3.Specification of connector................... 9 2. Software Features and Specification ..............11 3. Physical specification & Environment ..............13 Chapter 2 Configuring the SIP-Phone through LCD Phone menus ........14 1. Initialize SIP-Phone ....................14 2. LCD Menu Configuration ..................16 Chapter 3 Configuring the SIP-Phone through Web Pages ..........
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LP_SIP_100a 2004/04/07 Sabrina lp101/201sip.100 For first formal released software. LP_SIP_101 2004/12/1 Sabrina lp201sip.101 Modify: 1. Forward function 2. Call list function 3. LCD menu firmware upgrading section. 4. Add DDNS function 5. Add VLAN function SIP SIP-Phone Administration Guide...
It provides internal high-quality speakerphone, programmable keys and feature buttons. SIP-Phone also embedded with a dot matrix of two lines 24 characters LCD, which can display date and time, calling party name, calling party number, and digits dialed and etc.
Direct Line (DL) Button 1 – 10: User press DL button after off-hook to do speed dial according to phone book data from 1-10 (please refer to LCD configuration-3.Phone Book; Advanced Configurations via Telnet- 10.[pbook] command, or Web Configuration-Phone Book chapter. SIP SIP-Phone Administration Guide...
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MUTE: Mute the voice of MIC and let others can’t hear from user in communication. ♦ PSTN: Press PSTN to switch SIP-Phone as PSTN or IP Phone Mode. In PSTN mode, “PSTN” characters will be displayed on LCD left bottom side, then users can dial out as if standard telephone set in PSTN;...
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Note: 1. When SIP-Phone is in PSTN mode, only PSTN and SPEAKER function key can work. 2. On LCD will display ”…Incoming Call... ” to inform user when SIP-Phone has both IP and PSTN side incoming calls. 3. If in communication with IP side, user can press HOLD to hold IP side, then press PSTN to pick up PSTN side, after that can press HOLD again to retrieve IP side.
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1. All function keys mentioned above (except dialing keypad) are effective only in IP Phone mode. 2. When LAN Phone fail to register to Proxy server under Proxy mode, when user wants to dial out, LAN Phone will play busy tone, and on LCD will display “Register Failed.” SIP SIP-Phone Administration Guide...
There are two LED indicated lights: LINK/ACT and 10/100 for LAN port and PC port. When network status is regular, LED of LINK/ACT will light on; when SIP-Phone is transmitting or receiving data, LED will be flashing; when transmit rate is in 10 mbps or 100mbps, LED of 10/100 will light off or light on.
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2¡ B RJ-11connector¡ G RJ-11 connector is for connecting SIP-Phone with PSTN. RJ11connector SIP SIP-Phone Administration Guide...
Ten configurable speed dials Network Supported Fixed IP Dynamic Host Configuration Protocol (DHCP) PPPoE connection (When PPPoE disconnect, SIP-Phone can automatically re-connect) Behind NAT IP Sharing Device Support QOS by setting DSCP (Differentiated Service Code Point) parameters of VoIP packet Support 802.1p1q (VLAN)
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Ring tone, Speaker and Handset volume adjustable Dial path selection (PSTN or IP mode) Support DNS server inquiry Management Features: Software Upgrade: TFTP/FTP download Three easy ways for system configuration LCD Front Panel Web Browser TELNET Certification CE, 3C SIP SIP-Phone Administration Guide...
Chapter 2 Configuring the SIP-Phone through LCD Phone menus Note: 1. After any configuration is made for the SIP-Phone, user has to do Reboot in the selection 7 “Reboot”. 2. We suggest user to set IP address via LCD menu 5¡ ÷ 2.3.4.5 first, then go to chapter 3 to do other detail configurations via web browser.
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5. After pressing the PSTN button, the “Proxy” or “P2P” will be replaced by “PSTN”. Please notice that user must plug PSTN line in RJ-11 port when SIP-Phone is in PSTN mode. LAN Phone will always stay in IP mode, after a PSTN call is finished, LAN Phone will automatically return to IP mode..
There are 3 selections in Forward type, user must select under which condition to forward calls. (1) Busy When SIP-Phone is in busy status, the incoming call will be forwarded to the assigned phone number. A. Activate Enter a forwarded phone number to activate busy forward function.
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Style There are three tone styles for SIP-Phone. Move the “>” symbol by press on the keypad to select the tone style preferred, then press OK to confirm it. SIP SIP-Phone Administration Guide...
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(2) IP, Mask, Gateway: display current IP information. Network Mode Set network mode of SIP-Phone to be Static (Fixed IP), DHCP, or PPPoE. IP address Set IP address of SIP-Phone. Subnet Mask: Set subnet mask address of SIP-Phone.
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SNTP Configuration (1) SNTP Mode: User can set SNTP function to be on or off, which means SIP-Phone will capture current time from SNTP server or not. (2) SNTP Server: User can specify a SNTP server for SIP-Phone to capture current time.
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Set expire time of registration, in the duration of 2/3 expire time, SIP-Phone will re-register to Proxy Server again. (3) User Info A. User Name Set User Name of SIP-Phone to register to Proxy Server. If Proxy server doesn’t request specific User name, please enter Line number here. B. Line Number Set Line Number of SIP-Phone to register to Proxy Server.
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LCD Menu Password Set entry password of phone LCD menu. 7. Reboot Reboot machine. It is necessary and important for user to reboot SIP-Phone after any configurations has been made. SIP-Phone will ask user again before reboot. SIP SIP-Phone Administration Guide...
The configuration function and steps are similar with the way through command line. Please refer to the chapter 4-Configuring the SIP-Phone through Telnet command lines for more detail information. Below is a guide for user to configure via web interface.
Step 1. Browse the IP Address predefined via Keypad Please enter IP address (user have to set via LCD menu first) of SIP-Phone in web browser. If user failed to set IP address via LCD menu, the default IP address of SIP-Phone is 10.1.1.3, user can try to connect to SIP-Phone via this default IP via web interface.
Step 2. Input the login name and password Login name: root / administrator Password (The same with TELNET): Null (just press confirm, no need to key in password in default value) Note: User can set password later in 8.PASSWORD via web interface. SIP SIP-Phone Administration Guide...
Phone Book (in Peer-to-Peer mode). Please remember to configure these commands before start to work with SIP-Phone. Note: After change any settings, please remember to reboot (in Reboot System) SIP-Phone so that changes can take effect. SIP SIP-Phone Administration Guide...
When DHCP function enables, SIP-Phone will automatically search DHCP server after reboot. 3. PPPoE: If SIP-Phone is working with PPPoE connection, user have to set related parameters in “PPPoE Configure “page. Note: If User set “Get IP mode” as DHCP or PPPoE, IP address, Subnet Mask, and Default routing gateway will become 0.0.0.0 and not allow to be configured.
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Enable / Disable the Simple Network Time Protocol function SNTP Server Address: Set SNTP Server Address When SNTP server is available, enable SIP-Phone SNTP function to point to SNTP server IP address so that SIP-Phone can get correct current time. GMT: Set time zone for SNTP Server time User can set different time zone according to the location of SIP-Phone.
Phone Book Search: enable/disable phone book search function. If user enabled this function, SIP-Phone will search dialed number in phone book to see if there is any matched table before send to Proxy server, and if there is a matched data in phone book, SIP-Phone will make call to related IP address.
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For example, if Account Name is 123, SIP-Phone will sent out messages as Account Name @”IP address of Proxy”, if user set prefix as abc, SIP-Phone will set out as abc123@”IP address of Proxy”. This function is for special proxy server.
Set the DTMF inter digit time (second) To set the duration (in second) of two pressed digits in dial mode as timed out. If after the duration user hasn’t pressed next number, SIP-Phone will dial out all number pressed. End of Dial Digit: select end of dialing key, e.g.
SIP-Phone still can’t get contact with server, SIP-Phone will keep trying to connect. After re-connected, SIP-Phone will also restart system. On the other hand, if user disables this function, SIP-Phone won’t reboot and keep trying to connect.
Buffer: Set Min. Delay and Max. Delay of Jitter Buffer for voice packets. Optimized Factor (Jitter): Set Optimized Factor of voice, this is for advanced user only, please contact with your distributor before making any change. SIP SIP-Phone Administration Guide...
User can specify 20 sets of phone book via web interface. Please input index, Name, IP Address and E.164 number of the destination device. Delete Date: User can delete any configured phone book data by index. SIP SIP-Phone Administration Guide...
IP address check server IP Check Server 2: to set secondary IP address check server Check every /minutes /hours /off: to set the update interval time. LAN Phone will re-update its IP address in this time. SIP SIP-Phone Administration Guide...
Type-of-Service (TOS) octet in IPv4. The recommended DiffServ Code Point is defined in RFC2597 to classify the traffic into different service classes. The mapping of Code Point value of DS-field to egress traffic priorities is shown as follows. SIP SIP-Phone Administration Guide...
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100010 Medium Drop Precedence (AF12) (AF22) (AF32) (AF42) 001100 010100 011100 100100 (AF13) (AF23) (AF33) (AF43) High Drop Precedence 001110 010110 011110 100110 Please refer to RFC standard documents for more information about what is DSCP. SIP SIP-Phone Administration Guide...
Please refer to chapter 4.19 [password] command Change: First select login name as root or administrator, then enter current password, new password and confirm new password again to set new password. Abort: Press abort will clean all inputs. SIP SIP-Phone Administration Guide...
Select download method as TFTP or FTP FTP Login: Set FTP login name and password Target File Type: Select which sector of SIP-Phone to upgrade Note: 1. After 2mb file download is finished, all configurations might change to default values, user has to configure again.
11. Flash Clean Please refer to 4.15 [flash] command. Press CLEAN will clean all configurations of SIP-Phone and reset to factory default value. Note: User must re-configure all commands all over again (except Network Configure) once execute this function, SIP SIP-Phone Administration Guide...
12. Reboot System Press reboot will reset SIP-Phone. Note: To execute reboot via web browser, SIP-Phone will automatically save all data before reboot. To execute reboot via TELNET command, please remember to do Commit Data before Reboot System. SIP SIP-Phone Administration Guide...
Chapter 4 Configuring the SIP-Phone through Telnet command lines After setting the IP Address of SIP-Phone and reboot, (please refer to LCD Menu: 5-3.4.5), user can enter into Telnet command lines. Note: 1. After user enter SIP-Phone configuration via telnet, please use login: ”root”, password: null, press enter to enter command lines.
Type quit/exit/close will logout SIP-Phone and Telnet Program. 3. [debug] command This command is for engineers to debug system of SIP-Phone. User can add debug flag via command debug –add “debug flags”, and then start debug function via command debug –open. When SIP-Phone is working on screen will display related debug messages.
(pbook –modify “index” name “X” ip “xxx.xxx.xxx.xxx” e164 “X”) 6. [commit] command Save any changes after configuring the SIP-Phone. 7. [ping] command Command ping can test which the IP address is reachable or not. Usage: ping “IP address”...
When User set IP mode to be fixed IP, please set IP, subnet Mask, default gateway as mentioned in item 2. If User set IP mode to be DHCP, SIP-Phone will search for DHCP server to capture IP address after reboot.
5. -ipsharing: If SIP-Phone is behind an IP-sharing, user must enable IP sharing function and specify public IP address. (ifaddr –ipsharing 0/1 “public IP address of IP sharing”, 0 for disable and 1 for enable) Note: Some Proxy servers support endpoint behind NAT function, in this case SIP-Phone doesn’t have to enable IP sharing function, please contact with your Proxy Server...
1. -print: display all current configurations. 2. -idtime: set the duration (in second) of two pressed digits in dial mode as timed out. If after the duration user hasn’t pressed next number, SIP-Phone will dial out all number pressed. 3. –service: set LAN Phone to work as normal mode or hotline mode. If user wants to use hotline function, please set service to be 1, and set bureau related configuration.
7. –service: set SIP Phone to be normal mode or under hotline mode. (sysconf –service 0/1, 0 for normal service, 1 for hotline service.) Note: To set SIP-Phone as hotline mode, user has to set SIP-Phone as Peer-to-Peer mode, and hotline table under bureau command. 13. [sip] command 1.
9. –expire: set expire time of registration. SIP-Phone will keep re-registering to proxy server before expire timed out. 10. –port: set listening UDP port or SIP-Phone. 11. –rtp: set RTP port number. SIP-Phone will use this port to send and receive voice. 14. [security] command 1. –print: display all current configurations.
3. -priority: set codecs priority in order. Please notice that user can set from 1 to 5 codecs as their need, for example, voice –priority g723 or voice –priority g723 711a g711u g729 g729a means SIP-Phone can support only one codec up to five codecs.
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3. –siptype: set DSCP value of RTP packets from 0 to 63 Note: 1. This command won’t be functional until network environment can be capable with DSCP function. 2. tos -rtptype 14 -sigtype 10 is top priority of package. SIP SIP-Phone Administration Guide...
(ddns –delay 1-59m/1-24h , m means minute, h means hour) 10. –force: to force to execute DDNS update. Once user enters this command, LAN Phone will update DDNS data immediately. (ddns –force “IP address of LAN Phone”) SIP SIP-Phone Administration Guide...
6. –pcpriority : set pc port priority number. (From 0 to 7, 7 is the highest priority.) 7. –pcdroptag: enable/disable pc port drop tag function. If this function is enabled, LAN Phone will drop priority tag on packets sending out from PC port. SIP SIP-Phone Administration Guide...
1. -print: show all current configurations. 2. -hotline: set hotline table. User can set hotline function to specify one IP address for SIP-Phone to dial out directly. Once user picks up SIP-Phone, it will automatically dial out to the assigned IP address and number.
7. –ftp: specify user name and password for FTP download method. For example: User prepares to upgrade the latest app rom file – siplp.100, the TFTP server is 192.168.4.1.1. rom –app –s 192.168.1.1 –f siplp.100 Command rom –print can show current version installed in SIP-Phone . SIP SIP-Phone Administration Guide...
All methods are necessary to prepare the TFTP or FTP program on the host PC as TFTP/FTP server. After installing TFTP/FTP program on one PC and connecting to network, SIP-Phone is ready to be upgraded.
(7) Start to Upgrade Select YES or NO to start upgrade. After download is finished, press OK then SIP-Phone will ask if need to reboot. 2. Web Management Please refer to chapter 4.18 [rom] command FTP/TFTP Server IP Address: Set TFTP server IP address...
6. –method: specify download method to be TFTP or FTP(0 for TFTP.1 for FTP) 7. –ftp: specify user name and password for FTP download method. For example: User prepares to upgrade the latest app rom file – siplp.100, the TFTP server is 192.168.4.1.1. rom –app –s 192.168.1.1 –f siplp.100 SIP SIP-Phone Administration Guide...
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Command rom –print can show current version installed in SIP-Phone . SIP SIP-Phone Administration Guide...
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