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Grandstream GXW4216 User Manual

1. 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 4 and FXS PORTs configuration pages Please reference the GUI pages using the following link http grandstream com products gxw_series gxw42xx 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 Server The following three 3 settings are useful in the STUN Server scenario FIRMWARE 1 0 4 24 GXWA42XX USER MANUAL PAGE 30 OF 57 andstream Innovative IP Voice amp Video 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 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 D
2. 30 31 67 82 47 50 51 69 70 71 72 73 74 78 79 87 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 Disable SRTP Block CallerID for all config change Send CallerID for all config change Block CallerID per call Send CallerID 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 4 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
3. Decoder controlling via voice packet header PAGE 15 OF 57 header DHCP DHCP Client only Server Client 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 802 1P Q tagging Fax over IP QoS DiffServ Transport RTP Protocol DTMF Method In audio and or SIP Info IP Signaling Provisioning Security SRTP TLS SIPS 802 1x Management and Telnet access ETSI BT NTT FSK and DTMF based CID Dial Plan Yes 3 Way 3 Way Conference Caller ID Polarity Yes Reversal Wink Network IPv4 IPv6 TCP UDP Connectivity RTP HTTP HTTPS ARP RARP DNS DHCP FIRMWARE 1 0 4 24 T 38 compliant Group SIP RFC 3261 TFTP HTTP HTTPS Syslog support HTTP with local mixing Bellcore Type 1 amp 2 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 tagging RTP 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 HTTP HTTPS ARP RARP ICMP DNS DHCP NTP DHCP Client only T 38 compliant Group 3 Fax
4. Select the corresponding Profile ID 1 2 3 4 Enable or disable the FXS port Configure an auto dial number when offhook Assign this port to a certain hunting group If configured device will route the incoming call to designated port by request URI user ID in SIP INVITE 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 port 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 Theweb browser will then display a message window to confirm saved changes Grandstream recommends reboot or power cycle the gateway after saving changes FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 51 OF 57 andstream Innovative IP Voice amp Video 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 GXW42xx can be automatically configured from a central provisioning system When GXW4e2xx boots up it wi
5. Voice Detection VAD with CNG comfort noise Activity generation and PLC loss concealment LEC with NLP Packetized Unit supports RTP and packet Voice Protocol AAL2 protocol G 168 compliant Echo Cancellation Dynamic Jitter Buffer Modem detection amp auto switch to G 711 G 711 PLC VAD CNG encoder and decoder G 722 G 723 1A G 726 ADPCM with 16 24 32 40 bit rates Annex Annex II format 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 FIRMWARE 1 0 4 24 Voice Detection VAD with CNG generation and PLC Activity comfort noise loss concealment LEC with NLP Packetized Unit supports RTP and AAL2 protocol G 168 Echo packet Voice Protocol compliant Cancellation Dynamic Jitter Buffer Modem detection amp auto switch to G 711 G 711 PLC VAD CNG encoder and decoder G 722 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 Annex Annex II format 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
6. 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 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 4 24 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
7. Off Hook Auto Dial Delay Enable Call Features Disable Call Waiting Disable Call Waiting Caller 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 4 24 andstream 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 No 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 YESCall 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 h
8. 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 002 IP Address IP address The current WAN IP address is announced Enter 12 digit new IP address if in Static IP Mode 003 Subnet IP address Same as Menu option 002 004 Gateway IP address Same as Menu option 002 005 DNS Server IP address Same as Menu option 002 007 Preferred Vocoder Enter 9 to go to the next selection in the list e PCMU e PCMA e iLBC e G 726 e G 723 e G 722 e G 729 010 MAC Address Announces the Mac address of the unit 013 Firmware Server IP Announces current Firmware Server IP address Enter 12 digit Address new IP address 014 Configuration Server IP Announces current Config Server Path IP address Enter 12 digit Address new IP address FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 19 OF 57 015 016 017 047 086 099 700 748 Upgrade Protocol Firmware Version Firmware Upgrade Direct IP Calling Voice Mail RESET Phone calls between different ports of the same GW42xx Invalid Entry andstream Innovative IP Voice amp Video Upgrade protocol for firmware and configuration update Enter 9 to toggle between TFTP and HTTP Firmware version information Firmware upgrade mode Enter 9 to rotate among the following three opti
9. 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 a 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 4 24 GXW42XX USER MANUAL PAGE 22 OF 57 andstream Innovative IP Voice 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 Athen dials Caller C s number followed by or wait for 4 seconds 3 If Caller C answers the call
10. 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 GXW42XXwill 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 GXW42XXsupports Bellcore style 3 way Conference Instructions for 3 way conference Assuming that call party A and B are in conversation A GXW42XX wants to bring C in a conference A presses FLASH on the analog phone or Hook Flash for old model phones to get a dial tone Adials 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 Apresses FLASH during conference C will be dropped out OO gt PP P gt If A hangs up the conference will be terminated or transfer B to C if Transfer on Conference Hangup set to yes 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
11. Disturb DND When disabled incoming calls will be accepted Blind Transfer FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 27 OF 57 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 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 1 0 4 24 GXW42XX USER MANUAL PAGE 28 OF 57 andstream Innovative IP Voice amp Video CONFIGURATION GUIDE CONFIGURING GXW42XX VIA VOICE PROMPT DHCP Mope Select voice menu option 001 to enable GXW42XXto use DHCP STATIC IP MODE Select voice menu option 001 to enable GXW42XXto use STATIC IP mode then use option 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 GXW42XX to use PPPoEmode 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
12. 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 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 FolP 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 T 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 Preferred DTMF method in listed order Disable DTMF Negotiation DTM
13. LCD display e Added option Prompt Dial Tone Code to configure dial code FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 6 OF 57 andstream Innovative IP Voice amp Video WELCOME Thank you for purchasing the Grandstream GXW42XXAnalog FXSIP Gateway The GXW42XX offers an easy to manage easy to configure IP communications solution for any business with virtual and or branch locations The GXW42XX 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 VoIP technology integrate a traditional phone system into a VoIP network and efficiently manage communication costs This manual will help you learn how to operate and manage your GXW FXS 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 enterprise GATEWAY GXW42XX OVERVIEW The new GXW42XXseries 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 GXWa42XXseriesfeatures 16 24 32 or 48port FXS interface for analog telephones dual 10 1
14. PHONE OR EXTENSION NUMBERS siscctescsiustssccsenessteconsilecestotcaybacsebtensidecatneeisltasethantassdbeeds 20 DIRECT IP OE a nae een aiad 21 vvs 22 EET RE 22 MT cpns ee E E E 22 NTN 22 ENE 23 3 WAY CONFERENCING ad setae styasann bees shaea tetas enema edantiieashan febnetagtanictienaet 23 HUNTING ES ET 23 INTER PORT CALLING EE arrat raters at retten sternar raters a rran nsa arreen att 25 FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 2 OF 57 andstream Innovative IP Voice amp Video SENDING AND RECEIVING Pad rtdetdnikedplsdaesketideta 26 SUPPORT RADIUS PROTOCOL en 26 CALL FEATURES vssssisiseissscisanstsnssncinssnteassdasicanissncenstannaaaiseternsaitianaasnien 27 CONFIGURATION GUIDE visi sesissnisnceacncenstncncsnansacceeasstccansnasnsanssnancsencasasars 29 CONFIGURING GXW42XX VIA VOICE PROMPT sd inddbamd 29 CONFIGURING GXW42XX WITH WEB BROWSER siiscsinceisccsieensiceustacieon basnsacecanensustncesnistenuestndiied 29 ACCESS THE WEB CONFIGURATION MENU u 22 ssesscssssssssssssssssssssssssessssseesseesssnnitsssssnnnee 30 IMPORTANT SETTING EE a bea 30 VNR 30 DTMF METHODS EEE EE EE ER 31 PREFERRED VOCODER CODEC den bkeeegasidetndadeideri el 31 SAVING THE CONFIGURATION CHANGES ssseesssssssssesssessssnseeesesssnntessessssnmeieeeessnnneeesssneeess 51 REBOOTING FROM BEE ende Gdaebantnisnevune 52 CONFIGURATION THROUGH A CENTRAL SERVER sssssesssssssseeeesessssseesesesssneeeeesssenieesessnneness 52 SOFTWARE UPGRADE iwessisc
15. 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 tagging RTP 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 HTTP HTTPS ARP RARP ICMP DNS DHCP NTP GXW42XX USER MANUAL andstream Innovative IP Voice amp Video 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 tagging RTP 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 HTTP HTTPS ARP RARP ICMP DNS DHCP NTP PAGE 16 OF 57 TFTP PPPoE STUN TELNET TFTP TELNET TFTP PPPoE STUN PPPoE STUN HARDWARE SPECIFICATION The hardware specifications of the GXW FXS series are detailed in Table 4 Telephone Interface FXS LEDs Network interface Power Input LCD screen Telco conn
16. 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 4 24 GXW42XX USER MANUAL PAGE 23 OF 57 andstream Innovative IP Voice amp Video 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 42xxports 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 samephonenumber in a round robin fashion Example Configuration of a typical Hunting Group 1 Configure the SIP account from your VoIP Service Provider on FXS port 1underFXS Portswebpage 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 if port 1 is busy or times out You can configure the ring timeout on the Profile page Example configuration of a multipleHuntin
17. is 20 seconds The maximum interval is 3600seconds 1 hour Determines how many seconds before the previous registration expires that the port should reregister Defines the local SIP port the GXW42XX 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 GXW42XX HT50X are behind the same NAT Ti 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 Default is No If set to Yes the Outbound Proxy will be removed from the route header 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 then for Attended Transfer the Refer To header uses the transferred target s Contact header information Default is No When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically Enables the use of PRACK Provisional Acknowledgment method 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 ind
18. 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 GXW42XX USER MANUAL PAGE 50 OF 57 Ring Tones Port Settings Advanced Port Seitings FXO Mapping andstream Innovative IP Voice amp Video Configure ring cadences according to preference SIP User ID Authenticate ID Password Name Profile Enable FXS TR 069 Offhook Auto dial Hunting Group Request URI Routing ID Map to FXO Port Map to FXO Gateway IP And Port 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 subscriber s Authenticate ID used for authentication Can be identical to or different from SIP User ID SIP service subscriber s account password for GXW42XX to register to SIP servers of ITSP Any name to identify this specific user
19. 0 OF 57 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 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 GXW42XXand other VoIP Device have public IP addresses or 2 Both GXW42XXand other VoIP Device are on the same LAN using private IP addresses or 3 Both GXW42XXand other VoIP Device can be connected through a router using public or private IP addresses with necessary port forwarding or DMZ GXW42XXsupports 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 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 1
20. 00 1000Mbps network ports and RJ21analog 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 GXW42XXis 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 GXW42XX package Use of alternative power adapter may permanently damage the unit FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 7 OF 57 andstream Innovative IP Voice amp Video 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 If you purchased the product directly from Grandstream contact your Grandstream Support for an RMA Return Materials Authorization number Grandstream reserves the right to change thewarranty 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 This document contains links to HT502 GUI Interfaces Please download these examples from http www grandstream com products gxw_ser
21. 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 GXW42XX WITH WEB BROWSER The GXWa42XXseries gateway has an embedded Web server that allows users to configure the GXW42XXthrough a web browser It has language support to English Chinese French and Spanish FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 29 OF 57 andstream Innovative IP Voice amp Video ACCESS THE WEB CONFIGURATION MENU The GXW42XXHTML 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 GXW42XX IP Address the GXWA42XXIP Address is the Ethernet IP address for the GXW42XX NOTE 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
22. 68 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 FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 21 OF 57 Andst ream Innovative IP Voice amp Video b Ifthe target IP address port is 192 168 1 20 5062 then the dialing convention would be 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 call 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
23. F Payload Type FIRMWARE 1 0 4 24 The GXWA42XX 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 GXW42XX USER MANUAL PAGE 44 OF 57 Preferred Vocoder Voice Frames per TX 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 SRTP Mode Silence Suppression VAD FIRMWARE 1 0 4 24 andstream Innovative IP Voice amp Video The GXW42XX 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 payloa
24. Info Product Model Part Number Software Version System Up Time System Time Service Status Contains the product model info Product Part Number 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 Shows system uptime since the last reboot The time according to NTP server Shows the status of the VOIP applications Network Status MAC Address IP Address Mode IP Address Subnet Mask Gateway DNS Server FIRMWARE 1 0 4 24 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 Shows the current IP mode Shows IP address of GXW42XX ShowsSubnet Mask of GXW42XX ShowsDefault Gateway of GXW42XX ShowsDNS Server of GXW42XX GXW42XX USER MANUAL PAGE 32 OF 57 andstream Innovative IP Voice amp Video NAT Traversal Shows type of NAT the GXW42XX is connected to via its WAN port It is based on STUN protocol Displays relevant information regarding the individual FXS ports Example Port Hook SIP Registration DND Forward Busy Forward Delayed Forward FXS1 On HookRegistered No 613 FXS2 Off HookRegistered No 614 FXS3 On HookNot Registered No FXS4
25. N 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 GXW42XX 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 STUN with no specified STUN server the GXW42XX 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 The NAT IP address used in SIP SDP messages It should ONLY be used if required by your ITSP 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 Add Auth Header on Initial REGISTER Outgoing Calls Without Registration Register Expiration FIRMWARE 1 0 4 24 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 GXW42XX 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 an Authentication Header with blank nonce will be added in the initial REGISTER Default is No If set to
26. O Failover Failover to FXO Enable or disable the FailoverFXO 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 supported 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 Display Language Choose language for LCD display Prompt Dial Tone Simulates an analog PBX where a code is required to dial an outside line Code TABLE 10 PROFILES General Settings FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 39 OF 57 Profile Active SIP Server Failover SIP Server Prefer Primary SIP Server Outbound Proxy andstream Innovative IP Voice amp Video When set to Yes the SIP Profile is activated 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 B
27. OMMAND Reset default factory settings using the IVR Prompt Table 5 1 Dial for voice prompt Enter O99 and wait for reset voice prompt Enter the encoded MAC address Look below on how to encode MAC address AR oO N 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 4 24 GXW42XX USER MANUAL PAGE 56 OF 57 andstream Innovative IP Voice amp Video A 22 press the 2 key twice A will show on the LCD B 222 C 2222 D 33 press the 3 key twice D will show on the LCD E 333 F 3333 For example if the MAC address is 000b8200e395 it should be keyed in as 0002228200 333395 NOTE 1 Factory Reset will be disabled if the Lock keypad update is set to Yes 2 If 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 Put your PC under the same subnet and then you will be able to open the RECOVERY MODE web page on 192 168 1 234 by your web browser FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 57 OF 57
28. On HookRegistered Yes 615 FXS TABLE 8 MAINTENANCE Network Settings IP Address Mode Choose how the IP address obtained on the phone Preferred DNS Enter the preferred DNS server that should be used for DHCP and PPPoE Server 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 Used by clients and servers to exchange vendor class ID 60 Static IP Settings Configure IP Address Subnet Mask Gateway DNS Server 1 and DNS Server 2 Enable LLDP Configure to enable disable the LLDP Link Layer Discovery Protocol service 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 FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 33 OF 57 andstream Innovative IP Voice amp Video STUN server The IP address or Domain name of the STUN server Only non symmetric NAT routers work with STUN Number of STUN The Number of STUN response misses allowed before Response Misses restarting DHCP The minimum is 3 mis
29. R levels 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 error message Example May 19 02 40 38 192 168 1 14 GS LOG 00 0b 82 00 a1 be 000 Ethernet link is up 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 4 24 GXW42XX USER MANUAL PAGE 38 OF 57 andstream 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 ing ON ms and a pause of OFF ms and then repeat the pattern Dial tone Ringback tone e Busy tone e Reorder tone e Confirmation tone e Call Waiting tone Please refer the document below to determine your local call progress tones http Awww itu int ITU T inr forms files tones 0203 pdf FX
30. TMF 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 GXW42XXsupports a broad range of voice codecs Under Profile web pages choose your preferred order of different codecs e PCMU A or G711u a e G729 e G723 e G726 16 24 32 40 e G722 e iLBC e 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 4 24 GXW42XX USER MANUAL PAGE 31 OF 57 andstream 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 for 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 e FXS PORTS To configure each of the FXS ports and Hunting Groups etc TABLE 7 STATUS System
31. Voice Detection VAD with CNG comfort noise Activity generation and PLC loss concealment LEC with NLP Packetized Unit supports RTP and AAL2 protocol G 168 Echo packet Voice Protocol compliant Cancellation Dynamic Jitter Buffer Modem detection amp auto switch to G 711 G 711 PLC VAD CNG encoder and decoder G 722 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 Annex Annex Il format 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 GXW42XX USER MANUAL andstream Innovative IP Voice amp Video Voice Detection VAD with CNG comfort noise Activity generation and PLC loss concealment LEC with NLP Packetized Unit supports RTP and AAL2 protocol G 168 compliant Echo Cancellation packet Voice Protocol Dynamic Jitter Buffer Modem detection amp auto switch to G 711 G 711 PLC VAD CNG encoder and decoder G 722 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 Annex Annex Il format estimation Voice Play Out unit reordering fixed and adaptive jitter buffer clock synchronization AGC automatic gain control Status output
32. 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 without Registrationis set to No Allows the user to specify the time frequency in minutes for the GXW42XX 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 GXW42XX USER MANUAL PAGE 41 OF 57 SIP Registration Failure Retry Wait Time Reregister Before Expiration Local SIP Port Use Random SIP Port SIP T1 Timeout SIP T2 Timeout Remove OBP from Route Header Support SIP Instance ID Refer To Use Target Contact SUBSCRIBE for MWI Enable 100rel TEL URI Use Request Routing ID in SIP Headers Do Not Escape as 23 in SIP URI FIRMWARE 1 0 4 24 andstream Innovative IP Voice amp Video Allows the user to specify the time frequency in seconds for the GXW42XX to re register after registration failure The default interval
33. anel 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 4 24 GXW42XX USER MANUAL PAGE 11 OF 57 fream Innovative IP Voice amp Video Pin25 __ Pint FIGURE 2 50 PIN TELCO CONNECTOR NOTE e For GXW4216 only first 16 pairs are connected e For GXW4232 first 24 pairs on 15 RJ21 connector and first 8 pairs on the 279 RJ21 connector are FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 12 OF 57 andstream Innovative IP Voice amp Video connected e For GXW4224 and GXW4248 the last pair of pin 25 and 50 on each RJ21 is not connected e All LED lights display green when ON e LINK LED and 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 4 24 GXW42XX USER MANUAL PAGE 13 OF 57 andstream Innovative IP Voice amp Video GXW42XXFEATURES The GXW42XXis a next generation IP voice gateway that is interoperable and compatible with leading IP PBXs Softswitches and SIP platforms The GXW42XXFXS gateway is auto configurable remotely manageable and scalable The GXW42XX gateways come in fourmodel
34. bers i e P2 is associated with Admin Password in the ADVANCED SETTINGS page Fora 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 CIQXXXXXXXXXXXX where XXXXXXXXXXXX is the MAC address of the device i e cf9000b820102ab In addition device will also requests a XML configuration file cfgxxxxxxxxxxxx xml If the download of CEIQXXXXXXXXXXXX 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 1 0 4 24 GXW42XX USER MANUAL PAGE 54 OF 57 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 whenthe 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 Pref
35. d type for iLBC The valid range is between 96 and 127 Default value is100 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 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 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 rfc ric3711 html Default is No VAD allows detecting the absence of audio and conserve bandwidth by preventing the transmission of silent packets over the network GXW42XX USER MANUAL PAGE 45 OF 57 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 Gain Disable Line Echo Canceller LEC SLIC Setting Caller ID Scheme FIRMWARE 1 0 4 24 andstream Innovative IP Voice amp Video 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 t
36. 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 Awww solarwinds com products freetools free_tftp server aspx 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 GXW42XXdevice 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 Aconfiguration 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 num
37. 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 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 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 4 24 The IP address or URL of System log server The server collects system log information from the device GXW42XX USER MANUAL PAGE 37 OF 57 andstream Innovative IP Voice amp Video Syslog Level Select the GXW42XX 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 il Ooo J ef 6 N 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 ERRO
38. ector RJ 11 connectors NAND Flash DRAM Function Buttons Environmental Mounting Onhook Voltage Ring Voltage Ring Frequency Short Haul Loop Outdoor Protection Signaling FIRMWARE 1 0 4 24 TABLE 4 HARDWARE SPECIFICATION OF GXW42XX SERIES GATEWAYS GXW4216 GXW4224 GXW4232 16 RJ11 Ports 1 RJ21 Port 24 RJ11 Ports 1 RJ21 Port 32 RJ11 Ports 2 RJ21 Ports 16 24 32 LAN Single 10 100 1000 BASE TX RJ45 Input 100 240VAC 50 60Hz Output 12V DC 5 0A 4216 24 32 24V DC 6 25A 4248 only 128x32 pixel 1 RJ21 50 pins 1 RJ21 50 pins 2 RJ21 50 pins Yes Yes Yes 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 2km on 24 AWG wire 2REN Up to 2km on 24 AWG wire 2REN Up to 2km on 24 AWG wire Over voltage Protection and surge immunity FXS Loop start GXW42XX USER MANUAL TELNET TFTP PPPoE STUN An Innovative IP Voice amp Video TELNET GXW4248 2 RJ21 Ports only 48 2 RJ21 50 pins No 2REN Up to 2km on 24 AWG wire PAGE 17 OF 57 andstream Innovative IP Voice amp Video EMC EN55022 EN55024 and FCC part15 Class B Safety UL Compliance FCC CE C Tick GXW42XX LCD MENU The GXW42XX gateway series includes a small LCD screen for the display of bas
39. erver 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 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 GXW42XX USER MANUAL PAGE 36 OF 57 Secondary RADIUS Authentication Port Secondary RADIUS Account Port Secondary RADIUS Sever Secret RADIUS Timeout RADIUS Retry Download Device Configuration andstream Innovative IP Voice amp Video Default is 1812 Specifies the port to be used for the Secondary RADIUS Authentication 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
40. et to YES device will send inform packets to the ACS GXW42XX USER MANUAL PAGE 35 OF 57 Periodic Inform Interval Connection Request Username Connection Request Password andstream Innovative IP Voice amp Video 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 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 FIRMWARE 1 0 4 24 The GXW42XX 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 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 GXW42xx 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 s
41. g 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 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 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 UserID 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 required to indicate that the SIP account tied to port marked as Active will be used for all members FIRMWARE 1 0 4 24 GXWA42XX USER MANUAL PAGE 24 OF 57 andstream Innovative IP Voice amp Video 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 Example of not sequential configuration of a multipleHunting G
42. he high requirement e Medium initial 100ms min 20ms max 200ms e Low initial 50ms min 10ms max 100ms Depends on standard phone type and location Select the value according to the local Telco standard where the GXW42XX 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 GXW42XX uses this method for signaling call termination Default is No Default is 200 In 100 10000 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 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 GXW42XX is installed Please refer to the pull down list to select GXW42XX USER MANUAL PAGE 46 OF 57 andstream Innovati
43. ic 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 a at att ww 4 GHW4232 W 2 3 1P 192 168 254 37 etc Critical Events 2 3 Other Events 3 10 Back Main Menu 2012 09 28 12 34PM VolP Running Hardware 2 3A rea 12 30PM Software 1 0 0 33 ange m i TITT etc MAC 000B8227F539 Uptime 5d 13h Bret Mask 255 255 0 0 D Jest Pattern an Tes Auto Default On Back Factory AX Back Factory FIGURE 3 LCD MENU FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 18 OF 57 andstream Innovative IP Voice amp Video BASIC OPERATIONS UNDERSTANDING GXW VOICE PROMPTS GXWA42XX has a built in voice prompt menu for simple device configuration To enter the voice prompt menu press on the standard analog phone which is connected to device s FXS port TABLE 5 DEFINITIONS OF THE GXW VOICE PROMPTS Menu Voice Prompt User s Options Main Menu Enter a Menu Option Enter for the next menu option Enter to return to the main menu Enter 001 005 007 010 013 017 047 086 or 099 Menu option Enter 9 for confirming a option 001 DHCP Mode Enter 9 to toggle the selection PPPoEMode or Static IP If user selects Static IP Mode user need configure all the IP Mode
44. icate the E 164 number If server supports TEL URI format then this option needs to be selected Default is No If set to Yes device will use the configured Request URI Routing ID in the SIP Header This option is usually used under a SIP trunk account s configuration If set to Yes device will use instead of 23 in the send URI GXW42XX USER MANUAL PAGE 42 OF 57 Disable Multiple m Line in SDP Use Privacy Header Use P Preferred Identity Header andstream Innovative IP Voice amp Video Default is No If set to Yes device will send only one m line in SDP regardless how many m field in the incoming SDP If set to Default it will only add Privacy 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 SIP Settings Session Timer Session Expiration Min SE Caller Request Timer Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher Force INVITE Grandstream implemented SIP Session Timer The 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 ses
45. ication 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 GXW42xx 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 1 0 4 24 GXW42XX USER MANUAL PAGE 26 OF 57 andstream Innovative IP Voice amp Video CALL FEATURES GXW42XXsupports the traditional telephony features available in a PBX as well as additional advanced telephony features Key 02 03 16 17
46. icsstciannsancssciaancssunsertaacersaasnasanierimainaienecmens 53 FIRMWARE UPGRADE THROUGH TFTP HTTP HTTPS j ssssseessssssseeseeessssstteessssnmeieessssnnteesetsen 53 INSTRUCTIONS FOR LOCAL FIRMWARE UPGRADE csse ssssssseseesessssssseeeessssseeeeeesssnieeeetsee 54 CONFIGURATION FILE DOWNLOAD oscs cisscisssvadeescprsnasasterachscnsnecsenessdacedbenesiagusnsdbsnssnissnnechicnssiebesetiad 54 FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX ssssesssssssteeeesesssnneeeeessen 55 MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD 0s0 sssssteeessesseeeeeesseen 55 RESTORE FACTORY DEFAULT SETTING anannvnvevennnnnnnnnnnnnnnvneevennennunnne 56 Fl 56 FUN 56 MERE EE EE RE ED EE 56 FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 3 OF 57 andstream Innovative IP Voice amp Video TABLE OF FIGURES GXW42XX User Manual Figure 1 Diagram of GXW4216 24 32 48 Panel umsrrnannvnnonvnnrnrrennnrnnrnnvnnenrvnnnrrnennrnnennnnrrrrenrrrresnrnnenneneennn 11 Figure 2 50 Pin Telco COnMe Ctl c ccccceceeeeeceeeeeeeeeeeaaeceeaeeseaeeecaaeeeeaaeseeeeeseaeeeseaesseaeeseeeeesaeeesaeeeeneeeenees 12 Figur 3 LCD MENU eice cecceceietehint cies enact canis de bebe ese Tegent added eased casts viele kne aati eesti endene mun sadel 18 TABLE OF TABLES GXW42XX User Manual Table 1 Definitions of the GXW Connectors ccecccecceeeeeeeeeeeeee cece eeeeaeeeeneeseeeeesaeeeeeaeseeeeeseeessaeeeeeeseeeees 11 Table 2 Definitions
47. ies gxw42xx documents gxw42xx_qui zip for your reference This document is subject to change without notice The latest electronic version of this user manual is available for download at http www grandstream com products gxw series gxw42xx documents gqxw42xx usermanual english pdf Reproduction or transmittal of the entire or any part in any form or by any means electronic or print for any purpose is not permitted without the express written permission of Grandstream Networks Inc FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 8 OF 57 andstream Innovative IP Voice amp Video CONNECT YOUR GXW42XXGATEWAY Connecting the GXW42XX gateway is easy Before you begin please verify the contents of the GXW42XXpackage EQUIPMENT PACKAGING Unpack and check all accessories Equipment includes e one device unit e one RJ45 Ethernet cable e one 12V 5A universal power adapter 24V 6 25A for GXW4248 e mount CONNECT THE GXW42XX Follow these four 4 steps to connect your GXW42XXgateway to the Internet and access the unit s configuration pages 1 Connect standard touch tone analog phones to the GXW42XX s RJ21 port with a RJ11 to R21 cable 2 Insert anRJ45 Ethernet cable into the WAN port of GXW42XX and connect the other end to an uplink port a router or a modem etc 3 Plugthe power adapter into the GXW42XX gateway into a poweroutlet Follow the instructions from the topic Configuring GXW 42XX with Web Browser fo
48. ix or Postfix MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD When Automatic Upgrade is set to Yes Service 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 d No Yes every minutes 60 5256000 1 1 e Yes daily at hour 0 23 Yes weekly on day 0 6 FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 55 OF 57 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 C
49. ll send TFTP or HTTP HTTPS requests to download configuration files cfg000b82xxxxxx and cfgO0082xxxxxx xml where 000b82xxxxxx is the LAN MAC address of the GXW42xx 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 GXW42xx 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 p
50. llow 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 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 11900x 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 dialing 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 todial 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 4 24 GXW42XX USER MANUAL PAGE 48 OF 57 Use as Dial Key No Key Entry Timeout
51. n 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 Pathis left blank Choose Yes to enable automatic upgrade and provisioning When set to No GXW42XX 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 FIRMWARE 1 0 4 24 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 s
52. ndstream Innovative IP Voice amp Video Grandstream Networks Inc Analog IP Gateway GXW42XX series 16 24 32 or 48 FXS Ports GXW42XX User Manual www grandstream com Firmware Version 1 0 4 24 support grandstream com andstream Innovative IP Voice amp Video GXW42XX USER MANUAL INDEX GNU GPL INFORMATION sississsitsscsnisssacascctiatasinciacenstsasxavissntaastatninssntersis 5 CHANGE LOG sismicicniominimnmutinienninmndsinenmaainn 6 CHANGES FROM 1 0 4 17 USER MANUAL 0 sssseeesssssssseeeeecssssseeesesssnneeieessssnieesesssnmesessssnneees 6 CHANGES FROM 1 0 4 9 USER MANUAL 0 cssssseessssssssseeeseeessnseteeceesssnieeteesssnseteecesssnnmceeseaneeeet 6 CHANGES FROM 1 0 4 4 USER MANUAL 0 sccssssseeescssssssseeeeessssseteeceesssnmeeseesssnueniecesssnnmieseessaneeeet 6 CHANGES FROM 1 0 3 9 USER MANUAL 0 scsssseeecsessssseeesecesssseteeceesssnmeiteesssnuettecessssnietesssaneeneet 6 EE ee 7 GATEWAY GXW42XX EEE san duiiatiystaieeddetuntndeesldaysia 7 NNN 7 NNN 8 CONNECT YOUR GXW42XXGATEWAY revnnnnnvnvvvevenennnnnnnnnnnnnnvnvevevenennnnnn 9 EQUIPMENT PACKAGING eee AGudardgnne 9 EE GN 9 GXWA2XXFEATURES Luse baksana aenor aeaaeae aiaa En Enaria 14 SOFTWARE FEATURES OVERVIEW EE AR 14 HARDWARE SE CICA IOI EE 17 PENN 18 BASIC OPERATIONS weanicscnsesescsatecncracrcenacacncesanenesesasnancenteanndenscsamnanaaasesess 19 UNDERSTANDING GXW VOICE PRONE TE vea 19 PLACING A PHONE CALL isuncenoneiaininenen ian aa 20
53. of the GXW Display Panel rasnronannvnnnnvvnrnrrnnnnrnnnnnvnnerrnnnnrrvnnnrnnrnnnnrrrreesnrnnennensenneeenn 11 Table 3 GXW42XXSoftware Features mmumsrmmrmmmnemnmmmmnmnemnmnnnemnnninsen 14 Table 4 Hardware Specification of GXW42XX series gateways ccsceceeeeeceee esses eeeeeeseeeeetaeeeeeeeeeeeees 17 Table 5 Definitions of the GXW Voice Prompts ccccccececeeeeeeeeeeeeeeeeeeeeeseeeee sae eeeaaeeeeeeeseeeesaeeeeaeeeeeeeess 19 Table 6 Call Features Table Star Code ccccccecececeeeeeeneeeeeeee eee eeeaaeeeeaee sense saaeeeeaaeseeeeeseaeeesaeeeeeeeeeeeees 27 Table SETE EE EE dgstes RATER 32 Table 8 Maintenance eric iccecssescsisats ccczevedonsenandennenvaesudunaiy dcvsivasindinandacaanvae snd unatn daunavasyndanaeddanstvartndanet add de dun 33 Table 9 Advanced settings stuer dusker tieccesasecccecds eden br ieie aei aa ANEA EAE NERSE NERS EER 38 Table 1 0 ad SE EE 39 Table 11 FXS Pors seorsan nan E KENESE OTER A EREEREER E EEEE 51 FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 4 OF 57 andstream Innovative IP Voice amp Video GNU GPL INFORMATION GXWa42xx 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 ht
54. ons 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 Calf 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 GXW42xxX 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 701 732 Call individual port Automatically returns to Main Menu Five Success Tips when using the Voice Prompt 1 cow 2 returns to the main menu shifts down to the next menu option 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 Dial the number directly and wait for 4 seconds To change the default value modify the following setting No Key Entry Timeout or 2 Dialthe number directly and press Use as dial key must be configured in web configuration FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 2
55. ook for 60 seconds Default is No This is to disable the Reminder Ring that is played when a call is waiting on hold and 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 GXW42XX 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 49 OF 57 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
56. order Controller Used by GXW42XX 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 Primary IP Backup IP 1 Backup IP 2 FIRMWARE 1 0 4 24 One from the 3 modes 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 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 A Record Configures the primary IP address where the phone sends DNS query to when Use Configured IP is selected for DNS mode Configures the backup IP 1 address where the phone sends DNS query to when Use Configured IP is selected for DNS mode Configures the backup IP 2 address where the phone sends DNS query to when Use Configured IP is selected for DNS mode GXW42XX USER MANUAL PAGE 40 OF 57 NAT Traversal Use NAT IP Proxy Require andstream Innovative IP Voice amp Video This parameter defines whether the GXW42XX NAT traversal mechanism is activated or not If activated by choosing STUN and a STUN server is also specified then the GXWA42XX performs according to the STU
57. r initial configuration The GUI pages will guide you through the remaining steps to set up your gateway FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 9 OF 57 fream Innovative IP Voice amp Video GXW4216 LCD screen FXS ports LCD light Ethernet port Analog port GXW4224 FXS ports LCD light LCD screen Ethernet port Analog port GXW4232 Ethernet port FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 10 OF 57 andstream Innovative IP Voice amp Video GXW4248 LCD light LCD screen Ethernet port Analog port FIGURE 1 DIAGRAM OF GXW4216 24 32 48 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 GXW42XX 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 p
58. randstream 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 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 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 FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 53 OF 57 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
59. roup FXS Port 1 SIP UserlD 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 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 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 bu
60. rovides 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 1 0 4 24 GXW42XX USER MANUAL PAGE 52 OF 57 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 4 24 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 athttp www grandstream com support firmware Currently the HTTP firmware server URL is firmware g
61. s the GXW4216 GXW4224 GXW4232and GXW4248 each offering superb voice quality traditional telephony functionality easy deployment and 16 24 32 and 48FXS 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 16 24 32 or 48FXS ports no front panel FXS ports on GXW4248 e RJ 45 Ethernet ports e 4 configurable SIP profiles e Supports Voice Codecs G711 a u Annex I amp Il G723 1A G726 ADPCM with 16 24 32 40 bit rates G729 A B iLBC T 38 Fax e Comprehensive Dial Plan support for Outgoing calls e G 168 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 GXW42XXSOFTWARE FEATURES GXW4216 GXW4224 GXW4232 GXW4248 Telephone 16 FXS ports 24 FXS ports 32 FXS ports 48 FXS ports Interfaces SIP 16 SIP accounts 4 24 SIP accounts 4 32 SIP accounts 4 48 SIP accounts 4 Provisioning profiles profiles profiles profiles Network 10 100 1000 Mbps 10 100 1000 Mbps 10 100 1000 Mbps 10 100 1000 Mbps Interface RJ 45 RJ 45 RJ 45 RJ 45 Number of 16 Concurrent Calls 24 Concurrent Calls 32 Concurrent Calls 48 Concurrent Calls FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 14 OF 57 Concurrent Calls except when using SRTP Voice over Packet Capabilities Voice Compression
62. ses 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 Always send HTTP Basic Authentication Information Upgrade via Firmware Server Path Config Server Path Firmware File Prefix Firmware File Postfix FIRMWARE 1 0 4 24 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 Default is No If set to Yes device will send configured user name and password within HTTP request before server sends authentication challenge Allows users to choose the firmware upgrade method via TFTP HTTP or HTTPS IP address or domain name of firmware server That URL of the server that hos
63. sion 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 timer SIP Settings Security Settings FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 43 OF 57 Validate Incoming Messages Check SIP User ID for Incoming INVITE Accept Incoming SIP from Proxy Only Authenticate Incoming INVITE Fax Seitings andstream Innovative IP Voice amp Video Defines whether the incoming messages will be validated
64. sy 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 Groupsand 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 kkk For example on the GXW42xx inter port calling is achieved by dialing and 7 plus two extra digits KKK corresponding to the port number For example the user connected to port 1 can be reached by dialing FIRMWARE 1 0 4 24 GXWA42XX USER MANUAL PAGE 25 OF 57 andstream Innovative IP Voice amp Video and 701 the user connected to port 24 can be reached by dialing 724 SENDING AND RECEIVING FAX GXWA42XX 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 asfax 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 GXW42XX supports RADIUS for authent
65. tp wWww grandstream com support faq gnu _gpl FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 5 OF 57 andstream Innovative IP Voice amp Video CHANGE LOG This section documents significant changes from previous versions of GXW42xx 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 4 17 USER MANUAL e Added support for audio codec G 722 e Added option Always send HTTP Basic Authentication Information to enable sending HTTP authentication without server challenge e Added the options to enable disable Use P Preferred Identity Header and Use Privacy Header e Added the options to enable disable Do Not Escape as 23 in SIP URI CHANGES FROM 1 0 4 9 USER MANUAL e Added option Use Request Routing ID in SIP Headers to enable disable device to use configured Request URI Routing ID for certain FXS port in its SIP message Header when a trunk SIP account is used e Added option Add Auth Header on Initial REGISTER to enable disable including authentication header in the initial REGISTER CHANGES FROM 1 0 4 4 USER MANUAL e Added option Enable LLDP to enable disable LLDP e Added field Request URI Routing ID to allow device to route the calls to individual fxs ports based on the DID when only have one sip registration CHANGES FROM 1 0 3 9 USER MANUAL e Added option Display Language to choose language for
66. ts 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 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 GXW42XX USER MANUAL PAGE 34 OF 57 Config File Prefix Config File Postfix Allow DHCP Option 43 and Option 66 to Override Server Automatic Upgrade Authenticate Conf File Firmware Key Web Telnet Access andstream Innovative IP Voice amp Video 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 s information can be obtained using DHCP option 66 from DHCP server This optio
67. ve IP Voice amp Video Call Settings Early Dial 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 Dial Plan Prefix Sets the prefix added to each dialed number FIRMWARE 1 0 4 24 GXW42XX USER MANUAL PAGE 47 OF 57 An 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 xx at least 2 digits number xx at least 2 digits number a b c exclude d 8 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 11 161 7xxxxxxx A

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