User Manual for GXW4104/GXW4108
Contents
1. sae Te Dundee a ERR RUE 32 RESTORE FACTORY 5 00 00 33 EXAMPLES GXW410X ARN cua 34 GXW410x User Manual www grandstream com Firmware Version 1 3 1 6 support grandstream com TABLE FIGURES GXW410X USER MANUAL FIGURE 1 DIAGRAM OF GXW410X BACK nnn nnne 6 FIGURE 2 DIAGRAM OF GXW410X DISPLAY nennen 7 FIGURE 7 SCREEN SHOT OF VIDEO SURVEILLANCE nnne 30 APPLICATION ONE GXW CONNECTED WITH AN IP PBX OR 34 APPLICATION TWO GXW TO EXTEND A TRADITIONAL PBX 34 APPLICATION THREE GXW CONNECTED WITH AN IP PBX OR SIP SERVER AND VIDEO SURVEILLANCE ns ae cc cece dea ceca se DIM ots ond 35 APPLICATION FOUR USING A GXW FOR PURE IP IP COMMUNICATION CONFIGURATION 36 INDEX OF TABLES GXW410X USER MANUAL TABLE 1 DEFINITIONS OF THE GXW 652. 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 UAC Specify As a Caller select UAC to use the phone as the refresher or UAS to use the Callee or proxy Refresher server as the refresher UAS Specify As a Callee select UAC to use caller or proxy server as the refresher or UAS to use the Refresher phone as the refresher Force INVITE Session Timer can be refreshed using INVITE method or UPDATE method Select Yes to use INVITE method to refresh the session timer Enable 100rel The use of the PRACK Provisional Acknowledgement method enables reliability to be offered to SIP provisional responses 1xx series This is very important if PSTN inter networking is to be supported A user s request to use reliable provisional responses is invoked by the 100rel tag which is appended to the value of the required header of initial signalling messages Refer to uses 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 INVITE Ring no In case incoming call has arrived from PSTN to VoIP and INVITE message was generated by answer Timeout GXW device the
3. 4 4 0 6 TABLE 2 DEFINITIONS OF THE GXW DISPLAY PANELL cccssecsessseeeesseesenseeeeesneeeeesssensenseeeenees 7 TABLE 4 HARDWARE SPECIFICATION OF GXW41OX 11 TABLE 3 GXW410X SOFTWARE FEATURES 12 TABLE 6 WEB LOG IN DEFINITIONS BASIC SETTINGS 14 TABLE 7 STATUS PAGE DEFINITIONS 5 sac oae ence tee ce dace cet eene Che ases eset ene ce eee 15 TABLE 8 ADVANCED CONFIGURATION PAGE nnns 17 TABLE 9 FXO LINES CONFIGURATION 5 2 4 20 TABLE 10 FXO EINE TEST TAB BEFINITIONS roo nue eoo tpe ne Enn ku pen 23 TABLE 11 CHANNELS PAGE 5 4 0 24 TABLE 12 CHANNELS PAGE DEFINITIONS siirsin iaa aaan aeai a a 26 TABLE 13 PROFILE PAGE DEFINITIONS cricca 2 3 cece see a a aaan ed 26 GXW410x User Manual www grandstream com Firmware Version 1 3 1 6 support grandstream com GUI INTERFACES GXW410X USER MANUAL http www grandstream com support gxw series gxw410x documents gxw410x gui zip SCREENSHOT OF ADVANCED SETTINGS CONFIGURATION PAGE SCREENSHOT or Basic SETTINGS CONFIGURATION PAGE SCREENSHOT OF CHANNELS CONFIGURATION PAGE SCREENSHOT OF FXO
4. DNS Server 2 secondary fields This option specifies the name of the client This field is optional but may be required by some Internet Service Providers Default is blank This option specifies the domain name that client should use when resolving hostnames via the Domain Name System Default is blank Used by clients and servers to exchange vendor specific information Default is Grandstream GXW440x PPPoE user name Necessary if ISP requires you to use a PPPoE Point to Point Protocol over Ethernet connection PPPoE account password This field is optional If your ISP uses a service name for the connection enter the service name here Default is blank This field will let the user enter a preferred DNS server to be used instead of the one acquired by the service provider Controls how the date time is displayed according to the specified time zone Default is No If set to Yes time zone settings will originate from the DHCP server Daylight Savings Time This parameter controls whether the displayed time will be daylight savings time or not If set to Yes and the Optional Rule is empty then the displayed time will be 1 hour ahead of normal time The Automatic Daylight Saving Time Rule shall have the following syntax start time end time saving Both start time and end time have the same syntax Month day weekday hour minute month 1 2 3 12 for Jan Feb Dec day 1 2 3
5. This tool will run an automated test to determine the correct impedance value to match your lines Wait for Dial tone Stage Method Min Delay before Dial PSTN Unconditional Call Forward to VOIP Number of Rings Before Pickup Default is No When set to Yes the gateway will recognize dial tone from the Central Office CO before it completes a call If you can t make an outbound call set this is Yes if this is set to Yes make sure you have configured the dial tone settings correctly in the Channels tab and that there is not any major noise interference in the line Syntax ch1 8 1 all channels 1 to 8 are set to value 1 or 2 Stage method can be set to either 1 or 2 Set this parameter to 1 if you need to make a direct PSTN call from a VOIP endpoint When you set it to 2 you will first dial one of the VOIP channel accounts from the VOIP endpoint this will result in getting a PSTN line dial tone to then dial out the destination PSTN number Most implementations require this setting to be configured to 1 Default is 500ms This needs to be equal to or greater than the Current Disconnect threshold setting Once the threshold is reached the gateway can dial out This parameter should only be used if there are PSTN line detection issues This is an extremely important setting to make sure incoming PSTN calls are picked up and forwarded to the correct VOIP destination User ID This parameter allows users to configure a Use
6. Be m e Qu s B9 Dy Please check with your PSTN service provider or traditional PBX specs for which caller ID scheme they it support If you are not sure about which to use please refer to Table 10 FXO Lines Test Tab Definition This tool will run an automated test to determine the proper Caller ID Scheme so the gateway can properly detect the Caller ID Similarly to the cases explained above we can specify a caller ID scheme for each channel independently Default is relay via From header You may also select relay via P Asserted Identity header Disable Caller ID feature will be disabled Send anonymous All calls forwarded to VOIP end will be sent as anonymous This setting allows you to make several options related to facsimile You can select the method T 38 or Pass through G711 You can select the fax transmission rates 2400 4800 7200 9600 12000 14400bps You can enable or disable ECM Error Checking Mode 10 FXO LINE TEST TAB DEFINITIONS Line AC Impedance CPT Detection External Number External Call Timeout Note The user can only test the parameters for only one of the PSTN lines at the same time In all cases please enter the telephone numbers as if the lines were to dial each other locally For the AC Impedance Test we only need to select the line to be tested by clicking on the AC impedance box corresponding to that line telephone numbers are optional Remember that the AC
7. and what type of firewall NAT it is sitting behind through communication with the specified STUN server If the detected NAT is a Full Cone Restricted Cone or a Port Restricted Cone the GXW410x will attempt to 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 GXW410x will periodically every 20 seconds or so send a blank UDP packet with no payload data to the SIP server to keep the hole on the NAT open SIP Extension to notify SIP server that the unit is behind the NAT Firewall Default is No Use only if proxy supports 484 response Grandstream implemented SIP Session Timer The session timer extension enables SIP sessions to be periodically refreshed via a SIP request UPDATE or re INVITE Once the session interval expires if there is no refresh via a UPDATE or re INVITE message the session will be terminated Session Expiration is the time in seconds at which the session is considered timed out if no successful session refresh transaction occurs beforehand The default value is 180 seconds The minimum session expiration in seconds The default value is 90 seconds If selecting Yes the phone will use session timer when it makes outbound calls if remote party supports session timer If selecting Yes the phone will use session timer when it receives inbound calls with session timer request
8. impedance test is usually used to reduce the echo that might be present in the line For the CPT test call progress tones we will test current disconnect as well You will need 2 telephone numbers to perform the test You can only perform the test on one line row at the same time and it will be the one that has the box checked for testing This tested line will use another line connected to the gateway to perform the test by calling into it this is why you will have to enter the telephone number for a second line to help with the test For CID detection you will need 2 telephone numbers to perform the test You can only perform the test on one line row at the same time and it will be the one that has the box checked for testing This tested line will use another line connected to the gateway to perform the test by calling into it this is why you will have to enter the telephone number for a second line to help with the test To perform the test please select the line you want to test and the desired test to be performed Enter the information for this line as well as a second line if necessary Then click on the update button and then reboot Log back in and now you should see the information for the line selected as well as the check box already marked already there Go ahead and start the test now please wait a few minutes until the test is done Notes It is not required to enter a telephone number when testing for impedance as the
9. signal not very reliable with low bit rate codec via RTP RFC2833 or via SIP INFO Multiple DTMF transmission schemes can be selected e 1 in audio e 2 2833 e 3 in audio and RFC2833 e 4 SIP Info e 5 and RFC2833 e 6 SIP Info and RFC2833 1 7 in audio RFC2833 and SIP Info Default is 4 seconds Default is ch1 8 5060 The indicates increments by 2 so port 1 is set at 5060 port at 5062 and so on This setting can be used with Round Robin and or Flexible setting below to configure different ports to be placed under different Round Robin groups Default is disabled for all ports The user can select to either enable it but not force it or force it on an individual port basis When used the communication will be sent using Secure RTP Default is rr 1 8 The syntax is pretty straight forward here The rr stands for Round Robin and the numbers stand for the ports that belong to that round robin group For example rr 1 8 gt Round robin within the first 8 ports i e outgoing calls will be forwarded to the next available port within the group of ports 1 to 8 rr 1 3 6 8 rr 2 7 gt Round robin within port 1 3 4 5 6 and 8 Second round robin group within ports 2 and 7 i e outgoing calls to ports 1 3 4 5 6 and 8 will be forwarded to the next available port within this group ONLY Outgoing calls to port 2 and 7 will be forwarded to the next available port between ports 2 and 7 ONLY I
10. system does not place any actual calls for the test If you log into the Web Interface while the test is running will not interrupt the process Enter the telephone PSTN number that corresponds to this line Enter it as if you were going to dial it locally Select this box if you want to test for impedance on the line that is on the same row as the checked box Remember that you can only check one item at the same time Select this box if you want to test for call progress tones and current disconnect threshold on the line that is on the same row as the checked box Remember that you can only check one item at the same time Enter an external telephone PSTN number to be used as an auxiliary number for the test This is used only if we do not have at least 2 PSTN lines connected to the gateway This is only used for CPT and current disconnect threshold testing In order to use this function you will have to monitor the Syslog output This is only reserved for very advanced users This is the time the GXW will wait for the external telephone number to pick up during the test Apply test results automatically Apply test results to all ports Error Timeout Default is No If selected on Yes then all the results from the test will be applied automatically you select No you will have to monitor the Syslog output This is only reserved for very advanced users Default is No If selected on Yes then all the test results will be a
11. 31 weekday 1 2 3 7 for Mon Tue Sun or 0 which means the daylight saving rule is not based on week days but based on the day of the month hour hour 0 23 minute minute 0 59 If weekday is 0 it means the date to start or end daylight saving is at exactly the given date In that case the day value must not be negative If weekday is not zero and day is positive then the daylight saving starts on the first day th iteration of the weekday 1st Sunday 3rd Tuesday etc If weekday us not zero and day is negative then the daylight saving starts on the last day th iteration of the weekday last Sunday 3rd last Tuesday etc The saving is in the unit of minutes The saving time may also be preceded by a negative sign if subtraction is desired instead of addition The default value for Automatic Daylight Saving Time Rule shall be set to 03 11 0 02 00 11 04 0 02 00 60 which is the rule for US Examples US where daylight saving time is applicable 03 11 0 02 00 11 4 0 02 00 60 This means the daylight saving time starts from 11 March at 2AM and ends November 4th at 2AM The saving is 60 minutes 1hour You may also access the Device Status page which provides details of the GXW product The Device Status page terms are defined in Table 7 Status Page Definitions Taste 7 Status Pace DEFINITIONS Hardware Revision MAC Address IP Address Product Model Software Versio
12. D 8 LEDs GREEN Universal Switching Input 100 240V AC 50 60Hz 0 5A Max Power Adapter Output 12V DC 1 25A UL certified Dimension 225mm L x 172mm W x 42mm H Weight 0 29 Ibs 3 5 oz Temperature 32 104 F 0 40 C Humidity 10 90 non condensing Compliance FCC CE Taste 3 GXW410x Sortware Features IP settings Telephone Interface Network Interface LED Indicators On Off Switch Voice over Packet Capabilities Voice Compression Video Surveillance DHCP Server Client Fax over IP QoS IP Transport PSTN Signaling DTMF Method IP Signaling Provisioning Media Control Management Short and long haul Caller ID Polarity Reversal Wink EMC Safety GXW410x Analog Gateway Series GXW4104 4 ports 4 SIP accounts w choice of 3 SIP Server profiles GXW4108 8 ports 8 SIP accounts w choice of 3 SIP Server profiles Round robin port scheduling to ensure available lines to access PSTN networks FXO RJ11 Two 2 10 100 Mbps RJ45 Power Video and Line LEDs Yes G 168 compliant Echo Cancellation Dynamic Jitter Buffer Modem detection amp auto switch to G 711 G 711U G711A G 723 G 729A B GSM Real time H 264 base CIF resolution Switch Mode and PPPoE T 38 compliant Group 3 Fax Relay up to 14 4kpbs and auto switch to G 711 for Fax Pass through Diffserve TOS 802 1 P Q VLAN tagging RTP RTCP and RTSP FXO Loop start Current Disconnect Flexible DTMF transmission method User interface
13. Lines CONFIGURATION PAGE SCREENSHOT OF PROFILE 1 ConFIGURATION PAGE 5 SCREENSHOT STATUS CONFIGURATION PAGE GXW410x User Manual www grandstream com Firmware Version 1 3 1 6 support grandstream com WELCOME Thank you for purchasing the Grandstream GXW410x IP Analog FXO Gateway The GXW410x is a cost effective easy to use and easy to configure IP communications solution for any business The GXW410x gateway supports popular voice codecs and is designed for full SIP compatibility and interoperability with 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 FXO Analog IP Gateway and make the best use of its many upgraded features including simple and quick installation multi party conferencing etc 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 Enable the video surveillance port to give you peace of mind while you are away from your business Gateway GXW410x Overview GXW410x offers an easy to manage feature rich IP communications solution for any small business or businesses with virtual and or branch locations who want to leverage their broadband network and or add new IP Technology t
14. Path with the IP address of the PC 6 Update the change and reboot the unit e You can also download the free HTTP server from http httod apache org or just use Microsoft IIS web Restore Factory DeFAuLt SETTING WARNING Restoring the Factory Default Setting will DELETE all configuration information of the phone Please BACKUP or PRINT out all the settings 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 VoIP service provider The ONLY way to restore default factory settings is as follows 1 2 3 4 Unplug the Ethernet cable Locate the needle sized hole on the back panel of the gateway next to the Power connection Enter a thin object in this hole and keep it pressed for about 7 seconds You will see the Network port LEDs green and orange go off and on simultaneously this indicates the reset went through All settings have been erased and the gateway is back to factory settings Examples of GXW410x Configurations AppPLication OnE GXW connecten witH AN IP PBX or SIP Server Scenario A business with a traditional phone system with or without broadband access and an PBX or SIP Servers connecting to an Internet Telephone Service Provider ITSP Anywhere in the world IPPBX or Phones PSTN Analog Lines AppLication Two GXW ExrEND A TrapitionaL PBX Scenario Scenario a small business wit
15. SDP message Default is blank STUN Server Firmware Upgrade amp Provisioning Via TFTP Server Via HTTP Server Firmware Server Path Config Server Path Firmware File Prefix Firmware File Postfix Config File Prefix Config File Postfix Allow DHCP Option 66 to override server Automatic Upgrade Authenticate Conf File IP address or Domain name of the STUN Simple Traversal of UDP through NATs server This radio button will enable GXW410x to download firmware or configuration file through either or If selected the GXW410x will attempt to retrieve new configuration or new code image from the specified TFTP server at boot time It will make up to 5 attempts before timeout and then it will start the boot process using the existing code image in the Flash memory If a server is configured and a new code image is retrieved the new downloaded image will be verified and then saved into the Flash memory Note Please do NOT interrupt the TFTP upgrade process especially the power supply as this will damage the device Depending on the network environment this process can take up to 25 or 30 minutes The URL for the HTTP server used for firmware upgrade and configuration via HTTP For example http provisioning mycompany com 6688 Grandstream Here 6688 is the specific TCP port that the HTTP server is listening at it can be omitted if using default port 80 Note If Auto U
16. TP fails for any reason e g TFTP server is not responding there are no code image files available for upgrade or checksum test fails etc the GXW410x gateway will stop the TFTP process and simply boot using the existing code image in the flash Firmware upgrade process may take as long as 20 minutes over the Internet or just 20 seconds if it is performed on a LAN Grandstream recommends conducting firmware upgrades in a controlled LAN environment if possible Downtoap TFTP Server For users who do not have a local TFTP server Grandstream provides a NAT friendly TFTP server on the public Internet for users to download the latest firmware upgrade automatically Please check the Services section of Grandstream s Web site to obtain this TFTP server IP address Alternatively user can download and install a free TFTP or HTTP server in his LAN for a firmware upgrade A free Windows version TFTP server can be downloaded from http support solarwinds net updates New customerFree cfm Directions for Downloading TFTP Server 1 Unzip the file and put all of the files under the root directory of the TFTP server 2 Putthe PC running the TFTP server and the GXW410x in the same LAN segment 3 Go to File gt Configure gt Security to change the TFTP server s default setting from Receive Only to Transmit Only for the firmware upgrade 4 Start the TFTP server in the phone s web configuration page Configure the Firmware Server
17. are files are located For example The user can use the following URL in the Firmware Server Path firmware mycompany com 6688 Grandstream 1 3 1 6 where firmware mycompany com is the FQDN of the HTTP server It can also be in IP address format 6688 is the TCP port the HTTP server listening to default http server listens to port 80 Grandstream 1 3 1 6 is the RELATIVE directory to the root dir on HTTP web server UPGRADE THROUGH TFTP To upgrade firmware via TFTP set the field Firmware Upgrade and Provisioning Upgrade Via to TFTP The TFTP server can be configured in either IP address format or FQDN To configure the TFTP server via the Web configuration interface follow these five steps 1 Open your browser to input the IP address of the GXW41 0x 2 Enter the admin password to enter the configuration screen 3 Enter the TFTP server address or URL in the Firmware Server Path field near the bottom of the configuration screen 4 Once the Firmware Server Path is set update the change by clicking the Update button 5 Reboot or power cycle the unit If the configured upgrade server is found and a new code image is available the GXW41 0x will retrieve the new image files by downloading them into the GXW410 gateway s SRAM During this stage the GXW410x gateway s LED will blink until the checking downloading process is completed Upon verification of checksum the new code image will be saved into the Flash If TF
18. ateways are able to locate each other i e they should be on the same LAN or have public IP addresses GXW400x amp GXW410x Scenario Tott Free Carline Between Locations Branch A Boston MA 4 employees GXW 400x Branch B Denver CO 4 employees If you setup resembles the one in this diagram you need to configure the SIP Server field to be the IP address of the other gateway i e configure the IP address of GXW400x gateway to be the SIP Server for the GXW410x gateway and vice versa Please be sure you set SIP Registration to No Catt Frow analog Phone connected to the GXW400x gateway picks up and dials the destination PSTN number The call gets routed to the GXW410x gateway which dials out the digit string onto the FXO Lines thus reaching the destination PSTN endpoint On the reverse incoming calls from the PSTN endpoints connected to the GXW410x gateway will be routed automatically to the analog endpoints connected to the GXW400x gateway FXS FXO Gateway EXAMPLE GXW400x Gateway GXW410x Gateway Profile 1 Advanced Seitings SIP Server Set it to IP Address of STUN Server Blank GXW410x Use Random Port No SIP Registration No Outgoing Call without Registration Yes NAT traversal No Advanced Settings FXO lines STUN Server Blank Wait for Dial Tone Y or N whichever works for your PSTN Service Provider Stage Method 1 Unconditional Call F
19. call will be disconnected after pre configured timeout if not answered by VoIP extension Preferred Vocoder The GXW410x supports up to 5 different Vocoder types including G 711 A U law GSM G 723 1 G 729A B The user can configure Vocoders in a preference list that will be included with the same preference order in SDP message The first Vocoder in this list can be entered by choosing the appropriate option in Choice 1 Similarly the last Vocoder in this list can be entered by choosing the appropriate option in Choice 8 Special Feature Default is Standard Choose the selection to meet some special requirements from Soft Switch vendors like Nortel Broadsoft etc SAVING THE CONFIGURATION CHANGES When changes are made press the Update button in the Configuration Menu The GXW410x will display the following screen to confirm that the changes have been saved To activate changes reboot or power cycle the GXW410x after all changes are made FROM REMOTE The administrator can remotely reboot the unit by pressing the Reboot button at the bottom of the configuration menu The following screen will indicate that rebooting is underway The user can re login to the unit after waiting for about 30 seconds ViDEO SURVEILLANCE GXW410x gateway be used with an Analog Surveillance CCD Camera to perform video surveillance function This application should be used in a LAN environment or when bot
20. configuration 8 ConFiGuRATION Pace DEFINITIONS Admin Password G723 Rate Voice Frames per Tx Layer 3 QoS Layer 2 QoS Local RTP port Use Random Port Keep alive interval Use NAT IP Administrator password Only the administrator can configure the Advanced Settings page Password field is purposely left blank for security reasons The maximum password length is 25 characters G723 encoding rate 6 3kbps or 5 3kbps This field contains the number of voice frames to be transmitted in a single packet When setting this value the user should be aware of the requested packet time used in SDP message as a result of configuring this parameter This parameter is associated with the first vocoder in the above vocoder Preference List or the actual used payload type negotiated between the 2 conversation parties at run time For example if the first vocoder is configured as G723 and the Voice Frames per TX is set to be 2 then the ptime value in the SDP message of an INVITE request will be 60ms because each G723 voice frame contains 30ms of audio Similarly if this field is set to be 2 and if the first vocoder chosen is G729 or G711 or G726 then the ptime value in the SDP message of an INVITE request will be 20ms If the configured voice frames per TX exceeds the maximum allowed value the gateway will use and save the maximum allowed value for the corresponding first vocoder c
21. der uses call progress tones then it should be set to Yes in order to realize disconnect tone Please configure accurate Call Progress Tones on Channels web page based on PSTN provider or traditional PBX settings If you are not sure if this option should be enabled or what Call Progress Tones are required please refer to Table 10 FXO Lines Test Tab Definition This tool will run an automated test to determine the proper PSTN configuration that the gateway should have to work with your Service Provider or analog PBX Default is No This should be set to Yes only if the FXO lines are subscribed to PR service from PSTN Service provider It is merely a PR detect feature Note If there is no PR service from provider on the FXO line and this setting is configured to Yes calls will not be successful Terminate call after long silence detected Default is 60 seconds max 65536 Selects the impedance of the analog line connected to the FXO port on the GXW410x Here is some basic information which may be helpful for initial configuration 600 Ohm North America 270 Ohm 750 Ohm 150 nF Most of Europe 220 Ohm 820 Ohm 120 nF Australia New Zealand 220 Ohm 820 Ohm 115 nF Austria Bulgaria Germany Slovakia South Africa 370 Ohm 620 Ohm 310 nF UK India If this parameter is not configured properly you may experience echo or noise in the line Please refer to Table 10 FXO Lines Test Tab Definition
22. e SIP Server Under the Channels page you will need to fill in the information like SIP User ID Password etc Now when calls are made from an IP phone the call will be routed to the SIP Server which will forward it to one of the SIP accounts on the GXW410x which will then forward it to the PSTN line Without SIP accounts In this case you simply have to configure the SIP Server to perform forwarding of the SIP INVITE message with the FXO destination number to the gateway s IP Address The GXW410x gateway will receive the digits and immediately forward them on the FXO lines to the destination PSTN Most of the configuration on the Gateway for this case will remain default except Stage Method needs to be set to 1 and SIP Server IP Address DNS name has to be filled Functional Diacram IP PBX amp GXW410x Anywhere in the world IPPBX or eem Phones PSTN Analog endpoints For incoming calls from the PSTN analog endpoints to the GXW410x gateway the device will auto forward each call to a configured IP extension The SIP Server can then route the call based on its own configuration or IVR system FXS Gateway with GXW410x No SIP Server required Alternatively the GXW410x gateway can be used without a SIP Server It can be used in conjunction with an FXS Gateway Ex GXW400x and calls can still be originated from the IP network and terminated on the PSTN network and vice versa All you need to make sure is that both g
23. figuration e ADMIN PASSWORD is the device s web configuration password for admin DEVICE IP ADDRESS is the device IP DEVICE RTSP PORT is the RTSP port setting of the device If the port uses default value 554 the port portion can be omitted from the URL C Click OK to start the video Figure 7 SCREEN SHOT OF VIDEO SURVEILLANCE 5 media player File View Settings Audio video Navigation Help NW 0 01 22 0 00 00 51 00 Irtsp fadmin 123456 amp 72 72 74 219 2 client side running VLC as monitoring station Firmware UPGRADE Our latest official release be downloaded from http www grandstream com firmware html Firmware or software upgrades can be done either via TFTP or HTTP The corresponding configuration settings are on the configuration page End users should NOT modify the configuration settings that are useful for ITSPs To upgrade your unit firmware follow these steps 1 Under Advanced Settings web page enter your TFTP or HTTP Server IP address or FQDN next to the Firmware Upgrade Upgrade Server field 2 Select via TFTP or HTTP accordingly 3 If you plan to use Automatic Upgrade set it to Yes otherwise No this will make it check for upgrade every time you reboot UPGRADE THROUGH To upgrade firmware via HTTP the field Firmware Upgrade and Provisioning Upgrade Via needs to be set to HTTP The Firmware Server Path should be set to where the firmw
24. h a login screen There are two default passwords for the login page End user Level Only Status and Basic Settings Administrator Level All pages can be browsed After login the following page is the Basic Configuration page explained detail in Table 6 Web Log in Definition Taste 6 Wes Loc in Derinitions Basic Pace Web Access Web Port End user Password IP Address DHCP hostname DHCP domain DHCP vendor class ID PPPoE account ID PPPoE password PPPoE Service Name Preferred DNS server Time Zone Allow DHCP Option 2 to override Time Zone Settings Select HTTP or secure HTTPS protocol for Web Access By default HTTP uses port 80 and HTTPS uses port 443 This field is for customizable web port This contains the password to access the End user Web Configuration Menu Status and Basic Settings This field is case sensitive with a maximum length of 25 characters There are two modes to operate the GXW410x DHCP mode all the field values for the Static IP mode are not used even though they are still saved in the Flash memory The GXW410x acquires its IP address from the first DHCP server it discovers from the LAN it is connected Using the PPPoE feature set the PPPoE account settings The GXWA10x will establish a session if any of the fields is set Static IP mode configure the IP address Subnet Mask Default Router IP address DNS Server 1 primary
25. h sides have public IP address ViDEO SURVEILLANCE PROCEDURES gt Gateway side 1 In the ADVANCED SETTING page find the following field and change from default setting NO to YES reboot the device Enable Video Surveillance No Yes 2 Connect an analog surveillance camera to the VIDEOIN connection at the back panel of the unit gt PC side Monitor Device 1 Download VLC from http Awww videolan org vic This is the only player so far that supports RFC 3984 2 Launch VLC 3 Go to Preferences gt Input Codecs gt Demuxers gt H264 check Advanced options in the bottom The option Frames per Second will show Change that value to 5 and then save 4 Go to Preferences gt Input Codecs gt Access modules gt Real RTSP check Advanced options in the bottom The option Caching value ms will show Change that value to 1000 and then save You may change it to a smaller value to reduce the delay 5 f the viewer is under NAT go to Preferences gt Demuxers gt Access modules gt RTP RTSP check Advanced options in the bottom The option Use RTP over RTSP TCP will show Check that option box Grandstream does NOT recommend this network environment 6 Close the Preferences window and go to gt Network Stream a Select RTSP as the protocol b Enter the URL in the format of rtsp admin ADMIN PASSWORD QDEVICE ADDRESS DEVICE RTSP PORT Change the blue text according to your con
26. h traditional analog PBX lines and broadband access who want to extend their traditional PBX to virtually anywhere in the world using the Internet Any SIP End point such as Grandstream BugeTone HandyTone GXP 2000 or GXV 3000 are needed in this scenario Boston 6 employees Grandstream le NUS BE N Phones i Anywhere Dem ee uc 1 AN SIP the world Any branch endpoint anywhere APPLICATION THREE GXW connecten witH AN IP PBX SIP Server ano VIDEO SuRVEILLANCE Scenario The GXW410x offers an additional video surveillance port which can be configured for surveillance It is the only small business analog gateway that offers this security feature Branch A Boston MA 6 employees IPPBX or SIR Server IPPBX or SIP Server Grandstream IP A A Phones Grandstream IP Phones Branch B Denver 4 employees Anywhere in the world Apptication Four Usinc A GXW ron Pure IP Communication CONFIGURATION Scenario Four The GXW410x offers an IP to IP pure IP Communications System configuration where all locations use IP phones Branch A Boston MA 6 employees IPPBX or SIP Server IP LAN IPPBX or SIP Server A amp Grandstream IP Phones Grandstream IP Phones Branch Denver CO E p 4 employees Anywhere in the world
27. hoice The maximum value for PCM is 10 x10ms frames for 2726 it is 20 x10ms frames for G723 it is 32 x30ms frames for G729 G728 64 x10ms and 64 x2 5ms frames respectively This field defines the layer 3 QoS parameter which can be the value used for IP Precedence or Diff Serv or MPLS Default value is 48 Its range lies from 0 to 63 This contains the value used for layer 2 VLAN tag 802 1q VLAN tag Default value is 0 Range lies from 0 to 4095 802 1p Priority value Default value is 0 Range lies from 0 to 7 The above 2 settings need to be supported on the network and then configured accordingly on the GXW410x Incorrect configuration will cause blocked access which will result in Factory Reset as the only option to renew access This parameter defines the local RTP RTCP port pair the GXW410x will listen and transmit It is the base RTP port for channel 0 When configured channel 0 will use this port value for RTP and the port 1 for its RTCP channel 1 will use port value 2 for and port value 3 for its RTCP and so on The default value is 5004 This parameter when set to Yes will force random generation of both the local SIP and RTP ports This is usually necessary when multiple GXW410xs are behind the same NAT This parameter specifies how often the GXW410x sends a blank UDP packet to the SIP server in order to keep the hole on the NAT open Default is 20 seconds NAT IP address used in SIP
28. hout notice The latest electronic version of this user manual is available on our website http www grandstream com support gxw_series gxw410x documents gxw410x usermanual english pdf Reproduction or transmittal of the entire or in form or by any means electronic or print for any purpose without the express written permission of Grandstream Networks Inc is not permitted PACKAGING Unpack and check all accessories Equipment included in the package 1 One GXW410x Unit 2 One universal power adapter 3 One Ethernet cable THE GXW410x Figure 1 Diagram GXW410x Back Panet GXW410x ance LAN WAN 45 Switch FXO Parts Efhemet Ports VIDEO IN Jack TaarE 1 THE GXW Connectors LAN or PC Connect your PC to this port It will then be assigned an IP address from your Router DHCP Server The GXW410x acts as a switch only WAN Connect to the internal LAN network or Public Internet VIDEO IN Connection for Analog based Video Surveillance Camera RCA RESET Factory Reset button Press for 7 seconds to reset factory default settings POWER IN Power adapter connection OFF ON Off On switch FXO1 FXO8 FXO ports to be connected to physical PSTN lines from a traditional PSTN PBX or PSTN Central Office Figure 2 DiacRAM GXW410x PANEL GXW410x TaarE 2 or THE GXW DispLAv PANEL Power LED I
29. ined in Table 9 FXO Lines Configuration Definitions An example of the Channel Dialing Configuration is shown in Figure 6 Please note the default is always configured The user has the option to change the default settings as described in the Table 9 Taste 9 FXO Lines Conricuration DEFINITIONS Enable Current Disconnect Enable Tone Disconnect Enable Polarity Reversal Silence Timeout AC Termination Impedance Syntax for Channel Specific Settings Default 1 8 all channels 1 to 8 are set to use value X Additional Examples ch1 3 6 10 ch2 7 8 12 channels 1 3 4 5 and 6 are set to use value 10 while channels 2 7 and 8 are set to use value 12 When set to Y Current Disconnect is enabled Certain PSTN COs require this to be enabled order to realize disconnect signal from PSTN side Default is Y If enabled use threshold Default is 100ms Range is 40ms to 800ms Certain PSTN service providers have a threshold time within which the line stabilizes after off hook It is entirely dependent on provider however if you experience PSTN line detection issues please modify this setting appropriately in 100ms increments If you are not sure if this option should be enabled please refer to Table 10 FXO Lines Test Tab Definition This tool will run an automated test to determine the proper PSTN configuration that the gateway should have to work with your Service Provider or analog PBX Default is No If PSTN provi
30. n System Up Time Registered Hardware version number Main Board Interface Board The device ID in HEX format This is a very important ID for ISP troubleshooting This field shows WAN IP address of GXW410x This field contains the product model info GXW4104 or GXW4108 Program This is the main software release Boot and Loader are not changed often This field shows system up time since the last reboot This field indicates whether the different SIP Accounts configured under Channels page are successfully registered to the SIP server s FXO Line Connected This field will give the status of each physical FXO Line connected to the Gateway It will update the status regularly Yes Connected and Idle Busy Connected and Busy No Not connected Additionally it will also provide real time Caller ID information of Incoming as well as Outgoing calls PPPoE Link Up This field shows whether the PPPoE connection is running if connected to DSL modem Detected NAT Type This field shows what kind NAT the GXW410x is connected to via its WAN port It is based on STUN protocol User SETTINGS ApvANcED User CONFIGURATION The end user needs to login to the advanced user configuration page the same way as for the basic configuration page Advanced User configuration includes the end user configuration and advanced configurations including SIP configuration Codec selection NAT Traversal Setting and other miscellaneous
31. n order to terminate a call on FXO port 2 or 7 you will need to change its Local SIP Listen port accordingly Default is 99 Syntax to USE this feature prefix that is 99 ch could be anything from 1 to 8 dialing will result in this call forwarded to FXO port ch immediately Taste 12 Pace DEFINITIONS Dial Plan The Dial Plan feature implemented is applicable for VOIP to PSTN calls only You may configure a dial plan based on the following grammar 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 digit number xx at least 2 digit number exclude 3 5 any digit of 3 4 or 5 147 any digit 1 4 or 7 2 011 replace digit 2 with 011 when dialing WARNING illegal input will fall back to default Example 1 369 11 1617xxxxxxx Allow 311 611 911 and any 10 digit numbers of leading digits 1617 Example 2 1900 lt 1617 gt xxxxxxx Block any number of leading digits 1900 and add prefix 1617 for any dialed 7 digit numbers Example 3 1 2 9 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 Default PSTN Outgoing x Note If you do not plan to use this feature set to default x Hook flash Duration Default 600m
32. ndicates Power Remains ON when Power is connected and unit is turned ON Ready LED Remains ON after boot up LAN LED Indicates LAN or WAN port activity PC LED Indicates PC or LAN port activity Video LED Remains solid green on boot up If Video IN terminal is connected indicates video activity LEDs 1 8 Indicate status of the respective FXO Ports on the back panel Busy ON Available OFF NOTE All LEDs display green when ON The Ready light will only be ON when the network interface is ready and the Web User Interface is accessible During a firmware upgrade or configuration download the following LED pattern will be observed Power Ready Video and WAN LEDs will be ON The FXO port LED will keep flashing during download and then stay OFF while the new files are written The entire process may take between 20 to 30 minutes The firmware upgrade is complete when you can login into the web configuration pages APPLICATION DESCRIPTION A IP PBX Server with GXW410x A SIP proxy server such as Asterisk or a SIP registrar server can be deployed with the GXW410x series gateway In this environment the SIP server handles SIP registration and call control and the GXW410x processes media conversion between IP and PSTN calls There are 2 ways to configure GXW410x when using it with a SIP Server 1 With SIP accounts configured on Channels page In this case the GXW410x gateway acts as an endpoint requesting registration from th
33. ndstream Innovative IP Voice amp Video Grandstream Networks Inc Analog IP Gateway GXW410x 4 or 8 FXO Ports w Video Surveillance GXW410x User Manual www grandstream com Firmware Version 1 3 1 6 support grandstream com TABLE CONTENTS GXW410X USER MANUAL WELGOME eee eee PY 5 G teway GXWAIT Ox UA 5 Safety Compliances eee sessi eeee resins etn 5 WATT me I 5 PACKAGING 3 inert de rote aee Luder ore Md 6 CONNECTING THE tiem ein aee ot a eec o dite bte peut vites deco que d 6 APPLICATION DESCRIPTION mil iu S a AED M TEE 8 Functional Diagram of IP PBX amp GX W410OKX ccccsscessecsseeensececeeesecesceessccescecacecaeeeneeceeeecseceeacesecesaeceeneeeeeeeenaeeenens amp GXW400x amp GXW410x Scenario Toll Free Calling Between Locations eese enne 9 FXS and FXO Gateway Configuration Example 10 FEATUBES SE 10 SOFTWARE FEATURES 2 2 eene sense senes eise iiis ie tete esas sese setis TEE aeter retenir nnns anu 10 HARDWARE SPECIFICATION 2 11 CONFIGURATION GUIDE iace
34. o their current phone system The Grandstream enterprise analog VoIP Gateway GXW410x series converts SIP RTP IP calls to traditional PSTN calls and vice versa There are two models the GXW4104 and GXW4108 which feature 4 or 8 FXO ports respectively The installation is the same for either model Sarety CoMPLIANCES The GXW410x 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 GXW410x package Using an alternative power adapter may permanently damage the unit WARRANTY Grandstream has a reseller agreement with our reseller customer should contact the company from whom you purchased the product for replacement repair or refund If you purchased the product directly from Grandstream contact your Grandstream Sales and Service Representative for a RMA Return Materials Authorization number Grandstream reserves the right to remedy warranty policy without prior notification Caution Changes or modifications to this product not expressly approved by Grandstream or operation of this product in any way other than as detailed by this User Manual could void your manufacturer warranty e This document contains links to Grandstream GUI Interfaces Please download the GUI examples http www grandstream com support gxw series gxw410x documents gxw410x gui zip for your reference e This document is subject to change wit
35. of In audio RFC2833 and SIP Info SIP RFC 3261 TFTP and HTTP SRTP TLS and SIPS pending Syslog support HTTPS and telnet pending remote management using Web browser RENS Up to150 ft on 24 AWG line Bellcore Type 1 amp 2 ETSI BT NTT and DTMF based CID Yes Detection only The PSTN lines will need to be subscribed to PR service from the Service Provider GXW410x EN55022 Class B CFR Part 15 Class B 55024 GXW4104 FCC CE in addition GXW410x EN60950 1 GXW4108 UL60950 1 in addition CONFIGURATION GUIDE ConFIGURATION wirH WEB Browser The GXW410x gateways have an embedded Web server that will allow a user to configure the IP phone through any web browser Examples of GUI interfaces can be downloaded http www grandstream com support gxw series gxw410x documents gxw410x gui zi THE Wes Conricuration Menu 1 Navigate your browser to http www grandstream com DOWNLOAD IPQuery IPQuery zip 2 Run Grandstream IPQuery tool that you just downloaded 3 Click on button in order to begin device detection 4 The detected devices will appear in the Output field Grandstream IP Query M NIC Selection Interface Network adapter Broadcom 440x 10 100 Integrated Controller on loc Output MAC Address IP Address DOOBS20CD710 192 168 22 104 END USER CONFIGURATION When the Web Configuration Menu is accessed the GXW410x will respond wit
36. orward to VOIP ch1 8 444 ch1 8 p1 ch1 8 5060 Channels 1 8 5060 Profile 1 Local SIP Listen port For VOIP to PSTN calls 5060 Profile 1 SIP Server Set it to IP Address of GXW400x SIP Registration No NAT traversal No For more information regarding this setup email Grandstream technical support at support grandstream com or visit our User and Developer forum at forums grandstream com FEATURES The GX W410x is a next generation IP voice and video gateway that features full interoperability with leading IP PBXs Soft switches SIP platforms The GXW410x series gateways offer superb voice and video quality traditional telephony functionality simple configuration feature rich functionality and an additional video port that enables the gateway to act as a video surveillance gateway Sortware Features Overview 4 and 8 FXO port media gateways Video surveillance port External power supply Two RJ 45 ports switched or routed TFTP and HTTP firmware upgrade support Multiple SIP accounts multiple SIP profiles choice of 3 profiles per account Supports Audio Codecs G711U A G723 G729A B and GSM Supports Video Codecs H 264 G 168 echo cancellation Flexible DTMF transmission In Audio RFC2833 SIP Info or any combination of the 3 Selectable multiple LBR coders per channel T 38 compliant HanpwanE SPECIFICATION 4 Harpware SpeciFication GXW410x LAN interface 2xRJ45 10 100Mbps LE
37. pgrade is set to No GXW410Xx will only do HTTP download once at boot up IP address or domain name of firmware server IP address or domain name of configuration server Default is blank If configured GXW400X will request firmware file with the prefix This setting is useful for ITSPs End user should keep it blank Default is blank End user should keep it blank Default is blank End user should keep it blank Default is blank End user should keep it blank Default value is No If set to Yes configuration file will originate from the DHCP server Choose Yes to enable automatic upgrade and provisioning In Check for new firmware every field enter the number of minutes to enable GXWA410x to check the server for firmware upgrade or configuration When set to No GXW410x will only do upgrade once at boot up Other options are Always check for New Firmware Check New Firmware only when F W pre suffix changes Always skip the Firmware check If set to Yes configuration file is authenticated before acceptance This protects the configuration from an unauthorized change DTMF Payload Type Syslog Server Syslog Level NTP server Allow DHCP Option 42 to override an NTP server Enable Video Surveillance RTSP Port This parameter sets the payload type for DTMF using RFC2833 Default is 101 The IP address or URL of System log server This feature is especially useful for ITSP Internet Telephone Service P
38. pplied to all ports on the gateway If all the lines belong to the same service provider or PBX it will make sense to apply the results to all ports This is the time the gateway will wait to exit the test mode when something unexpected or an error has occurred Taste 11 Pace DEFINITIONS Channels SIP User ID Authentication ID Authentication Password Profile ID Call Progress Tones Channel Voice Settings Tx to PSTN Audio Gain dB Rx from PSTN Audio Gain dB Silence Suppression Note The channels here are basically SIP endpoints that will act as clients registering to the SIP Server configured under the appropriate Profile page It should be set same as the channel number i e 1 2 4 or 8 depending on number of FXO ports t is NOT the same as SIP Account ID This is the SIP account information Enter the SIP User ID part of the account SIP service subscribers Authenticate ID used for authentication It can be identical to or different from SIP User ID SIP account password needs to be entered here Note After entering the password it will show up as blank but the password still remains active Select the corresponding Profile ID 1 2 3 Profiles are SIP Server configurations Using these settings user can configure tone frequencies according to user preference default the tones are set to North American frequencies Frequencies should be configured with known values to avoid
39. r ID or extension number to be automatically dialed upon FXO line off hook SIP Server You also need to specify the Profile of the user id configured above p1 stands for Profile 1 p2 stands for Profile 2 and so on SIP Destination Port Along with the user id and Profile you also have the option to choose the destination port where you would like to send the call By default it should be set to ch1 x 5060 x can be 4 or 8 depending on number of ports We can also specify a different destination for each port For example under User ID we can type in ch1 104 ch2 227 ch3 5 501 ch6 7 856 Under Sip Server we can type in ch1 p1 ch2 4 p2 ch5 p3 Under Sip Destination Port we can type in ch1 2 5060 ch2 7080 ch3 8 5066 Default is 4 This is the number of rings the gateway will wait to send the call to the VOIP side in case the Caller ID has yet to be detected If there s CID information the call will be sent right away If your lines don t have the CID service set this to 1 Caller ID Scheme Caller ID Transport type T 38 Setting The GXW410x supports 5 different types of schemes 1 Bellcore US standard ETSI FSK during ringing ETSI FSK prior to ringing with DTAS ETSI FSK prior to ringing with LR ETSI FSK prior to ringing with PR ETSI DTMF during ringing ETSI DTMF prior to ringing with DTAS ETSI DTMF prior to ringing with PR ETSI DTMF prior to ringing with PR 0 SIN 227 BT 1 NTT Japanese standard c 53 50
40. rovider Select the ATA to report the log level Default is NONE The level is one of DEBUG INFO WARNING or ERROR Syslog messages are sent based on the following events 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 9 SLIC chip exception WARNING and ERROR 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 99 m e c B9 D c Example May 19 02 40 38 192 168 1 14 GS LOG 00 0b 82 00 a1 be 000 Ethernet link is up URI or IP address of the NTP Network Time Protocol server which will be used by the phone to synchronize the date and time Default value is No If set to Yes the NTP server will originate from the DHCP server When set to Yes GXW410x will start converting video feed received from analog camera to packets In order to view this video feed please follow instructions given under Video Surveillance chapter on page 23 By default it is 554 ConFIGURING THE FXO CHANNELS Configuring the FXO channels on the GXW410x gateway is an easy process Follow the GUI interfaces The Device Status page terms are def
41. s If symmetric NAT is detected STUN will not work and ONLY outbound proxy can correct the problem Default is No If set to Yes the client will use DNS SRV to look up server If the GXW410x has an assigned PSTN telephone number this field should be set to Yes Otherwise set it to No If Yes is set a user phone parameter will be attached to the From header in SIP request This parameter controls whether the GXW410x needs to send REGISTER messages to the SIP Server The default setting is Yes Default is No If set to yes the SIP user s registration information will be cleared on reboot This parameter allows the user to specify the time frequency in minutes for the GXW410x to refresh its registration with the specified registrar The default interval is 60 minutes or 1 hour The maximum interval is 65535 minutes about 45 days This parameter is most used by Service Providers It prevents message REGISTER overload of SIP Server in case of downtime due to maintenance or power failure By increasing interval length common message load is decreased Interval range is 1 3600 seconds This parameter defines whether the GXW410x NAT traversal mechanism will be activated or not If activated by choosing Yes and a STUN server is also specified then the GXW410x will behave according to the STUN client specification Under this mode the embedded STUN client inside the GXW410x will attempt to detect if
42. s This value can accept any value in the 100 2000ms range X10ms Use DTMF Parameter Default Yes No means to use DTMF parameter settings according to DTMF Digit from RFC2833 or SIP Info Length DTMF Digit Volume and DTMF Dial Pause DTMF Digit Length Default value is 100ms Please not that the value will be multiplied by 10ms DTMF Digit Volume Default value is 11dB DTMF Dial Pause Default value is 100ms Please note that the value will be multiplied by 10ms PROFILES Profiles are basically IP PBX SIP Server configuration templates If you have more than one IP PBX system or SIP Server that you would like to use with the GXW410x then you can configure Profile 2 or 3 Note Make sure you select the correct profile for each channel under Channels pace Tase 13 Prorite Pace DEFINITIONS Activate Profile When set to Yes the SIP Profile is activated Profile Name A name to identify a Profile SIP Server Outbound Proxy Use DNS SRV User ID is Phone Number SIP Registration Unregister on Reboot Register Expiration SIP Registration Failure Retry Wait Time NAT Traversal Proxy Require Early Dial Session Expiration Min SE Caller Request Timer Callee Request Timer SIP Server s IP address or Domain name provided by VoIP service provider IP address or Domain name of Outbound Proxy or Media Gateway or Session Border Controller Used by GXW410x for firewall or NAT penetration in different network environment
43. s ete eco Eruca adeste so teu reme id aa aere orte eue Fest etm ee 13 CONFIGURATION WITH WEB EE vdaces dees Rest Fee een ee Eee Eee Eden 13 Accessing the Web Configuration Menu aee 13 End user Conftguration 1 5 eta esteso lo ade ede reae e Ead S 13 ADVANCED USER S BETINGS i550 eate cre RAE ER REY Feo eben ela eo epe E ENERO RR NER 17 Advanced User Configuration deii i beali eode Ye eed bee Pee aede aede 17 CONFIGURING THE FXO CHANNELS RET 20 26 Saving the Configuration Changes 28 Rebo ting from Remote itte cies e 28 VIDEO SURVEILLANCE scii etie iei E isc i 29 VIDEO SURVEILLANCE PROCEDURES lt 5 ertet ede tr EE Ee 29 FIRMWARE UPGHADE iie MEI UM M I M ML I E 31 lJPGRADE THROUGH HI TD crt Db EC E eR RES 31 31 Download
44. uncomfortable high pitch sounds ON is the period of ringing ON time in ms while OFF is the period of silence In order to set a continuous ring OFF should be zero Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern e Dial tone e Ringback tone e Busy Re order tone e Confirmation 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 or run the FXO Line Test Table 10 Channel voice settings mentioned below Allows user to set a value in dB for transmission to PSTN Audio Gain Default is 1 Range is from 12 12dB Allows user to set a value in dB for receive from PSTN Audio Gain Default is 0 Range is from 12 12dB This controls the silence suppression VAD feature of G723 and G729 If set to Yes when a silence is detected small quantity of VAD packets instead of audio packets will be sent during the period of no talking If set to No this feature is disabled Echo Cancellation Channel specific Setting DTMF Method No Key Entry Timeout Local SIP Listen Port SRTP Mode Round Robin and or Flexible Prefix to specify Port 1 stage dialing method When set to Y Echo cancellation is enabled Channel specific settings mentioned below This parameter specifies the mechanism to transmit DTMF digits There are7 modes supported in audio which means DTMF is combined in audio
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