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Gate104 User Manual

1. 192 168 2 1 DHCP IP Lease Time 120 Hows Default is 120 hours or 5 days LAN Subnet Sets the LAN subnet mask Default value is 255 255 255 0 Mask LAN DHCP Base IP for the LAN port which functions as a Gateway for the Base IP subnet Default value is 192 168 2 1 DHCP IP Value is set in units of hours Default value is 120hr 5 Days Lease Time The time IP address are assigned to the LAN clients 3 NAT Settings VOIP DEVICE STATUS BASIC OPTIONS SUPER OPTIONS FXS PORT FXO PORT BASIC OPTIONS gt NAT Settings DMZ IP WAN Port WAN Port WAN Port WAN Port WAN Port WAN Port WAN Port WAN Port Port Map DMZ Forward all WAN IP traffic to a specific IP address if no matching port IP is used by Gate104 itself or in the defined port forwarding Port Allow users to forward a matching TCP UDP port to a specific LAN IP Map address with a specific TCP UDP port 0 LAN IP LAN Pcri 0 Protocol s 0 LAN IP LAN Pert Protocol v 0 LAN IP LAN Pert Protocol v 0 LAN IP LAN Peril Protocol v 0 LAN IP LAN Pert Protocol v 0 LAN IP LAN Pert Protocol 0 LAN IP LAN Pert 0 Protocol 0 LAN IP LAN Pert 0 P otocol Sa
2. PPPoE Account ID PPPoE Password Preferred DNS Server Ta OStatic IP P Address Subnet Mask 19 gt Default Router DNS Server 1 f ore OIOI DNS Server 2 a PL PL P Cloned WAN MAC Adar l In hex format FAL o fol fo IP Address There are 2 modes under which the Gate104 can operate If DHCP mode is enabled then all the field values for the Static IP mode are not used even though they are still saved in the Flash memory The Gate104 will acquire its IP address from the first DHCP server it discovers from the LAN it is connected To use the PPPoE feature the PPPoE account settings need to be set The Gate104 will attempt to establish a PPPoE session if any of the PPPoE fields is set If Static IP mode is enabled then the IP address Subnet Mask Default Router IP address DNS Server 1 primary DNS Server 2 secondary fields will need to be configured These fields are reset to zero by default Cloned WAN MAC Address Allow the user to set a specific MAC address Set in Hex format 2 LAN Settings VOIP DEVICE STATUS BASIC OPTIONS WANS 5 tings SUPER OPTIONS FXS PORT FXO PORT BASIC OPTIONS gt LAN Settings LAN Subnet Mask 255 255 255 0 Default is 255 255 255 0 LAN DHCP Base IP L gt
3. Auto Dial number to be automatically dialed upon offhook Please note that only the user part of a SIP address needs to be entered here The Gate104 will automatically append the and the host portion of the corresponding SIP address Enable Call Default is Yes If set to Yes Call Forwarding amp Do Not Disturb are Features supported locally Disable Default is No Call Waiting 4 Other Settings VOIP DEVICE STATUS BASIC OPTIONS SUPER OPTIONS SUPER OPTIONS gt Other Settings SUBSCRIBE for MWI No do not send SUBSCRIBE for Message Waiting Indication O Yes send perodic SUBSCRIBE for Message Waiting Indication FXS Impedance New Zealand 2 370 Ohm 620 Ohral 310nF h Special Feature Standard x Onhook Voltage 36v Polarity Reversal ONo Yes Reverse polarity upon call establishment and termination Lock Keypad Update e O Yes If Yes configuration update via keypad is disabled SaveSet Reboot SUBSCRIBE for MWI Default is No When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically FXS Impedance Selects the impedance of the analog telephone connected to the Phone port Special Feature Selects to work well with some Soft Switch Onhook Voltage Select the onhook voltage to suit different area or PBX Polarity Select Polarity Reversal to adapt some call charge
4. billing Reversal system Default is No Lock Keypad If this parameter is set to Yes the configuration update Update via keypad is disabled FXO Port 1 SIP Settings Same as FXS port page 2 Audio Settings Same as FXS port page 3 Phone Feature VOIP Early Dial No O Yes Use Yes only if proxy supports 484 response DEVICE STATUS Dial Plan Prefix ___ This prefix string is added to each dialed number BASIC OPTIONS Use as Dial Key ONo Yes Ifsetto Yes will function as the Redial key SUPER OPTIONS PSTN AC Terninati 320 Ohm 1050 Ohm 230 nF i i FXS PORT i IL vi aeea E j PSTN Disconnect Tone Fregquency fl 480 f2 620 Hz 0 inactive default is 480Hz 620Hz PSTN Disconnect Tone Cadence Choice 1 On 0 of o lige 0 disabled Choice 2 On 0 om0 ms 0 disabled Choice 3 On 0 om0 mas 0 disabled PSTN Silence Timeout 60 Se Terminate call after long silence detected default is 60 sec max 65536 SUBSCRIBE for MWI No do not send SUBSCRIBE for Message Waiting Indication Il Yes send periodical SUBSCRIBE for Message Waiting Indication Special Feature I Standard x SaveSet Reboot PSTN AC Termination Selects the impedance of the analog telephone connected to the Line port PSTN Disconnect Tone The tone you will hear when the PSTN is disconnected PSTN Disconnect Tone We suggest our customers to use 0 Cadence PST
5. dial tone Then dial the forward number and for a dial tone then hang up 73 Cancel Unconditional Call Forward To cancel Unconditional Call Forward dial 73 and get the dial tone then hang up 90 Busy Call Forward To use this feature dial 90 and get the dial tone Then dial the forward number and for a dial tone then hang up x91 Cancel Busy Call Forward To cancel Busy Call Forward dial 91 and get the dial tone then hang up 92 Delayed Call Forward To use this feature dial 92 and get the dial tone Then dial the forward number and for a dial tone then hang up 93 Cancel Delayed Call Forward To cancel this Forward dial 93 and get the dial tone then hang up Flash Hook When in conversation this action will switch to the new incoming call if there is a call waiting indication When in conversation without an incoming call this action will switch to a new channel for a new call 4 3 2 PSTN Pass Through When Gate104 is out of power or loses registration or if the network connection is down the RJ 11 line jack on the side of Gate104 will function as a pass through jack Users will be able to use the same analog phone for PSTN calls 4 4 LED Light Pattern I ndication Following tables show the LED light pattern indication RED LED indicates abnormal status DHCP Failed or WAN No Button flashes every 2 seconds if DHCP i
6. seconds Enable WAN Web Accece O No Yes If Yes WAN WED access to this configuration page is enabied TFTP Server o o Jo 0 Remote software upgrade and configuration FXS PORT i FXO PORT NTP Server time nist gov URL or IP addvess Super Password Device password to configure this page Layer 3 OoS 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 Layer 2 OoS This contains the value used for layer 2 VLAN tag Default setting is blank No Key Default is 4 seconds Entry Timeout Enable WAN If this parameter is set to No the HTML configuration update Web Access via WAN port is disabled TFTP Server This is the IP address of the configured TFTP server If it is non zero or not blank the Gate104 will attempt to retrieve new configuration file or new code image from the specified TFTP server at boot time It will make up to 3 attempts before timeout and then it will start the boot process using the existing code image in the Flash memory If a TFTP server is configured and a new code image is retrieved the new downloaded image will be verified and then saved into the Flash memory NTP Server This parameter defines the URI or IP address of the NTP server which is used by the Gate104 to display the current date time Super This contains the password to access the Advanced Web Pas
7. the local RTP RTCP port pair theGate104 will listen and transmit It is the base RTP port for channel 0 When configured channel O will use this port _value for RTP and the port_value 1 for its RTCP channel 1 will use port_value 2 for RTP and port_value 3 for its RTCP The default value for FXS port is 5004 The default value for FXO port is 5008 Use Random Port Proxy Require This parameter when set to Yes will force random generation of both the local SIP and RTP ports This is usually necessary when multiple Gate104 are behind the same NAT SIP Extension to notify SIP server that the unit is behind the NAT Firewall Send DTMF This parameter controls how DTMF events are transmitted There are 3 ways in audio which means DTMF is combined in audio signal not very reliable with low bit rate codec via RTP RFC2833 or via SIP INFO DTMF Payload Type This parameter sets the payload type for DTMF using RFC2833 Caller ID Select the Caller ID Scheme to suit the standard of different Scheme area Bellcore North America ETSI FSK France Germany Norway Taiwan UK CCA ETSI DTMF Finland Sweden Denmark DTMF Denmark CID Canada Send If this parameter is set to Yes the From header in Anonymous outgoing INVITE message will be set to anonymous essentially blocking the Caller ID from displaying Send Flash This parameter allows users to control whether to send a
8. Gate104 User Manual 1 Welcome Gatel104 is an all in one VoIP integrated access device that features superb audio quality rich functionalities high level of integration compactness and ultra affordability The Gate104 is fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market It allows call origination and termination from to the PSTN network via FXO port remotely and automated emergency call routing through PSTN network 2 Key Features e Supports SIP 2 0 RFC 3261 TCP UDP IP RTP RTCP HTTP ICMP ARP RARP DNS DHCP both client and server NTP PPPoE STUN TFTP etc e Built in router NAT Gateway and DMZ port forwarding e Supports call origination and termination from to the PSTN network via FXO port e Powerful digital signal processing DSP to ensure superb audio quality advanced adaptive jitter control and packet loss concealment technology e Support various vocoders including G 711 a law and u law G 723 1 5 3K 6 3K G 726 40K 32K 24K 16K as well as G 728 G 729A B and iLBC e Support Caller D Name display or block Hold Call Waiting Flash Call Transfer Call Forward in band and out of band DTMF Dial Plans etc e Support 3 way conferencing e Support fax pass through and T 38 e Support Silence Suppression VAD Voice Activity Detection CNG Comfort Noise Generation Line Echo Cancellation G 168 and AGC Autom
9. N Silence Timeout Terminate call after long silence detected default is 60 sec Max 65536 6 Restore Factory Default Setting Warning Restore the Factory Default Setting will DELETE all configuration information of the device Please backup or print out all the settings before you approach to following steps We will not take any responsibility if you lose all the parameters of setting and cannot connect to your service provider Please disconnect network cable and power cycle the unit before trying to reset the unit to factory default The steps are as follows Step 1 Find the MAC Address of the device The MAC address of the device is located on the bottom of the device It is a 12 digits hex number e Step 2 Encode the MAC address to decimal digits Please use the following mapping T P ee Ce N N N N For example for MAC address 000a8200f395 the user should encode it as 0002282003333395 e Step 3 Access the voice menu by pressing or the LED button then dial 99 and get the voice prompt RESET e Step 4 Key in the encoded MAC address decimal digits after hear the IVR prompt Once the correct encoded MAC address is entered the device will reboot automatically and restore the factory default setting NOTES e Be advised by default the Gate104 WAN side HTTP access is disabled After the factory reset the ONLY way to get access to the configuration page is connecting your
10. atic Gain Control e Support standard encryption and authentication DIGEST using MD5 and MD5 sess e Support for Layer 2 802 1Q VLAN 802 1p and Layer 3 QoS ToS DiffServ MPLS e Support automated NAT traversal without manual manipulation of firewall NAT e Support device configuration via built in IVR Web browser or central configuration file through TFTP or HTTP e Support firmware upgrade via TFTP or HTTP with encrypted configuration files e Ultra compact wallet size and lightweight design great companion for travelers e Compact lightweight Universal Power adapter 3 Hardware Specifications LAN Interface 1 WAN Interface 1 FXS Telephone Port 1 FXO Port 1 Button 1 LED Green and Red Power Adaptor Input 100 240VAC 50 60 Hz Output 9VDC 1200mA UL certified Dimension 70mm W 130mm D 27mm H Weight 0 9 Ibs 0 4 kg Temperature 40 1300F 5 450C Humidity 10 90 Compliance FCC amp CE 4 Basic Operations 4 1 Voice Prompt Gate104 has stored a voice prompt menu for quick browsing and simple configuration To enter this voice prompt menu simply press the button or from the analog phone Main Enter a Menu Enter for the next menu Menu Option option Enter to return to the main menu Enter 01 06 47 86 99 menu option 01 DHCP Mode Enter 9 to toggle the Static IP Mo
11. can either hang up or make another call e 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 e Busy tone keeps playing This means we have failed to receive the second NOTIFY from the transferee and decided to time out Note this does not indicate the transfer has been successful nor does it indicate the transfer has failed When transferee is a client that does not support the second NOTIFY such as our own earlier firmware this will be the case In bad network scenarios this could also happen although the transfer may have been completed successfully 4 2 4 Attended Transfer Assuming that call party A and B are in conversation A wants to Attend Transfer B to C 1 A presses FLASH on the analog phone or Hook Flash for old model phones to get a dial tone 2 Athen dial C s number then or wait for 4 seconds B and C now in conversation 3 A can hang up Note When intended Transfer failed if A hangs up the Gate104 will ring user A again to remind A that B is still on the call by pressing FLASH or Hook again will restore the conversation between A and B 4 2 5 3 way Conferencing Assuming that call party A and B are in conversation A wants to bring Cina conference 1 A presses FLASH on the analog pho
12. computer to the LAN port of Gate104 7 Warranty End users should contact the company from whom you purchased the product for replacement repair or refund Warning Please do not attempt to use a different power adaptor Using other power adaptor may damage the Gate104 and will void the manufacturer warranty Caution Changes or modifications to this product not expressly approved by us or operation of this product in any way other than as detailed by this User Manual could void your manufacturer warranty
13. ddress is used in SIP SDP message _ Sesa ia NAT This parameter defines whether the Gate104 NAT traversal Traversal mechanism will be activated or not If activated by choosing Yes and a STUN server is also specified then the Gate104 will behave according to the STUN client specification Under this mode the embedded STUN client inside the Gate104 will attempt to detect if 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 Gate104 will attempt to use its mapped public IP address and port in all its SIP and SDP messages If the NAT Traversal field is set to Yes with no specified STUN server the Gate104 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 Keep This parameter specifies how often the Gatel04 sends a blank Connected UDP packet to the SIP server in order to keep the hole on the Interval NAT open Use NAT IP NAT IP address used in SIP SDP message Default is blank 2 Sys Feature SUPER OPTIONS gt Sys Feature VOIP oa OM Lever 2 QoS 302 1O VLAN Tag 0 802 1p priority value 2 Jon Lever 3 QoS 48 DiffServ or Precedence value BASIC OPTIONS 7 pe AA a No Key Entry Timeout 4 Seconds Default 4
14. de selection 02 IP Address IP The current WAN IP address address is announced Enter 12 digit new IP address if in Static IP Mode 03 Subnet IP Same as menu 02 address 99 Pending No Voice RESET 04 Gateway IP Same as menu 02 address 05 DNS Server Same as menu 02 IP address 06 TFTP Server Same as menu 02 IP address 47 Direct IP Calling When entered you will be prompted a dial tone then enter 12 digit IP address This menu can also be entered by pressing the button again 86 Voice Messages Enter 9 to dial pre configured phone number to retrieve VM Messages Enter 9 to reboot the phone Enter encoded MAC address to restore factory default setting Notes Invalid Entry Automatically returns to main menu e Once the button is pressed it enters the voice prompt main menu If the button is pressed again while it is already in the voice prompt menu it jumps to Direct IP Call option and a dial tone is prompted e shifts down to the next menu option e returns to the main menu e 9 functions as the ENTER key in many cases to confirm an option e All entered digit sequences have known lengths 2 digits for menu option and 12 digits for IP address Once all of the digits are collected the input will be processed e Key entry can not be deleted b
15. ed ring tones a dial tone is played At this time users can dial a VolP telephone number then or wait for 4 seconds The call will be established afterwards If no VolP number is entered after the dial tone Gate104 will hang up automatically in 10 seconds In the web configuration page if the Route to VoIP field is configured the second stage dialing is eliminated That is after users dial the FXO port telephone number the VolP number will be called automatically 4 2 9 Route Calls to PSTN If configured certain calls will be routed to PSTN line automatically This call feature is especially useful for emergency calls or local telephone calls To use this feature users need to specify a prefix or a telephone number in the Route to PSTN field in the web configuration page If the dialed digits match one of the specified prefix outbound calls will be routed to PSTN port 4 3 Call Features 4 3 1 Call Features Table Following table shows the call features of Gate104 30 Block Caller ID for all subsequent calls 31 Send Caller ID for all subsequent calls 67 Block Caller ID per call 82 Send Caller ID per call 50 Disable Call Waiting for all subsequent calls 51 Enable Call Waiting for all subsequent calls 70 Disable Call Waiting Per Call 71 Enable Call Waiting Per Call x72 Unconditional Call Forward To use this feature dial 72 and get the
16. his 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 3 Dial Settings VOIP DEVICE STATUS BASIC OPTIONS SUPER OPTIONS SUPER OPTIONS gt Dial Settings Early Dial m JL No Dial Plan Prefix This prefix string is added to each dialed number Useras Dialer ONo Yes Ifset to Yes will function as the Redial key Offhook Auto Dial User D extension to dial automatically when offhook ONo Yes Enable Call Feat psima Qf Yes Call Forwarding amp Call Waiting Disable are supported locally Disable Call Waiting Ono O ves Early Dial Default is No Use only if proxy supports 484 response Dial Plan Sets the prefix added to each dialed number Prefix Use as This parameter allows users to configure the key to be used Dial Key as the Send or Dial key If set to Yes pressing this key will immediately trigger the sending of dialed string collected so far In this case this key is essentially equivalent to the Re Dial key If set to No this key will then be included as part of the dial string to be sent out Offhook This parameter allows users to configure a User ID or extension
17. lained in details as below Device Status MAC Address 00 09 45 70 99 B WAN IP Address 192 168 1 166 Product Main Chip TIS000 Sofware Version 2 0 22 System Up Time O day s 0 hour s 5 minute s Registered Status Yes PPPoE Link Up _ disabled NAT i VOIP DEVICE STATUS BASIC OPTIONS SUPER OPTIONS FXS PORT FXO PORT MAC Address The device ID in HEX format This is a very important ID for ISP troubleshooting WAN IP This field shows WAN port IP address Address Product Main Chip Model I nfo Chip Software Program This is the main software release its number is Version always used for firmware upgrade Bootloader This is normally not changed HTML This is the web user interface normally not changed VOC This is the codec program normally not changed System Up This field indicates how long the device has been up since the Time last reboot Registered This field indicates whether the device is registered to the SIP Status server PPPoE Link Up This field shows whether the PPPoE connection is enabled or not NAT This field shows what kind NAT the Gate104 is connected to via its WAN port It is based on STUN protocol Basic Options 1 WAN Settings VOIP DEVICE STATUS BASIC OPTIONS gg FXS PORT FXO PORT BASIC OPTIONS gt WAN Settings Dynamically Assigned IP DHCP by default or PPPoE
18. n Event SIP NOTIFY message indicating the Flash event or just to switch to the voice channel when users press the Flash key Fax Mode Select to send amp receive fax via Internet or PSTN Default is T 38 protocol via internet 2 Audio Settings VOIP DEVICE STATUS BASIC OPTIONS SUPER OPTIONS SUPER OPTIONS gt Audio Settings Preferred Codecs iLBC Payload Type 99 Between 96 and 127 default is 98 Voice Frames per TX Preference 1 G729 Preference 2 G711A Preference 3 G723 Preference 4 G729 Preference 5 G726 32 Preference 6 ILBC Y lt lt lt 1 lt G723 Rate 63 kbps encoding rate 5 3 kbps encoding rate iLBC Frame Size 20ms 30ms 2 lt i Up to 10 20 32 and 64 for G711 G726 G723 and other codecs respectively Silence Suppression I No O Yes Preferred The Gate104 supports up to 7 different Codecs types including Codecs G 711 A U law G 723 1 G 726 G 728 G 729A B iLBC Depending on the product model some of these Codecs may not be provided in standard release Users can configure Codecs in a preference list that will be included with the same preference order in SDP message The first Codec in this list can be entered by choosing the appropriate option in Choice 1 Similarly the last Codec in this list can be entered by choosing the appropriate option in Choice 7 G 723 Rate This defines
19. ne or Hook Flash for old model phones to get a dial tone 2 A dials 23 then C s number then or wait for 4 seconds A and C are now in conversation 3 A presses FLASH again to begin conference 4 2 6 Send and Receive PSTN Calls Users can send and receive calls from PSTN To receive PSTN calls simply take the phone off hook when the analog phone rings To make a PSTN call first press 00 or your own PSTN Access Code to get the PSTN line dial tone and dial the PSTN number There should not be any PBX between the Gate104 and the PSTN wall jack or else the making PSTN call function will be failed 4 2 7 Vol P to PSTN Calls To make a VolIP to PSTN call users need to dial the FXO SIP account phone number first A ring tone is played once followed by a dial tone At this time users can dial a PSTN telephone number or a mobile telephone number then or wait for 4 seconds The call will be established afterwards If no PSTN number is entered after the dial tone Gate104 will hang up automatically in 10 seconds In the web configuration page if the Route to PSTN field is configured the second stage dialing is eliminated That is after users dial the FXO SIP account number the PSTN number will be called automatically 4 2 8 PSTN to Vol P Calls To make a PSTN to VolP call PSTN callers need to originate a call to the FXO port telephone number first If no one answers the FXS phone after 4 default value can be configur
20. o Calls are unconditionally forwarded to the specified PSTN phone PSTN number once users dial the FXO port VolP number Forward to Calls are unconditionally forwarded to the specified VolP phone VoIP number once users dial the FXO port PSTN number 5 Other Settings VOIP DEVICE STATUS BASIC OPTIONS SUPER OPTIONS FXS PORT FXO PORT BASIC OPTIONS gt Other Settings Basic User Password Basic wer password to configure this device Time Zone GMT 5 00 US Eastern Time New York No Daylight Savings Time i Oves If set to Yes display time willbe 1 hourahead of normal time Sees Reboot Basic User This contains the password to access the Web Configuration Password Menu Time Zone This parameter controls how the displayed date time will be adjusted according to the specified time zone Daylight This parameter controls whether the displayed time will be Savings Time daylight savings time or not If set to Yes then the displayed time will be 1 hour ahead of normal time Super Options 1 Sip Settings VOIP DEVICE STATUS BASIC OPTIONS F E FXS PORT FXO PORT SUPER OPTIONS gt SIP Setungs On NAT Traversal Oi Yes STUN server is URL or IF Port Keep Connected Interval 20 Seconds Default 20 seconds Use NAT IP If specified this IP a
21. ollowing is a table of the encoding scheme for the most commonly used characters 00 01 02 03 04 05 06 07 08 WW 00 Ni Oy UI BY WW NF 09 O dot character 4 column character Examples If the target IP address is 192 168 0 160 the dialing convention is Voice Prompt with option 47 then 192168000160 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 If the target IP address port is 192 168 1 20 5062 then the dialing convention would be Voice Prompt with option 47 then 192168001020 45062 followed by pressing the key if it is configured as a send key or wait for 4 seconds 4 2 3 Blind Transfer Assuming that call party A and B are in conversation A wants to Blind Transfer B to C 1 A presses FLASH on the analog phone or Hook Flash for old model phones to get a dial tone 2 Then A dials 87 then dials C s number and then or wait for 4 seconds 3 Acan hang up Note Call Feature has to be set to YES A can hold on to the phone and wait for one of the three following behaviors e A quick confirmation tone temporarily using the call waiting indication tone followed by a dial tone This indicates the transfer is successful transferee has received a 200 OK from transfer target At this point A
22. s Cable configured Gatel104 fails to register Button flashes every 2 seconds if SIP server is configured GREEN LED indicates normal working status Message Waiting Indication Button flashes every 2 seconds Ringing Button flashes at 1 10 second Ringing Interval Button flashes every second 5 Configuration Guide 5 1 Configuration through Voice Prompt 5 1 1 DHCP Mode Follow section 4 1 with voice menu option 01 to enable Gate104 to use DHCP 5 1 2 STATIC IP Mode Follow section 4 1 with voice menu option 01 to enable Gate104 to use STATIC IP mode then use option 02 03 04 to set up Gate104 s IP Subnet Mask Gateway respectively 5 2 Web Configuration 5 2 1 Access the Web Configuration Menu The Gate104 HTML configuration menu can be accessed via LAN or WAN port e From the LAN port use the default LAN gateway IP address http 192 168 2 1 e Get the WAN IP address of the Gate104 through section 5 1 with menu option 02 Gate104 s Web Configuration page can be accessed by the following URI via WAN port http IP Address 5 2 2 Web Configuration Page Once this IP address is entered and sent from a Web browser the Gate104 will respond with the following login screen Login The password is voip After a correct password is entered in the login screen the embedded Web server inside the Gate104 will respond with the Configuration page which is exp
23. sword Configuration page FXS Port 1 SIP Settings SUPER OPTIONS gt SIP Settings VOIP ISIP Server Address IP address or URL AAA Outbound Proxy IP address or URL if any BASIC OPTIONS SUPER OPTIONS pe Ueer ID L Assigned user ID ur pluie nuber I Account ID a __ Canbe same as or different from SIP User ID Authentication Password For secarity password does not display Name Optional ee Use DNS SRV Ono Yes User ID is phone number ONo Yes SIP Registration Ove Oro Unnegister On Reboot pi O Yea No Register Expiration 7 60 Minutes Default is 1 hour max 45 days Local SIP Port 5060 Default 5060 Local RTP Port 5004 102465535 default 5004 7 Use Random Port No O Yes Proxy Require If specified the content will appear in Proxy Require header Send DTMF InAudio O Via RTP RFC2833 Via SIP INFO DTMF Payload Type 101 Caller ID Scheme v Send Anonymous No Yes If Yes caller ID will be blocked Send Flash Event No O Yes Flash will be sent as a DTMF event if set to Yes FexMode 7 38 Auto Detect Pass Through SIP Server SIP Server s URI or IP address Address Outbound Proxy SIP Outbound Prox
24. the encoding rate for G723 Codec By default 6 3kbps rate is chosen iLBC Frame This sets the iLBC size in 20ms or 30ms Size iLBC Payload This defines payload time for iLBC Default value is 98 The Type valid range is between 96 and 127 Voice Frames per TX 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 Codec in the above Codec Preference List or the actual used payload type negotiated between the 2 conversation parties at run time e g if the first Codec 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 Codec 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 Gate104 will use and save the maximum allowed value for the corresponding first Codec choice The maximum value for PCM is 10 x10ms frames for G726 it is 20 x1Oms frames for G723 it is 32 x30ms frames for G729 G728 64 x10ms and 64 x2 5ms frames respectively Silence Suppression T
25. ut the phone may prompt error once it is detected 4 2 Make Phone Calls 4 2 1 Calling phone or extension numbers To make a phone or extension number call a Dial the number directly and wait for 4 seconds default No Key Entry Timeout Or b Dial the number directly and press assuming that Use as dial key is selected in web configuration Other functions available during the call are call waiting flash call transfer and call forward 4 2 2 Direct IP calls Direct IP calling allows two phones that is a Gate10x with an analog phone and another VoIP Device to talk to each other in an ad hoc fashion without a SIP proxy VoIP calls can be made between two phones if e Both Gate10x ATA and the other VoIP device i e another Gate10x ATA or other SIP products have public IP addresses or e Both Gate10x ATA and the other VoIP device i e another Gate10x ATA or other SIP produces are on the same LAN using private or public IP addresses or e Both Gate10x ATA and the other VoIP device i e another Gate10x ATA or other SIP products can be connected through a router using public or private IP addresses To make a direct IP call first pick up the analog phone or turn on the speakerphone on the analog phone follow Section 4 1 with voice prompt 47 followed by the 12 digit target IP address Destination ports can be specified by using 4 encoding for followed by the encoded port number F
26. veSet Reboot 4 Call Settings VOIP DEVICE STATUS BASIC OPTIONS JS 5 BASIC OPTIONS gt Call Settings Number of Rings 2 Number of phone rings before a PSTN incoming call is forvrarded default 4 PSTN Access Key Key pattem to use PSTN line default is 00 Key pattem to authorize calling PSTN numbers from VOIP no cefault VOIP Call PSTN Key PSTN CALL VOIP Key Key pattem to authorize calling VOIP terminals from PSTN no default Shine tting SUPER OPTIONS FXS PORT FXO PORT Outbound calls will be routed to PSTN port when dialed digits match one of the following Route Call to PSTN Forward to PSTN Forward to VoIP _ PSTN calls will be forwarded to the specified VoIP number Number of This parameter specifies the number of phone rings for incoming Rings PSTN calls to FXO port Default is 4 PSTN This field allows users to customize their own code to access the Access Key PSTN line Default is 00 VolP Call Key pattern to authorize calling PSTN numbers from VoIP no PSN Key default PSTN Call Key pattern to authorize calling VolP terminals from PSTN no VoIP Key default Router Call If the dialed digits match one of the specified prefix here to PSTN outbound calls will be routed to PSTN port This field is especially useful for emergency calls Forward t
27. y Server s URI or IP address SIP User ID SIP service subscriber s User ID Account ID SIP service subscriber s Account ID Can be identical to or different from SIP User ID Authentication SIP service subscriber s account password Password Name SIP service subscriber s name which will be used for Caller ID display Use DNS SRV Default is No If set to Yes the client will use DNS SRV for server lookup User ID is phone number If the Gate104 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 SIP Registration This parameter controls whether the Gatel04 needs to send REGISTER messages to the proxy server The default setting is Yes Unregister On Reboot Default is No If set to yes the SIP user will be unregistered on reboot Register Expiration This parameter allows the user to specify the time frequency in minutes the Gate104 refreshes its registration with the specified registrar The default interval is 60 minutes or 1 hour The maximum interval is 65535 minutes about 45 days Local SIP Port Local RTR Port This parameter defines the local SIP port the Gate104 will listen and transmit The default value for FXS port is 5060 The default value for FXO port is 5062 This parameter defines

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