURATION GUIDE CONFIGURING GXW4248 VIA VOICE PROMPT DHCP MODE Select voice menu option 001 to enable GXW4248 to use DHCP STATIC IP MODE Select voice menu option 001 to enable GXW4248 to use STATIC IP mode then use option
57. tbound calls using the SIP account of port 1 even if Outgoing Call without Registration is set to No Allows the user to specify the time frequency in minutes for the GXW4248 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 Allows the user to specify the time frequency in seconds for the GXW4248 to re register after registration failure The default interval is 20 seconds The maximum interval is 3600 seconds 1 hour Determines how many seconds before the previous registration expires that the port should reregister Defines the local SIP port the GXW4248 will listen and transmit The default value for Profile 1 is 5060 and 6060 for Profile 2 Default is No If set to Yes the device will pick randomly generated SIP ports This is usually necessary when multiple GXW4248 HT50X are behind the same NAT T1 If the network latency is high select larger value for more reliable usage is an estimate of the round trip time between the client and server transactions Maximum retransmission interval for non INVITE requests and INVITE responses GXW42XX USER MANUAL PAGE 39 OF 55 Innovative IP Voice amp Video Remove OBP from Default is No If set to Yes the Outbound Proxy will be removed from the route header Route Header Support SIP Default is Yes If set to Yes the contact header in REGISTER request will c
58. te calls when run out of pre paid credit FIRMWARE 1 0 3 4 GXW42XX USER MANUAL PAGE 24 OF 55 andstream Innovative IP Voice amp Video CALL FEATURES GXW4248 supports the traditional telephony features available in a PBX as well as additional advanced telephony features Key 02 03 16 17 30 31 67 82 47 50 51 69 70 71 72 73 74 78 79 87 FIRMWARE 1 0 3 4 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 r32 0272640 G726 r40 027201 iLBC Disable LEC call Dial 03 number No dial tone is played in the middle Enable SRTP Disable SRTP Block for all config change Send for all config change Block 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 Wai
59. tification 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 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 1 0 3 4 GXW42XX USER MANUAL PAGE 8 OF 55 andstream Innovative IP Voice amp Video CONNECT YOUR GXW4248 GATEWAY Connecting the GXW4248 gateway is easy Before you begin please verify the contents of the GXW4248 package EQUIPMENT PACKAGING Unpack and check all accessories Equipment includes one device unit RJ45 Ethernet cable e one 24V 6 5A universal power adapter e mount CONNECT THE GXW4248 Follow these four 4 steps to connect your GXW4248 gateway to the Internet and access the 5 configuration pages 1 Connect standard touch tone analog phones to the GXW4248 s RJ21 port with RJ11 to R21 cable 2 Insert an RJ45 Ethernet cable into the WAN port of GXW4248 and connect the other end to an uplink port a router or a modem etc 3 Plug the power adapter into the GXW4248 gateway into a power outlet Follow the instructions from the topic Configuring GXW 42XX with Web Browser for initial configuration The GUI pages will guide you through the remaining st
60. ty Pointer according to RFC 2915 Use Configured IP If selected please fill in Primary IP Backup IP 1 and Backup IP 2 One mode can be chosen for the client to look up server The default value is Record Primary IP Configures the primary IP address where the phone sends DNS query to when Use Configured IP is selected for DNS mode Backup IP 1 Configures the backup IP 1 address where the phone sends DNS query to when Use Configured IP is selected for DNS mode Backup IP 2 Configures the backup IP 2 address where the phone sends DNS query to when Use Configured IP is selected for DNS mode NAT Traversal This parameter defines whether the GXW4248 NAT traversal mechanism is activated or not If activated by choosing Yes and a STUN server is also specified then the GXW4248 performs according to the STUN client specification Under this mode the embedded STUN client will detect if and what type of firewall NAT is being used If the detected NAT is a Full Cone Restricted Cone or a Port Restricted Cone the GXW4248 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 GXW4248 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 Use NAT IP The NAT IP address used in SIP SDP messages It should ONLY be used if required by your ITS
61. vider 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 GXW4248 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 server for billing purposes RAIDUS server can send requests to terminate calls when run out of pre paid credit The GXW4248 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 termina

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