ATA V600 Operation & User’s Manual

CHAPTER 1 Installation………………………..……………………………….4 CHAPTER 2 Production Summarize……………………..…………………5 I、Key Features……………………………………………………………………….5 II、Technical Parameter……………………….………………….………..…………5 CHAPTER 3 Basic Manipulation………..…….………………..…………….6 I 、 K e y s t r o k e s A n d V o i c e P r o m p t … … … … … … … … … … . . … … … . . … … . … … … … … 6 II、Make a Call……………………………………………..…………….…………….8 i、How to Make a PSTN Call ……..………………………………………………8 i i 、 H o w t o M a k e a V o I P

iii、Direct IP Call……………..…………………………………………………….9 III、Call Features ………………….………………………..…………..……………11 IV、Call Function………..……………….………………………………………….12 i、 Blind Transfer …………………….……………………………………………12 ii、Attended Transfer …………….……………………………………………13 iii、Conference Call ……………………………………………………….………13 iv 、 VoIP-to-PSTN Calls………….………………………………………………….14 v、PSTN-to-VoIP Calls……….………………..………………………..………….14 vi、Route Calls to PSTN………………………………………………….…………14 vii、Fax ……………………………………………………………………..………15 viii、LED Light……………..……………………………………..………………15 CHAPTER 4 Configuration guide ……..…………………..………………16 I、Configuring Wan IP Through Voice Prompt……………………….…………16 i、Dhcp Mode….. ……………………………………………………………16 ii、Static IP Mode …….…………………………………………………………17 II、Configuring With Web Browser ……………..…….…………………………17 i、Access the Web Configuration Menu ….…..……………….………………18 ii、End user Configuration ………………………….…………………………18 III、Status……………………………..…………………………………….……...18 IV、Basic Optino…….……………….…………………….………………….……18 i、Lan Settings …………………………….………………………….18 ii、Wan Settings ………………………….….………………………….19 iii、Nat Settings……………………………….….………………………………21 iv、Other Settings……..……………………………………………………………22 v、 Call Settings(V611Fxo port)………..……..……………………………………22 V、Super Options……..…….…………………………………………………..24 i、SIP Settings……………………………………………….………….24 ii、Audio.Settings ………………..………………………………….…………...28 iii、Dial.Settings ……………………………………………………….………...31 iv、Other Settings ………………..………………………….…………….……….32 v、Fxo Port --->Phone Feature……...…………..…….….……………........……….34 CHAPTER 5 Restore Factory Defaults Setting………..…………………37 CHAPTER 6 Upgrade………………………………………………….……...…..38 CHAPTER 7 Ordinary Quality Checking……….…………………………39 CHAPTER 8 FAQ………………………………..………………….......................39

SIP2.0, and is compatible other SIP products or software. The IP voice gateway is an all-in-one VoIP integrated access device that features superb audio quality, rich functions, and high level of integration, compactness and ultra-affordability.

What’s in ATA package? 1)One ATA 2)One universal power adapter 3)One Product Enchiridion 4)One Ethernet cable 5)One phone line Interconnection diagram of ata

the defective products. Changes or modifications to this product not expressly approved by our company, or operation of this product in any way other than as detailed by this User Manual, could void your manufacturer warranty. CHAPTER 2 Production Summarize I、 Key features • Support SIP2.0 (RFC3261), TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP/RARP, DHCP, NTP, PPPoE, STUN and TFTP etc; • Inside Rooter、NAT、gateway and DMZ Port transmit; • Support the PSTN making or receiving calls (by the FXO Port); • Use the DSP CNOS chip to keeping a wonderful audio-quality, advanced shaking control, and the technology of hiding the lost massage;

G.723.1 (5.3K/6.3K) G.726 (40K/32K/24K/16K), G.729A/B, and iLBC; • Support incoming call on show, restricting, and holding, disconnection, call transfer, call divert, DTMF, dialing project, etc; • Support conference call; • Support passing through and T.38 fax, voice restrain and jerquers, CNG, echo restrains (G.168), AGC, and DIGEST using MD5 and MD5-sess; • Support layer 2 (802.1Q VLAN, 802.1p) and layer 3 (QoS, DiffServ, ToS); • Support NAT auto- penetrating, no need to modify the setting of the NAT; • Support configuration files by inside IVR equipment, Web browser, or TFTP and HTTP Center Server; • Support upgrading encrypts configuration files by TFTP or HTTP; • Microminiaturize and legerity design (size as a wallet), a voice gateway that convenient for schlepping II、Technical parameter ATA(SIP2.0) YES route YES(V610、V610、V620) bridge YES(V610、V620) DHCP YES 1WAN/1LAN(V610、V610、 V620) Ethernet Port 1WAN(V600) 1FXS(V600、V610、V610) FXS 2FXS(V620) FXO 1FXO(V610) Exit YES(V610、V610) 60×110×25mm(V600) 74×106×25mm(V610) Size 101 × 101 × 31mm （V610 、 V620） 316g(V600) Weight 405g(V610)

0—40℃ Humidity 10％—95％ Power supply adapter AC IN:100V-240V DC OUT:+9V/V600mA Use power：<2.5W Authentication FCC/CE CHAPTER 3 Basic Manipulation I、Keystrokes and voice prompt There is a inside voice shortcut menu in ATA, can do look through fleetly and do simple collocation, press button can enter the voice hint menu. Press the button or the ‘***’ of the common phone, can enter the voice shortcut menu. Menu Voice Prompt User’s Options Dial ‘***’ to Main Menu “Enter a Menu Option” Dial ‘****’ to review the configuration of the device; or Dial 01 – 06, or 99 menu option; or Dial ‘#’ to return to the Main Menu Dial ‘01’ to configure the DHCP Mode “Static IP Mode”, or “Dynamic IP Mode” Dial ‘9’ to toggle the selection. If user selects “Static IP Mode”, user need configure the all IP address information through menu 02 to 05. If user selects “Dynamic IP Mode”, the device will retrieve all IP address information from DHCP server automatically when user reboots the device. Dial ‘02’ to configure the IP address of the device “IP Address” + IP address Such as “IP Address xxx.xxx.xxx.xxx” Enter 12-digit new IP address if in Static IP Mode. for example the address is “192.168.1.103”, user should dial “192168001103”

configure the subnet mask “Subnet mask” + subnet mask Enter 12-digit new subnet mask address if in Static IP Mode, for example the address is “255.255.255.0”, user should dial “255255255000”. Dial ‘04’ to configure the default gateway “Gateway “ + IP address Enter 12-digit IP address of the default gateway if in Static IP Mode. Dial ‘05’ to configure the DNS server “DNS Server” + IP address Enter 12-digit IP address of the DNS server if in Static IP Mode. Dial ‘06’ to configure the TFTP server “TFTP Server “ + IP address Enter 12-digit IP address of the TFTP server TFTP server is used to update the firmware of the device. Dial ‘47’ to make a direct IP call “Direct IP Calling” When dial ‘47’, user will prompt a dial tone, then dial the 12-digit IP address. For Example, user wants call another SIP phone at “192.168.1.101”. User should dial ‘47’ and dial ‘192168001101’. (For detail, see “CHAPTER 3 Make a Direct IP Call”.) Dial ‘86’ to check the voice message “No Voice Messages”; or “Voice Messages Pending” If there are voice messages, user can dial ‘9’ and dial pre-configured phone number to retrieve voice message. Dial ‘99’ to reset the device “RESET” Dial ‘9’ to confirm the RESET; or Enter MAC address to restore factory default setting (For detail, see CHAPTER 5) Dial the invalid number or keypad “Invalid Entry” Automatically return to Main Menu Note: Press button can enter the voice shortcut menu. Press it again, can switch to ‘making IP call’ or tonality adjusting mode, when entered the voice hint

menu, Button ‘#’ can go back to the main menu, Button ‘9’ is like the button ‘enter’, can notarize the current option. All the typing digits can be recognized by length, two digits is menu option, 12 digits is IP address. If key in string digits, the system will administer the relevant order by its length. If key in the wrong dictate, cannot be deleted, but the system will hint that you have made a mistake by voice. Keyboard input cannot by deleted, but the phone will hint when being deleted. II 、Make a call i、How to make the PSTN call (V610、V610) Connect it exactly as the scheme. Press *00, change to the PSTN line; dial the number you want immediately. The PSTN number holds the line. ii、How to make a VoIP call 1.Connect it exactly as the scheme. Register to the SIP terrace .Dial the number you want according to the rules that offered by the SIP Internet terrace and wait for 4 seconds (default) 2. Connect it exactly as the scheme. Register to the SIP terrace .Dial the number you want according to the rules that offered by the SIP Internet terrace and then press ‘#’ (assuming that “Use # as dial key” is selected in web configuration) iii、Direct IP call  Connect it exactly as the scheme. Dial the number you want

 Insuring the both sides are in the same network range, according with one of the following condition. (The IP of ATA in the following condition means the WAN IP)  The ATA or IP equipment of the both sides must have the public WAN IP address  The ATA or IP equipment of the both sides must in the same LAN, and the IP must in the same Internet range. E.g. 192.168.1.10 and 192.168.1.20 Make a call immediately according to the IP address. Pick up the phone or press the speakerphone Press ‘***47’ Input the 12 digits IP address of the other side with the port at the end, e.g. if the IP address is 192.168.1.188, port is 5066, press 192168001188*45066, if the port is 5060, it can be omitted. Press ‘#’ for affirming sending or wait for 4 seconds for auto-sending Coding form of the character in most common use 00 0 01 1 02 2 03 3 04 4 05 5 06 6

11 07 7 08 8 09 9 *0 . (Dot) *1 _（Underline） *2 - (Hyphen) *3 @ *4 :(Colon) 21 A 22 B 23 C 31 D 32 E 33 F 41 G 42 H 43 I 51 J 52 K 53 L 61 M 62 N 63 O 71 P 72 Q

coding is ‘21’, B is the second letter of the button ‘2’, so its coding is ‘22’, and so on. III、Call features Keys Call features 30 Block Caller ID (for all subsequent calls) *31 Send Caller ID (for all subsequent calls) *67 Block Caller ID (only once) *82 Send Caller ID (only once) *50 Disable Call Waiting (for all subsequent calls) *51 Enable Call Waiting (for all subsequent calls) *70 Disable Call Waiting (only once) *71 Enable Call waiting (only once) *72 Unconditional Call Forward. To use this feature, dial ‘*72’ and get the 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

To use this feature, dial ‘*90’ and get the dial tone. Then dial the forward number and ‘#’ for a dial tone, then hang up. *91 Cancel Busy Call Forward To cancel ‘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/H ook When in conversation, this action will switch to the new i 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. IV、 Call function i、 Blind transfer The function is limited to the VOIP calling with FXS port.Assuming that A and B are in conversation. A wants to Blind Transfer B to C:  A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone.  Then A dials # then dials C’s number, and then #(or wait for 4 seconds)  A can 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: A quick confirm 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 can either hang up or make another call. A quick busy tone followed by a restored call (on supported platforms only). This means the transferee has received a 4xx response for the INVITE and we will try to recover the call. The busy tone is just to indicate to the transferor that the transfer has failed. 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

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 has been completed successfully ii、 Attended transfer The function is limited to the VOIP calling with FXS port. Assuming that call CHAPTERY A and B are in conversation. A wants to Attend Transfer B to C:  A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone  A then dial C’s number then # (or wait for 4 seconds). A and C now in conversation.  A can hang up.Note: When intended Transfer failed, if A hangs up, the ATA 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. iii、 Conference call The function is limited to the VOIP calling with FXS Port.Assuming that call CHAPTERY A and B are in conversation. A wants C to join the conversation:  A press ‘FLASH’ (HOOK FLASH of common or medieval phone) to get a dial tone  A then dial #+ C’s number + # (or wait for 4 seconds)， then talk with C  A press flash, begin a conference call. iv、VoIP-to-PSTN Calls(V610 FXO port) To make a VoIP-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, V610 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

automatically. v 、PSTN-to-VoIP Calls(V610 FXO port) To make a PSTN-to-VoIP call, PSTN callers need to originate a call to the FXO port telephone number first. If no one answers the FXS phone after 4 ring tones, a dial tone is played. At this time, users can dial a VoIP telephone number then # (or wait for 4 seconds). The call will be established afterwards. If no VoIP number is entered after the dial tone, V610 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 VoIP number will be called automatically. Vi、 Route Calls to PSTN(V610 FXO port) 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. vii、 Fax ATA supports two kinds of fax, T.38 and pass through. According to different SIP roof, selected audio coding as PCMU or PCMA, then can realize the fax function. viii、 LED light Red LED means in order DHCP fail or WAN cannot connect Flash every 2 seconds (if DHCP has set) ATA fail to register Flash every 2 seconds (if DHCP has set)

Hint information waiting Flash every 2 seconds Calling Flash endless Call stoppage Flash every second CHAPTER 4 Configuration Guide I、 Configuring WAN IP by voice prompt V620 LED explicate As shown from the above figure ,there are five indicator light in V620. 1、The first one on the right indicates whether the gateway has registered to the SIP server. （The red light flashes if it doesn't register, it doesn't flash on the contrary. ） 2、The second and the forth on the right indicate whether the LAN\WAN interface of the gateway has been connected.(They are connected if the lights are bright.) 3、The third and the fifth show whether there're data pass throught the interfaces. V610 LED explicate As shown from the above figure ,there are five indicator light in V610. 1、The first one on the right indicates whether the gateway has registered to the SIP server. （The red light flashes if it doesn't register, it doesn't flash on the contrary. ） 2、The second and the forth on the right indicate whether the LAN\WAN interface of the gateway has been connected.(They are connected if the lights are bright.) 3、The third and the fifth show whether there're data pass throught the interfaces. V610 LED explicate V610 has 4 LED and one the facing of a quilt 1、the facing of a quilt one indicates whether the gateway has registered to the SIP server. （The red light flashes if it doesn't register, it doesn't flash on the contrary. ） 2 、The green one indicate whether the LAN\WAN interface of the gateway has been connected.( It is connected if the light is bright.) 3、The left one show whether there're data pass throught the interfaces. V600 LED explicate V600 has 2 LED 1、The left one on the right indicates whether the gateway has registered to the SIP server. （The red light flashes if it doesn't register, it doesn't flash on the contrary. ） 2、The right one show whether there're data pass throught the interfaces.

i、 DHCP mode According to the 2.2, enter the menu ‘01’ option by voice hint function, set to DHCP mode. ii、Static IP mode According to the 2.2, enter the menu ‘01’ option by voice hint function, set to static IP mode, and set the IP address of ATA, subnet mask, and the IP of gateway in the option of ’02 03 04’. II、Configuring With Web Browser ATA series products the inside Web server, it can respond the HTTP GET or POST request from Web Browser, and it has the inside HTML page layout, it allow the user setting by the Web Browser of the Microsoft’s IE and AOL’s Netscape. The collocation menu of ATA can be visited by WAN or LAN port.  The default gateway IP address of LAN port is：  http://192.168.2.1  Get the IP of WAN port: Press ‘***’ or press the button of ATA equipment to enter the voice menu Press ‘02’ can get the IP in English  Use the WAN port to enter the setting Web page, must change the option ‘Enable Web Access’ to ‘yes’. Then can enter the Web page by WAN port.

Open the IE browser, input WAN IP address, will appear the following page: The password must case sensitive, and default is ‘voip’. Input the correct password in the login page, the Web server of ATA will appear setting menu. The following is the definition of the parameter in the setting menu III、status MAC Address The device ID, in HEX format. This is a very important ID for ISP troubleshooting. WAN IP Address This field shows WAN port IP address. Software Version Program: This is the main software release; its number is always used for firmware upgrade. Registered Status This field indicates whether the device is registered to the SIP server. PPPoE Link Up This field shows whether the PPPoE connection is enabled or not. NAT State This field shows what kind NAT the HT is connected to via its WAN port. It is based on STUN protocol.

since the last reboot. IV、BASIC OPTIONS i、LAN Settings LAN Subnet Mask Sets the LAN subnet mask. Default value is 255.255.255.0 LAN DHCP Base IP Base IP for the LAN port which functions as a Gateway for the subnet. Default value is 192.168.2.1 DHCP IP Flash Time Value is set in units of hours. Default value is 120hr (5 Days.) The time IP address are assigned to the LAN clients ii、WAN Settings

IP Address There are 2 modes under which the V600 ata can operate: If DHCP mode is enabled then all the field values for the static ip mode are not used,The V600 will acquire its IP address from the first DHCP server it discovers from the connected. To user the PPPOE feature the PPPOE account settings need to be set, the V600 will attempt to establish a PPPOE session if any of the PPPOE field is set If static IP mode is enabled then IP address 、Subnet Mask、 Default Router IP address、 DNS Server1、DNS

reset to zero by default. Cloned WAN MAC Addr Allow the user to set a specific MAC address. Set in Hex format Web server port Default for HTTP is 80 WAN ICMP safety Unit will not respond to PING from WAN side if set to “Yes” iii、NAT Settings Device Mode Default to Router mode ,or set up to bridge mode as the connection of two Ethernet ports. DMZ IP Forward all WAN IP traffic to a specific IP address if no matching port is used by V600 itself or in the defined port forwarding Port Map Allow the user to forward a matching(TCP/UDP) port to a specific LAN IP address with a specific(TCP/UDP) port

Key pattern to use PSTN line, default is "*00" Time Zone This parameter controls how the displayed date/time will be adjusted according to the specified time zone. Daylight Saving Time This parameter controls whether the displayed time will be daylight savings time or not. If set to Yes, then the displayed time will be 1 hour ahead of normal time. Optional Rule Default is 4.1.7.2.0:10.-1.7.2.0:60 Basic User Password This contains the password to access the Basic Options. This field is case sensitive. V、Call settings(V610 FXO PORT)

connect immediately Choosing yes, The phone switch to voip mode in on time while change now. while choosing no, Number of rings before PSTN incoming call is forwarded. Default is 4. PSTN Access Key This field allows users to customize their own code to access the PSTN line. Default is “*00”. VOIP CALL PSTN KEY Key pattem to authorize calling PSTN numbers from VOIP,no default PSTN CALL VOIP KEY Key pattem to authorize calling VOIP numbers from PSTN,no default Route Call to PSTN If the dialed digits match one of the specified prefix here, outbound calls will be routed to PSTN port. This field is especially useful for emergency calls.

Calls are unconditionally forwarded to the specified PSTN phone number once users dial the FXO port VoIP number. Forward to VoIP Calls are unconditionally forwarded to the specified VoIP phone number once users dial the FXO port PSTN number. V、SUPER OPTIONS i、SIP Settings

SIP Server’s URI or IP address Outbound Proxy SIP Outbound Proxy Server’s URI or IP address SIP User ID SIP service subscriber’s User ID Account ID SIP service subscriber’s Authenticate ID. Can be identical to or different from SIP User ID Authentication Password SIP service subscriber’s account password Display Name SIP service subscriber's name which will be used for Caller ID display Home NPA Default is Black. Use DNS SRV Default is No. If set to Yes the client will use

User ID is phone number If the ATA 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 ATA 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 SIP INFO Safety If set to"YES",the device will ignore any SIP message that does not come from the IP address (Source IP in the IP header) that it is registered to.Default setting is "NO" Register Expiration This parameter allows the user to specify the time frequency (in minutes) the ATA refreshes its registration with the specified registrar. The default interval is 3V600seconds. The maximum interval is about 45 days Local SIP Port This parameter defines the local SIP port the ATA will listen and transmit. The default value is 5060(V611FXO port and V620FXS2 port default 5062) Local RTP Port This parameter defines the local RTP-RTCP port pair the ATA will listen and transmit. The default

port default 5008) Use Random Port This parameter, when set to Yes, will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple ATA’s are behind the same NAT NAT Traversal This parameter defines whether the ATA NAT traversal mechanism will be activated or not. If activated (by choosing “Yes”) and a STUN server is also specified, and then the ATA will behave according to the STUN client specification. Under this mode, the embedded STUN client inside the ATA 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 ATA 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 ATA 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 Connected Interval This parameter specifies how often the ATA sends a blank UDP packet to the SIP server in order to keep the “hole” on the NAT open Use NAT IP NAT IP address used in SIP/SDP message. Default is blank Proxy-Require SIP Extension to notify SIP server that the unit is behind the NAT/Firewall DTMF Send Type This parameter controls how DTMF events are transmitted. There are 3 ways: in audio which

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 Scheme Bellcore (North America) ETSI-FSK (France, Germany, Norway, Taiwan, UK-CCA) ETSI-DTMF (Finland, Sweden Denmark-DTMF CID-Canada DTMF-Sweden DTMF-Brazil Send Anonymous If this parameter is set to “Yes”, the “From” header in outgoing INVITE message will be set to anonymous, essentially blocking the Caller ID from displaying Send Flash Event This parameter allows users to control whether to send an SIP NOTIFY message indicating the Flash event, or just to switch to the voice channel when users press the Flash key Fax Mode T.38 mode and pass-through mode ii、Audio Settings

including G.711A-/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 Codecs in this list can be

‘Choice 1’. Similarly, the last Codecs in this list can be entered by choosing the appropriate option in ‘Choice 7’ G723 Rate This defines the encoding rate for G723 vocoder. By default, 6.3kbps rate is chosen iLBC Frame Size This sets the iLBC size in 20ms or 30ms iLBC Payload Type This defines payload time for iLBC. Default value is 98. The valid range is between 96 and 127 Layer 3 QoS 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 QoS(VOIP) This contains the value used for layer 2 VLAN tag. Default setting is blank Layer 2 QoS(PC) This contains the value used for layer 2 VLAN tag. Default setting is blank Silence Suppression This controls the silence suppression/VAD feature of G.723 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 Sending or Reveiving Volume It can be adjusted the volume in the process Call progress Tones It fits to the signal of different countries and areas.

Dial Plan Sets the dial plan add to dialed number Offhook Auto-Dial This parameter allows a user to configure a User ID or extension number to be automatically dialed upon offhook. Please note that only the user CHAPTER of a SIP address needs to be entered here. The ATA will automatically append the “@” and the host portion of the corresponding SIP address Use # as Dial Key This parameter allows the user to configure the “#” key to be used as the “Send”(or “Dial”) key. Once 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 CHAPTER of the dial string to be sent out

Default is No. If set to Yes, Call Forwarding & Do-Not-Disturb are supported locally Disable Call-Waiting Default is No Use Bell-style 3-way Conference If "Yes",# will be disabled iv、 Other Settings

upgrading software, ATA can set a TFTP server of updating code, use LAN or WAN IP address, and ATA can work in static IP or DHCP mode. Suggestion: TFTP server must WAN IP address or in the same LAN with the ATA NTP server The parameter defines URL or IP address of NTP server, ATA show the current date and time Syslog Server The IP address or URL of system log server.This feature is especially useful for ITSP(Internet Telephone Service Provider) Syslog level Select the ATA to report the log level.Default is NONE.The level is one of DEBUG ,INFO,WARNING or ERROR.Syslog message are sent based on the following events: 1. Product model/Version on boot up(INFO level) 2. Sent or received SIP message(DEBUG level) 3. SLIC chip exception(Warning and error levels) 4. Memory ecxception(ERROR level) 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

some special requirements from Soft Switch vendors like Nortel,Broadsoft,etc Onhook Threshold The amount of time the hookflash is pressed that will cause the device to onhook.Default is 800ms Onhook Voltage Select the onhook voltage to suit different area or PBX Polarity Reversal Select polarity reversal to calculate the cost for some system, default is No Lock Keypad Update If this parameter is set to "Yes",the configuration update via keypad is disabled Super Password Administrator password.Only administrator can configure the "Advanced Settings"page.This field is case sensitive and the maximum password length is 25 characters. V FXO PORTPhone Feature

response Dial Plan Prefix Sets the prefix added to each dialed number Use # as Dial Key This parameter allows the user to configure the “#” key to be used as the “Send”(or “Dial”) key. Once 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 CHAPTER of the dial string to be sent out Offhook Auto-Dial This parameter allows a user to configure a User ID or extension number to be automatically dialed upon offhook. Please note that only the user CHAPTER of a SIP address needs to be entered here. The ATA will automatically append the “@” and the host portion of the corresponding SIP address SUBSCRIBE for MWI If set “Yes”, send periodic SUBSCRIBE for Message Waiting Indication PSTN Silence Timeout Terminate call after long silence detected, default is 60 sec, max 65536 PSTN Disconnect Tone This configuration should be configured by the VOIP service provider.Some country use single frequency tone to signal PSTN disconnection,some country use double

Cadence This setting can be configured to suit the telephone company's standard in different country. Once a change is made, the user should press the “Update” button in the Configuration Menu. The ATA will then display the following screen to confirm that the changes have been saved Users are recommended to power cycle the ATA after seeing the above message The administrator of the ATA can remotely reboot the ATA by pressing the “Reboot” button at the bottom of the configuration menu. Once done, the following screen will be displayed to indicate that rebooting is underway

seconds. CHAPTER 5 Restore Factory Default Setting Warning: Restore the Factory Default Setting will delete all configuration information of the device. Step one: 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 number. Step two: Encode the MAC address. The encode rule is:  “2” is the first letter on the button “2” so its encoding is “2”.  “A” is the second letter on button “2” so its encoding is “22”.  “B” is the third letter on button “2” and its encoding is “222”.  “C” is the fourth letter on button “2” and its encoding is “2222”. For example, for MAC address 000b8200e395, the user should encode it as “0002228200333395”. Step three: Access the voice menu, then dial “99” and get the voice prompt “RESET” Step four: Dial in the encoded MAC address. Once the correct encoded

the factory default setting. Chapter 6 Upgrade The 6xx Series Products support two types of update modes: TFTP and HTTP Attention : Please clear the SIP Server address and Account before remote update; Disconnect to the internet before local update (optional). The whole process will cost at least 10 minutes, in the period the power can never be cut off and any manipulation to the phone is not allowed. The gateway will reboot many times during the update process. In order to avoid physical break to the gateway , do not cut off the power immediately after the reboot. 一、 TFTP UPDATE V6xx Series products can work both in DHCP and Static IP mode. The TFTP server is supposed to have a public IP address or the TFTP server and the V6xx products are in the same LAN. 1 LOCAL UPDATE 1) Set up a TFTP server, put the update software package to the root directory of this server, open the TFTP server. 2) Open the configuration page of the gateway, find the place as the following picture: 1. Please specify the Update Mode as :TFTP 2. Fill the Firmware Server item with the Local TFTP server. 3. Click SaveSet then reboot. 二、 HTTP UPDATE V6xx Series products can work both in DHCP and Static IP mode. The HTTP server is supposed to have a public IP address or the TFTP server

1 LOCAL UPDATE 1) Set up a HTTP server, put the update software package to the root directory of this server, open the HTTP server. 2) Open the configuration page of the gateway, find the place as the following picture: 1. Please specify the Update Mode as :HTTP 2. Fill the Firmware Server item with the Local HTTP server. 3. Click SaveSet then reboot. HFS—HTTP Server 2.0a is commended for remote software. 2 REMOTE UPDATE Please contract us to get a TFTP Server address if you need remote update. CHAPTER 7 Ordinary Quality Checking  Bobbling the ATA, no exceptional noise。  Use the standard electrical source, connect it without any other equipment,If the electrical source is OK, you will see: the red LED light will on, after a sound 'pa', it will shine slowly. CHAPTER 8 FAQ 、I can open ‘Status’ and ‘Basic setting’, but cannot open the ‘Super options’, why?  Because when you input the password, you make a mistake. There are two password, one is super user password (default is ‘admin’), it can open all the pages. The other one is user password (default is ‘123’), it cannot open the ‘Super options’ page.

 First, checking the parameter on the configuration pages, whether there is a parameter has been set as ‘*’ or ‘**’ or ‘***’, if have, please change to others’  Then, ‘Offhook Auto-Dail’ parameter must be empty. 、The ATA got an IP, but I cannot open the configuration pages, why?  First, make sure that your computer and the ATA connect the NET.  Then, make sure that your computer and the ATA are in the same segment.  Last, make sure that the phone connected to the ATA is onhook. 、I can visit Internet when I make a VOIP call, why?  It’s always caused by the bandwidth. Please change the Voice Code, so that it cost less bandwidth. 、When I press Speaker or make a call, there is some cacophony, why?  First, make sure that all the equipments are OK, then, debug the impedance and onhook voltage of FXS port. 、The ATA cannot get IP automatically, why?  First, check the way of connecting; make sure it’s no problem,  Then, visit the configuration through LAN port, make sure all the equipments in the LAN are use DHCP. 、The IP address of the ATA is 0.0.0.0, why?  Main problem: some thing wrong with the connecting, or there is no DHCP in the LAN. 、Cannot visit the configuration pages through WAN port, why?  First, the option on configuration pages called ‘Enable WAN Web Access’, when you choose ‘no’, you cannot visit the configuration pages through WAN port.  Then, checking the connection, make sure it is no problem; click Start then Run, input ‘ping (WAN IP) –l 65000 –t’ order, it can help you to check the status of the NET. 、Use PPPOE, the ATA can make a VOIP call, but cannot connect the

、Cannot visit the configuration pages through LAN port, why?  Make sure that all the connecting is OK,  Pull out the connection to the WAN port (if the SIP Account is invalidation, but the connection to the WAN port is OK, it will make the ATA power off automatically.  Set the computer as DHCP, then reboot the ATA  Check the default gateway, the default gateway of the computer is the LAN address of ATA