User guide

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Room 1004, Building 2, Phoenix Building, 2008 ShenNan Ave., Futian District,Shenzhen ,China

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GP1260

SIP Phone

User Guide

Summary of content (41 pages)

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GP1260 SIP Phone User Guide Room 1004, Building 2, Phoenix Building, 2008 ShenNan Ave.
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INDEX 1. 2. PRODUCT OVERVIEW ......................................................................................... 4 FEATURES AND SPECIFICATION ....................................................................... 4 2.1. Hardware Features.......................................................................................... 4 2.2. Software Features ........................................................................................... 5 2.3. Standard and Protocol...............................
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6.2.1. Answering a call ................................................................................. 31 6.2.2. Call Hold ............................................................................................. 31 6.2.3. Call Waiting ........................................................................................ 31 6.2.4. Switch between two calls.................................................................... 32 6.3. Call Transfer........................................................
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1. PRODUCT OVERVIEW IP is the acronym for Internet Protocol. An IP phone is a telephone which transmits voice over network based on IP protocol, for example LAN (Local Area Network), MAN (Metropolitan Area Network) and INTERNET. The most significant feature of IP phone is transporting voice over data communication network with almost no extra cost and comparable voice quality and various supplementary services. Using IP phone, you will save tremendously on international calls and long distance calls.
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● Program Memory—2 MB Flash memory ● Display LCD—128 X 64 Graphics Dot. ● Ethernet Port—2X 10/100M Connectors 2.2. Software Features ● DHCP support for LAN or Cable modem ● PPPoE dialup ● Built in web server for configuration and upgrade ● On phone menu for configuration through keypad ● Automated provisioning of firmware and configuration via HTTP ● Manual upgrade firmware via HTTP; ● Multiple audio codec support: PCMU;PCMA;G.726-32;GSM6.10;iLBC;Speex;G.
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● Call Forwarding – Unconditional, No Answer, On Busy ● Auto-Answer ● Do-Not-Disturb ● Configurable Dial Plan Support ● Phone Book with 100 entries ● Speed Dialing Support (10 entries) ● Call History: Missed, Answered, Dialed Numbers with call time (10 each) ● Adjustable volume for handset, ring volume and speaker etc. ● Hot Line Calling 2.3. Standard and Protocol ● SIP (RFC2543; RFC3261;RFC3262 ; RFC2976 ; RFC3264 ; RFC3311 ; RFC3515 ; RFC3666 ; RFC3420 ) ● IEEE 802.3 /802.
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● DHCP：Dynamic Host Configuration Protocol ● DNS：Domain Name Server ● TFTP：Trivial File Transfer Protocol ● HTTP：Hyper Text Transfer Protocol ● SNTP：Simple Network Time Protocol ● Syslog：The BSD syslog Protocol 2.4. Electric Requirements ●Voltage: 9V DC ●Power: 4.5W (max.) 2.5. Size 238 x 188 x 70 mm (L x W x H) 2.6. Operating Requirements ●Operation temperature: 0 to 45° C (32° to 113° F) ●Storage temperature: -30° to 85° C (-22° to 185° F) ●Humidity: 10 to 90% no dew 3. INSTALLATION 3.1.
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all items. If any item is not included in the package, please contact the distributor. 1) One GP1260 SIP Main Case 2) One Handset 3) One Universal Power Adapter 4) One product qualification and guarantee 5) One User Guide CD-ROM 6) Quick Simple User Manual 7) One RJ-45 Ethernet cable 3.2. Connecting the Phone 1. Connect handset to base: insert the handset cord into a handset cord jack on the left side of the base. 2. Connect the phone to Network: plug the RJ-45 Ethernet cable into the Ethernet Jack.
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● GP1260 SIP Phone Front Illustration (Refer to Fig 4.1.1)： Power LCD Display Network1 Network2 LED LED LED Mute/ DND Speed dial Transfer M1---M10 Hold VOL+/UP Message Right Menu Left VOL-/Down Cancel Redial Speaker OK Flash Call Fig 4.1.1 GP1260 Phone Front ● GP1260 SIP Phone Back Illustrations(Refer to Fig 4.1.2 ) Network connect 1 Network connect 2 Power Jack Fig 4.1.2 GP1260 SIP Phone Back Room 1004, Building 2, Phoenix Building, 2008 ShenNan Ave.
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4.2. Function Keys Keys Function Keys MENU M1-M10 Speed-Dial key, each corresponds to a speed dial number which will be called by a single press. UP(▲) Function When phone is idle, press this key to enter MENU mode. Increase the output volume of handset or speakerphone. In MENU mode, press this key to scroll up. DOWN(▼) Decrease the output volume of handset or speakerphone. In MENU mode, press this key to scroll down menu options. ▲ ) LEFT( ▼ ) RIGHT( In EDIT status, press this key to backspace.
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Off: Network cable is disconnected or network error. POWER LED MESSAGE LED On: the power supply is all right. Off: the power supply is disconnected. HOLD LED On: There are unread voice messages. Off: No new voice message. MUTE/DND LED On: The call is on hold . On: Local voice is muted when in a call; Do-Not-Disturbed is turned on when phone idle. 5. CONFIGURATION GUIDE After the phone is properly installed, users can use keypad or web browser to configure its parameters. 5.1.
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Fig 5.1.1 Main Menu and Phone Settings Menu Structure 2. Scroll the menu Press UP（▲）, DOWN（▼） key to scroll through menu items. 3. Enter the submenu Press OK key to enter the next level of the menu. 4. Edit and confirm Press OK key to enter the edit mode, when current menu item has no submenu. A cursor will appear in this mode. Press OK key to confirm the input. 5. Delete a character and move cursor Use LEFT key to backspace and RIGHT key to shift cursor right in edit mode. 6.
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Or press CANCEL key, the phone will abandon all modifications and stay in menu mode. 5.1.2. Viewing System Info Step 1: With the phone on-hook, press MENU key to enter the main menu. Step 2: Press UP（▲）, DOWN（▼） key to scroll to System Info submenu. Step 3: Press OK key to enter the System Info submenu. Step 4: Use UP（▲）, DOWN（▼） key to scroll through the basic information of the phone, including IP Address, Phone Number, MAC Address, Phone Model, Protocol , Language , OEM Tag and Version .
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5.1.3. Network Settings Fig 5.1.2 Network submenu Please check the Web Configuration 5.2.2 for details about these items. 5.1.4. Voice Settings Fig 5.1.3 Voice submenu Room 1004, Building 2, Phoenix Building, 2008 ShenNan Ave.
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Please check the Web Configuration 5.2.3 for details about these items. 5.1.5. Protocol Settings Please check the Web Configuration 5.2.4 for details about these items. Fig 5.1.4 Protocol submenu Room 1004, Building 2, Phoenix Building, 2008 ShenNan Ave.
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5.1.6. Dialplan Settings Fig 5.1.5 Dialplan submenu Please check the Web Configuration page 5.2.5 for details about these items. 5.1.7. System Settings Fig 5.1.6 System submenu Please check the Web Configuration 5.2.6 for detail about these items. Room 1004, Building 2, Phoenix Building, 2008 ShenNan Ave.
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5.2. Configuration with Web Browser Open a web browser and input the IP address of the phone into address bar (default IP: 192.168.1.200). Then put password of the phone into the following page. Default password is empty. Fig 5.2.1 http settings NOTE: Make sure that the phone is in idle mode when viewing or setting GP1260 SIP Phone with web browser. 5.2.1.
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Fig 5.2.2 Connection Type IP Address Subnet Mask Default Gateway PPPoE User ID Network Setting Static IP：Select this item to let users to set IP address, subnet mask and router IP address manually. DHCP：Have IP address and other network parameters assigned by the DHCP server. Set IP address of the phone manually when Connection Type is set to Static IP. Set the subnet mask of the network when Connection Type is set to Static IP.
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PPPoE User PIN Set the User PIN of the PPPoE when Connection Type is set to PPPoE. Automatically Get DNS Server IP Have DHCP server assign the IP address of DNS server automatically. Use following DNS Server IP Set the IP address of DNS server manually. Primary DNS Secondary DNS Set the IP address of the first DNS server. Set the IP address of the second DNS server. Layer 3 Qos When Layer 3 Qos is adopted, fill in the Precedence value of IP frames.
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Preferred Voice Codec GP1260 supports up to 7 different Voice Codec types including PCMU, PCMA, G.726-32, GSM 6.10, iLBC, Speex, G.729. Selecting Null disables the corresponding voice codec. Voice Frames Set the number of voice frames transmitted per packet. The suggested number is below 3(including 3) in order to reduce delay. The maximum allowable value is 8. The default value is 1. per Tx iLBC Frame Size Speex rate Select the frame size of iLBC codec: 20ms or 30ms.
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Enable/disable registration with SIP server. To make calls SIP Registration through SIP Proxy Server, please check this box; otherwise, only IP to IP calls is allowed. SIP Server SIP Server Port SIP Domain SIP Server as Outbound Proxy Fill in the IP address or URI of SIP Proxy Server. Fill in the port of SIP Proxy Server. The default value is 5060. Fill in the domain name of the SIP Proxy Server. Enable/disable Outbound proxy.
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5.2.4. Advanced Protocol Settings Fig 5.2.5 Advanced Protocol Settings Fill in the local port registered with SIP server. The phone will Local SIP Port send and receive SIP messages from this port. The default value is 5060. Local RTP Port Fill in the local port to send and receive for RTP. This is an even number between 1024 and 65535. The default value is 6000. Set the interval of refreshing registration with SIP Proxy Server in Register expiration seconds.
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on the NAT device active. Default is 20 seconds. Send DTMF Select the scheme used to send DTMF signals, including inband audio, rfc 2833 and sip info. Fill in the RTP payload type value of DTMF event. The range of DTMF payload type this value is 96-127. Default is 101. G726-32 payload type Fill in the RTP payload type value of G726-32 codec. The range of this value is 96-127. Default is 111. iLBC payload type Fill in the RTP payload type value of iLBC codec. The range of this value is 96-127.
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NAT IP STUN sever STUN server port With NAT Traversal set to use NAT IP, fill in WAN port IP address of NAT device here. With NAT Traversal set to use STUN server, fill in IP address or URI of STUN server here. Fill in the service port of STUN server. Default is 3478 5.2.5. Dialplan Settings Fig 5.2.6 Dialplan Settings Forward-to Number Enter the number to which you want to forward the call. Forward Unconditionally Enable/Disable unconditional call forwarding.
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busy. Forward when no answer Enable/disable call forwarding when no answer. If set to yes, the incoming call will be forwarded to Forward-to Number when this call is not answered within a certain period of time - No answer timeout. No answer timeout Set the time in seconds before the phone answer the call automatically or forward the calls to another party. Auto Answer Enable/disable auto answer.
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5.2.6. System Settings Fig 5.2.7 System Setting Administration Password Syslog IP Handset input volume Password to access settings, with the maximum length of 32 characters. Set the syslog server IP address. Set the input volume of the handset, ranging from 0 – 7. Handset output volume Set the output volume of the handset, ranging from 0 – 31. Speaker output volume Set the output volume of the speaker, ranging from 0-31. Ring volume Set the ring volume of the speaker, ranging from 10-31.
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Auto-provisioning Server Auto-provisioning port Auto-upgrade interval Fill in the URI or IP address for the auto-provisioning server. Fill in the port of the auto-provisioning sever. Default is 80 (we use HTTP to do auto-provisioning). Set the interval of auto-upgrade in minute. The maximum value is 65535 minutes. SNTP Server Fill in the URI or IP address of the SNTP server. Time Zone Select the time zone in list box with user location.
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5.2.8. Digit Maps Fig 5.2.8 Digit Map Digit map is a set of rules to determine when the user has finished entering digits. With digit map, users don’t have to press "call" key after dialing. Fig 5.2.8 gives an example of digit map. X represents any number between 0 and 9. 13xxxxxxxxx: Any 11 digits number starting with 13. 013xxxxxxxxx: Any 12 digits number starting with 013.
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X.T: Any digit number. The number is sent out in T seconds after user dialed the last digits. X. [T#*]: Any digit number ended with * or # or after T seconds of waiting. If the number entered matches an item in the digit map perfectly, or it doesn’t match any item at all, this number will be sent out immediately. NOTE About the detail of digit map, refer to RFC3435 2.1.2. 6. USING GP1260 SIP PHONE 6.1. Placing Phone Calls 6.1.1.
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2. Dial the desired number or IP address (Press * for“. “). 3. Press CALL to complete the call. To make a direct IP address call, disable the” SIP Registration” option. 6.1.3. Redial 1. Pick up handset or press SPEAKER key. 2. Press REDIAL key to dial the last call. 6.1.4. Call from CALL HISTORY 1. Press MENU key. Scroll to “Call History” and press OK key to select the desired call record type: “Missed Calls”, “Received Calls” and “Dialed Numbers”. 2.
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3. Scroll to “Call ?” item and press OK key to call this contact. 6.1.6. Speed dial When phone is in off-hook or speaker-phone mode, press speed dial key (M1-M10) to call the number associated with each speed dial key. When phone is in idle mode, press M1-M10 to view the name and number information of each speed dial entry. 6.2. Answering calls 6.2.1. Answering a call Pick up the handset or press the SPEAKER key to answer a call. Put down the handset or press the SPEAKER key to hang up. 6.2.2.
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current call, press FLASH key to place the current call on hold and answer the new incoming call. 6.2.4. Switch between two calls Press FLASH key to switch between two calls. You can also press the hook to end current call and retrieve the on-hold call. 6.3. Call Transfer 6.3.1. Blind Transfer During a call, press (TRANSFER) key to place the current call on hold and obtain the dial tone. Enter the number to which you want to transfer the call.
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dial tone. Enter the number to which you want to transfer the call and press CALL key to make a consulting call. After consulting with the third party, Press TRANSFER key to complete the transfer. If the third party is busy or nobody answers, press FLASH key or hook once to resume the original call. 6.4. Call Forward 6.4.1. Forward Unconditionally When Forward Unconditionally is set to yes, all incoming calls will be forwarded to Forward-to Number. 6.4.2.
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period of time (defined by No Answer Timeout), the new call will be answered automatically which means the speaker phone will be automatically turned on. 6.6. Mute During a call, press MUTE key to mute local voice. The mute LED will light up, indicating that the other party cannot hear you. Press MUTE key again to resume the conversation, the mute LED will turn off. 6.7. Do Not Disturb (DND) When phone is on-hook, press MUTE DND status.
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1. Press MENU key. Scroll to “Call History” and press OK key to enter. 2. Select the desired call record type from “Missed Calls”, “Received Calls”, and “Dialed Numbers”. 3. Delete all call records：Press OK key and scroll to “Delete ALL?” item. Then press OK to delete all call records. 4. Save a call record: 1). Press OK key and scroll to “Add to Phonebook” item. 2). Press OK key to confirm. 3). Press CANCEL key twice. “Save Phonebook?” will show up. 4). Press OK to save the phone book. E.g.
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the modifications, or press CANCEL key to reject all modifications. 6.10.2. Add a contact 1. Press MENU key. Scroll to “Phone Book” and press OK key to enter. 2. Scroll to “—End of List—“. Press OK key to select “Add Newly?” item. Or when any contact is displayed on LCD, press OK key and scroll to “Add Newly?” item. 3. When “Enter name” shows on LCD, enter the name through the keypad. Press OK key to confirm. Then “Enter Number” will show up on display. Enter the number through the keypad.
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Use UP key or LEFT key to delete unwanted input. Press OK key to confirm. Then “Enter number” item will be displayed on LCD, enter the phone number through keypad. Press OK key to confirm. 4. Repeat step 2 and 3 to modify more contacts. E.g. Edit Aimee (82378008) to Kingon (100083) 6.10.4. Delete a contact entry 1. Press MENU key. Scroll to “Phone Book” and press OK key to enter. 2. Scroll to the desired contact and press OK key to select the action to be taken next. 3.
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2. Press OK key to select the action to be taken next. 3. Scroll to “Delete All?” item and press OK key to confirm. E.g. 7. FIRMWARE UPGRADE 7.1 Manually upgrade Select Upgrade item in the login page and enter the upgrade page below. Fig 7.1.1 manually upgrade 7.1.1. Select upgrade item The available options include Firmware, Settings, Phonebook, Ring tone and Room 1004, Building 2, Phoenix Building, 2008 ShenNan Ave.
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Call Hold Music. 7.1.2. Locate upgrade file Use“browse……”button to select the upgrade file. As to how to get these files, please refer to AR1688 Develop Manual. 7.1.3. Start upgrade Click “Start Upgrade” button to start the upgrade. NOTE: Upgrade must NOT be interrupted! Upgrading firmware may take a few minutes, Please don’t turn off the power. 8. LOAD AND STORE FACTORY SETTINGS 8.1 Load Defaults Settings 1. Press MENU key. Scroll to “Phone Settings” and press OK key to enter this submenu. 2.
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1. Press MENU key. Scroll to “Phone Settings” and press OK key to enter this submenu. 2. Scroll to “Store Defaults” and press OK key. 3. Will be displayed on LCD. Press OK key to confirm. GP1260 SIP IP phone will reboot and store the current settings to the factory default settings. Room 1004, Building 2, Phoenix Building, 2008 ShenNan Ave.
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Annex: GP1260 IP phone digital-character key map Keys Once Twice Thrice quartic quintic sixth seventh eighth ninth 1 1 . _ - @ ： ; / , 2 2 a b c A B C 3 3 d e f D E F 4 4 g h i G H I 5 5 j k l J K L 6 6 m n o M N O 7 7 p q r s P Q R S 8 8 t u v T U V 9 9 w x y z W X Y Z 0 0 space Room 1004, Building 2, Phoenix Building, 2008 ShenNan Ave.