User Guide VoIP Telephone Model No.: SP5101 Website: http://www.micronet.
TABLE OF CONTENTS 1. INTRODUCTION ..................................................... 5 1.1 Packet Content............................................................................................... 5 1.2 Key Features .................................................................................................. 5 2. GENERAL DEFINITIONS ....................................... 6 2.1 Telephone appearance .................................................................................. 6 2.
.1.2 Dialed Calls .......................................................................................................22 3.1.3 Missed ...............................................................................................................22 3.1.4 Delete All ...........................................................................................................23 3.2 Phone Book .................................................................................................. 23 3.2.
.2 Making Calls ................................................................................................. 36 4.2.1 OFF-HOOK Dialing............................................................................................36 4.2.2 Redial (OFF-HOOK Dialing) .............................................................................36 4.2.3 Pre-Dial ..............................................................................................................36 4.2.
6.3 [debug] command ........................................................................................ 71 6.4 [reboot] command ....................................................................................... 71 6.5 [pbook] command ........................................................................................ 71 6.6 [commit] command ...................................................................................... 72 6.7 [ping] command .....................................
1. INTRODUCTION The Micronet SP5101 VoIP desktop telephone highly integrates DSP/codec system-on-chip solutions to provide the industry’s highest voice quality. For ease-of-use functionality, SP5101 provides well compatibility with many well-known IP-PBX, and many user-friendly feature buttons of conference, call pick up, transfer, Redial, Hold …etc. The simple operation and configuration are most suitable for residential, SOHO, and enterprise applications. 1.
2. General Definitions 2.
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2.2 Key Definition NO KEYPAD 1 OK 2 C FUNCTION 1. Press this key to confirm a menu selection, a setting, or a phone entry. 2. When in menu mode, pressing this key will select the menu selection highlighted. 3. When entering phone setting or menu selection entry, pressing this key will confirm the entry. 1. Cancel for the menu setting or number typing. 2.
NO KEYPAD FUNCTION K7 DND K8 Missed K9 VMS K10 Blind Tran K11 Mute K12 Headset K13 Conf Conference K14 Pick Call Pickup K15 Trans Transfer K16 Redial Redial K17 Hold DND Missed calls Voice Mail Blind transfer Mute Headset mode switch Call hold / Call retrieve This phone could support three methods for the text input, such like this: “123”, “ABC”, “abc”. “123” is for the number inputs, “ABC” is for the uppercase character inputs; “abc” is for the lowercase character inputs.
* # S29 7 S10 8 S11 9 S04 * S17 0 S12 Input method switch key 10
2.4 Keys with “ABC” character mode Users could enter the characters with the uppercase in this mode. The key could only input the special symbols and press more times to pick up the symbols you want. If users press two times on the key character will be E.
2.5 Keys with “abc” character mode Users could enter the characters with the uppercase in this mode. The key could only input the special symbols and press more times to pick up the symbols you want. If users press two times on the key , the input character will be e.
2.6 Menu keys There are four keys to help users enter, exit or pick up the configuration tables for changing. Please check out the following info: Key M3 Functions descriptions Press the right arrow key in the IDLE mode to enter the main menu. Press the left arrow key will help users to return the original table or exit the main menu Press the up arrow key to scroll up configuration items. 1. Press the down arrow key to scroll down configuration items. 2. Enter Phone Book directory.
2.7.3 LCD Display in OFF-HOOK state 2006/05/10 10:10 Network Fail Network fail during the IDLE mode (WAN port detection or get IP failed with DHCP and PPPoE)1 Line 1 Dial… LCD displaying while the phone was in the OFFHOOK state. Line 2 Dial… LCD displaying while the phone was in the OFFHOOK state. 2.7.4 LCD Display in DISCONNECTED state 2006/05/10 10:10 SipPhone SP5101 2.7.5 LCD Display for RINGING state Incoming call… 208 2.7.
2.7.7 LCD Display for Incoming Call state Jaosn 65605 2.7.8 When it received incoming call, it will show incoming call display name and Line number LCD Display in CONNECTED state Line 1 Talking… 00:00:10 During the talking state and running the timer Line 2 Talking… 00:00:10 During the talking state and running the timer Mute… 00:00:10 The timer will still go on while the mute function was enabled during the conversation. Transfer… Hold… Hold the established call. Parking… In Conference…..
presses C button to reject call waiting. 2.7.10 LCD Display for firmware upgrading mode Download… Download… Writing… D l Firmware file downloading mode Writing the firmware after the firmware downloading d Completed… Please reboot Request for the rebooting after the firmware upgrading 2.7.
cursor will be blinking every 500ms just like the example as above: John → John 208 → 208 If the users enter the incorrect info, the LCD will show as following: Invalid Input… 2.8 Editing Display a Cursor Under the characters or digits, the cursor will be displayed and the characters or digits will be blinking. It was shown as “_” with the LCD showing. Users could press the right or left arrow keys to make the cursor move to the right or left side. The blinking time for the on and off will be 500ms.
2.9 LED Display While users power on this phone set, all the LED will be lighted up before the system initializing procedure finished. The function LED will be light while users press the function keys; and will be blink while users press the function keys for twice. The following definitions are for the LED: LED Key Function description LED1 1 Flashing for the incoming call. Flash time : 200ms on; 200ms off LED2 2 Lighting for the OFFHOOK. 1.
hook/speaker button. When having incoming call, press HEADSET button can pick up call and press again can hang up call. 3. When in communication, press HEADSET button can switch voice path to headset.
3. LCD Menu Operation During the menu operation, there is no cursor for the configuration tables selecting. The LCD will show just like this: >Call Records >Phone Book Users could press the up or down arrow keys to move on different item. This sign will guide you to choose the configuration tables. In the main menu, it will exit the menu and back to the IDLE mode if there is no action for 30 seconds.
3.1.1 Received M4 Showing the received calls; pressing the out the other received calls. >Received Calls >Dialed Calls → OK → and M1 button to check >John >12345678 The LCD will show as following while there is no any record for the received calls: No Records!! ● Dial out >John >12345678 >Dial Out >Detail ● → M4 → >John >12345678 → OK → Line 1 Dialing..
entry added, show all the entries in the phone book to verify by users. If users press the button, the flow will be just like the actions in the phone book >John >12345678 → >Dial Out >Detail → M4 → >Dial Out >Detail → → >Detail >Add to Book → → >Dial Out >Detail → → >Detail >Add to Book → M4 → → OK → → OK → Entry name ● Delete >John >12345678 → >Dial Out >Detail → M4 >Add to Book >Delete → OK Delete ok… 3.1.
3.1.4 Delete All This will delete all the records of the received, dialed and missed calls. Delete it? → Delete ok… → 3.2 Phone Book This phone set could support 90 entries of the phone book. Users could add, modify, delete, and dial out all the entries in the phone book. If the name and number had been added in the phone book, the LCD has to show the name if it is the incoming call. In the view mode, the name has to be sorted by the characters.
This is allowed the same number with the different name. For the name of the incoming call displaying, the phone set will show up the latest record configured by users. This isn’t permitted for the empty info about the name in this mode.
Users could press the button to select the action in the view mode. If there is no any entry for the phone book, LCD will show the massage as following: No Records!! Dial Out on specific phone book entry, phone will be off-hook and dial out Press automatically. Modify Entry Here user can modify name and number of existed phone book data. Delete Entry Delete it? → OK Delete ok… → Detail Press to see detail name and number of this phone book entry. 3.2.
2 Used 88 Free 3.3 Networking Setting IP Mode Fixed DHCP PPPoE ¾ ¾ ¾ ¾ ¾ ¾ ¾ IP Address Net Mask Default GW DNS Setting Primary DNS Second DNS PPPoE ID Password Reconnect SNTP SNTP Server IP Time Zone Mode 3.3.1 IP Mode M4 Press the M1 or button to select the IP mode. 3.3.2 IP Address Enter IP address for Fixed IP mode. Under this mode can only input digits. IP Address: 10.1.1.
3.3.3 Net Mask All the operation is just like the IP Address configuring. It will be shown as following from the LCD: IP Mask: 255.0.0.0 3.3.4 Default GW All the operation is just like the IP Address configuring. It will be shown as following from the LCD: Default GW: 10.1.1.254 If users input the invalid value for the IP address, Net Mask or Default Gateway, LCD will show as following and return to the original setting of this configuration table: Invalid Input… 3.3.
3.3.6 PPPoE ID It could support the number or character typing in this mode. User Name: pppoe Password For the protecting policy, LCD will use the asterisk to replace the info showing. Password: ***** For the password modify displaying, the LCD will clean all the asterisks and showing the cursor as following: Password: ***** → → OK Password: → Modify? → OK Password: Reconnet Please press the button 3.3.7 M1 or to enable or disable this function. or to enable or disable this function.
Server IP To enter SNTP server IP address. It supports the number typing and Domain typing. Server IP: 168.95.195.12 Time Zone M4 Pressing the The M1 for decreasing and for increasing the zone value. button will save the changed. Zone: GMT +8:00 3.4 SIP setting All the menu operations are as same as the Networking setting. ¾ ¾ Proxy Setting Proxy IP Proxy Port OutPx IP OutPx Port User Setting ID Password Phone Num Local Port 3.4.1 Proxy Setting 3.4.1.1.
Proxy IP: 10.1.1.2 Proxy IP: proxy.com 3.4.1.2. Proxy Port Configuring the Proxy port in this table; it could only support the number typing. The max value is 65535. Proxy Port: 5060 3.4.1.3. OutPx IP This is the setting for the Outbound Proxy. It could support the number and character typing for the IP or domain. OutPx IP: 10.1.1.2 OutPx IP: outpx.com 3.4.1.4. OutPx Port This could only support the number typing only. The max value is 65535 OutPx Port: 5060 3.4.2 User Setting 3.4.2.1.
the cursor as following: Password: *** → → Password: → Modify? → Password: 3.4.2.3. Phone Num It could support the number typing only. PhoneNum: 100 3.4.2.4. Local Port It could support the number typing only and the max value is 65535. Local Port: 5060 3.
3.5.1 Alarm Setting There are some definitions for the Alarm clock, please check out the detail as following: If the alarm clock had been set and the time is up: z z z z z z The phone will be ring in the IDLE and MENU state only. If the phone isn’t in the IDLE or MENU state, the phone will ring while the state back to the IDLE or MENU. If the state don’t return to the IDLE or MENU for 30 minutes, the clock will cancel and don’t ring the phone. Ringing the phone every 20 seconds.
>01 06/01/01 10:20 >02 EveryDay 10:11 3.5.1.2. View All >01 06/01/01 10:20 >02 EveryDay 10:11 → → M4 01 06/01/01 10:20 >>02 EveryDay 10:11 → >Delete Entry → → → Deleting… → → Del the alarm? >01 06/01/01 10:20 >02 Empty 3.5.1.3. Del All Delete All? → OK Deleting… → → >View All >Delete All 3.5.2 Ring Setting 3.5.2.1. Ringer Volume There are four levels for the volume adjustment and the default is level 2. The symbol “>” will point out the current level for this phone set.
the ring volume. 3.5.2.2. Ringer Melody This phone set could support four types of ring melody. >Melody 1 >Melody 2 Saving … M4 → → → >Melody 1 >Melody 2 → OK → >Melody 1 >Melody 2 3.6 Mail Box ¾ ¾ ¾ ¾ Information MailBox No. MailBox Key Voice Mail Dial 3.6.1 Information If VNS LED lights up, you can view voice mail information from this item. Below: Information: 2 new, 1 old 3.6.2 MailBox No. 3.6.3 MailBox Key 3.6.4 Voice Mail Dial 3.
¾ ¾ Input Number Press the key… From PhoneBook Phone Book List Press the key… View Entry Function Key list F1: F2: F3: Users could configure the Function Keys for the speed dial or special IP-PBX function. For the function keys configuration, users could input the new number or select the entries from the Phone Book. 3.7.1 New Entry Set function key number and key button. User needs to input number first and then defines which key to match the number. 3.7.
4. Basic Function 4.1 Power on and initialization During the initialization procedure: 1 2 3 All the LED will light on till the initialization procedure was finished. All the LED will light on if there is error during the procedure. On LCD will show “Initializing”. 4.2 Making Calls There are some ways to make outgoing calls: 1 OFF HOOK Dialing 2 3 4 5 6 Redial (OFF-HOOK Dialing) Pre-Dial (ON-HOOK Dialing) Redial (ON-HOOK Dialing) Dial during connected Memory Dial 4.2.
During the Pre-dial function: 1 2 This is permitted to enter and dial out the digits “*” and “#”. If the dialed number could be matched with the list in the phone book, LCD will display the name of this record. 4.2.4 Redial (ON-HOOK Dialing) This will show all the records of the dialed list. After the records selected, pressing the 4.2.5 or button, the number will be dialed out. Dial during connected Pressing the or buttons could switch the Line 1 and Line 2.
4.4 Hold and Retrieve Calls There are two ways to hold the current calls and retrieve them back: 1 Press another idle line button or to hold the current line and switch to another IDLE line. 2 Press the Hold function keys to hold the current calls. During the hold status, the LED of Line will flash. 4.5 Transfer SP5101 could support the call transfer with two types, one is the Consultant, and another is the Blind transfer. The transfer could be initialized only for the state as above.
4.6 Conference User needs to specify conference function to be local conference or server-based conference. Conference Scenario: A. Local Conference: (1) A communicate with B→Conf or Line hear dial tone+ C’s number→C pick up → press Conf again to build conference (2) A communicate with B→Conf or Line hear dial tone+ C’s number→C refuse to join conference → press Line to hang up C and hear dial tone again → press original Line to retrieve call B.
5. Web Administration 5.1 Factory Default DESCRIPTION IP DEFAULT VALUE 10.1.1.3 Net Mask: Default Gateway: 255.255.0.0 192.168.1.254 SIP Proxy: SIP account 10.1.1.2 1001 SIP Port: Proxy Port: RTP Port: Digit Timeout: Call waiting: 5060 5060 16384 5s Off Call transfer On 5.2 Configuring the SIP Phone through Web Pages The HTTPD web management interface provides user an easier way to configure rather than command line method through TELNET.
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 / user Note: Login with “user” only has authority as below: 1. Modify network configuration 2. Modify Phone Book 3. Change login password of “user” 4. Reboot Password (The same with TELNET): Null (just press confirm, no need to key in password in default value) Note: User can set password later via web interface.
Step 3. Enter the web interface main screen After enter login name and password, user can see web interface main screen as below. Step 4. Start to configure After enter web management interface, user can see four main items. 1. Installation Wizard: User can follow steps in wizard to make first-time initial configuration. 2. Advanced Configuration: This menu includes other advanced configuration items. Please press triangle figure to list all items below Advanced Configuration. 3.
5.3 Installation Wizard Installation Wizard includes three steps: (A) Network Connection Mode: User has to select SIP Phone network mode as Static IP, DHCP or PPPoE.
(B) Network Configuration: After selecting network connection mode, user has to input related network parameters. (1) Static IP: User has to input IP, subnet mask, default gateway, and DNS server address. (2) PPPoE: User has to input PPPoE connection user name and password.
(C) Protocol Configuration: After setting network, user has to set SIP related parameters. - Primary Proxy Address and port: If user select Proxy mode in item A, please input Primary Proxy address and signaling port of Proxy. - Secondary Proxy Address and port: User can also input secondary Proxy server and port for backup. - Outbound Proxy Address and port: User can input outbound Proxy and port if necessary. - Phone Number: Registering Phone number of SIP Phone.
5.4 Advanced Configuration 5.4.1 Network Configuration (1) Network Configuration Network Connection Mode: User has select network configuration mode first, then configuration page will display related items. (a) Static IP: If user selects Static IP mode, following items will be displayed.
address here. - HTTP Port for WEB Management: Set port number for user to configure SIP Phone via WEB management. Default value is 80.
(c) PPPoE: If user selects PPPoE mode, following items will be displayed. - DNS Server Obtained Mode: When SIP Phone is in DHCP or PPPoE mode, user can determine DNS address is obtained from server or set manually. - Primary DNS Server: If user determines to set DNS address manually, please set Primary Domain Name Server IP address here. - Secondary DNS Server: If user determines to set DNS address manually, please set Secondary Domain Name Server IP address here.
(2) Behind NAT - Behind IP Sharing: Select if enable SIP Phone behind IP Sharing router function. - IP Sharing Public IP Address: Set Public IP Address of IP Sharing router for SIP Phone to work behind IP sharing. - STUN Server address: If user wants to use STUN function, user must enable Behind NAT Device function then inputting STUN Server address. - STUN Server port: If the STUN server port doesn’t any restriction, you don’t input any port data.
(3) SNTP - - SNTP Mode: 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. Time Zone-GMT: Set time zone for SNTP Server time. User can set different time zone according to the location of SIP Phone. For example, in Taiwan the time zone should be set as 8,which means GMT+8.
5.4.2 SIP Configuration (1) SIP Main Configuration - Primary Proxy Address and port: If user select Proxy mode in item A, please input Primary Proxy address and signaling port of Proxy. - Secondary Proxy Address and port: User can also input secondary Proxy server and port for backup. - Outbound Proxy Address and port: User can input outbound Proxy and port if necessary. - Phone Number: Registering Phone number of SIP Phone.
5.4.3 System Configuration (1) Feature Configuration: - Inter Digit Time: 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. - Keypad DTMF Type: set DTMF type. User can select DTMF type SIP Phone transmits. - End of Dial Key: select end of dialing key, e.g.
(2) Function Key Configuration: - Conference : Set the Conference mode. Default: Local Conference. But now it is not support Server Conference mode yet! - Group Pick up: Set Group Pick up code, you might contact with your IP-PBX system administrator. - Specific Pick up: Set Specific Pick up code, you might contact with your IP-PBX system administrator. - F1: User-defined function key. - F2: User-defined function key. - F3: User-defined function key. - Forward: “Local forward”. Set Forward type and number.
5.4.4 Number Configuration (1) Phone Book - Add Data: User can specify 50 sets of phone book via web interface. Please input index, Name, E.164 number, IP Address, port of the destination device, drop prefix, and insert number. 1. name: Specify name for one pbook data 2. e.164: set phone number of callee. - Delete Data: User can delete any configured phone book data by index.
(2) Hotline If user set SIP Phone as hotline mode, once SIP Phone is off-hook, it will automatically dials phone number (Proxy Mode) set in hotline table.
(3) Digit Manipulation - add: Add a rule to drop or insert prefix digits of incoming call. - prefix: Set which prefix number to implement digit manipulation rule. - drop: Enable or disable drop function. If this function is enabled, Phone will drop prefix number on incoming call. - insert: Set which digit to insert. - delete: Delete a digit manipulation rule by index.
5.4.5 Media Configuration (1) Codec - - Codec Priority: set codecs priority in order. Please notice that user can set from 1 to 5 codecs as their need. For example, user can only set first priority as G.723.1, and set the others as x, that means only G.723.1 is available. Packet Size: User can set different packet size for each codec.
(2) Voice - Volume: Adjust the volume of Ringer, Receive (Local side hearing), transmit (remote side hearing), DTMF. Echo Cancelor: Enable / Disable (it is suggested to always Enable this function).
(3) Tone Configuration - Ring Back Tone: Set ring back tone parameters. - Busy Tone: Set busy tone parameters. - Dial Tone: Set dial tone parameters. - 2nd Dial Tone: Set 2nd dial tone parameters. - Ring Tone: Set ring tone parameters.
(4) Payload Type - RFC2833 Payload Type: Change RFC2833 Payload type. This is for special request from the other site, if RFC2833 payload types of 2 sites are different, it may cause some problem of connection.
5.4.6 Device Management (1) Login Password - Change password configuration: First select login name as root or user, then enter current password, new password and confirm new password again to set new password.
(2) Software Upgrade - Download Mode: Select download method as TFTP or FTP - TFTP/FTP Server IP Address: Set TFTP server IP address - FTP Login: Set FTP login name and password - Target File name: Set file name prepared to upgrade - Target File Type: Select which sector of SIP Phone to upgrade Note: After upgrade Application, please remember to execute Flash Clean, which will clean all configurations become factory values except Network settings..
(3) Provision Server - Provision Server Address: set Provision server IP address. - Provision Server Login User Name: set Provision Server login user ID. - Provision Server Login User Password: set Provision Server login user password. - Provision Server Cycle Time: set Provision Server update cycle time.
(4) Flash Clean Press CLEAN will clean all configurations except Network Configuration of SIP Phone and reset to factory default value. Note: User must re-configure all commands all over again (except Network Configuration).
5.5 System Status 5.5.1 Network Status Display all current network status of SIP Phone.
5.5.2 Version Information: Display software version.
5.6 Reboot 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.
6. 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. If user forgets password, please contact with the distributor, we will generate a specific password according to your MAC address of SIP Phone.
Note: 1. After user enter SIP Phone configuration via telnet, please use login: ”root”, password: null, press enter to enter command lines. 2. If user forget login password, please contact with your distributor, we will generate one new password according to LAN Phone’s MAC address. Please login with “mac” and this new password. 3. User must input lower-case command, but contents of configurations such as SIP alias or user name etc, user can set as capital case. 4.
6.3 [debug] command This command is for engineers to debug system of SIP Phone SP5101. User can add debug flag via command debug –add “debug flags”, and then start debug function via command debug –open. When SIP Phone SP5101 is working on screen will display related debug messages. Most frequently used debug flag are “sip”, “fsm”, “msg”…etc. 6.4 [reboot] command After typing commit command, type reboot to restart the SIP Phone SP5101. 6.5 [pbook] command SIP Phone SP5101 can support 90 phone book data.
6.6 [commit] command Save any changes after configuring the SIP Phone SP5101. 6.7 [ping] command Command ping can test which the IP address is reachable or not. Usage: ping “IP address” The message will display packets transmitting condition or no answer from the IP address. 6.8 [time] command When SIP Phone SP5101 enable SNTP function and be able to connect with SNTP server, type time command will show the current time retrieved from SNTP server.
6.9 [ifaddr] command Configure and display the SIP Phone SP5101 IP information. 1. –print: print out all current configurations of ifaddr command. 2. -ip, -mask, -gate: Set SIP Phone SP5101 IP Address, subnet mask and default gateway respectively. 3. -ipmode: Set SIP Phone network mode to be Fixed IP, DHCP or PPPoE. When User set IP mode to be fixed IP, please set IP, subnet Mask, default gateway as mentioned in item 2.
11. –id: Input ID of Provision server. 12. –pwd: Input Password of Provision server. 13. –emstime: Set provision cycle time. 14. –stun: Input STUN server address. (ifaddr –stun 1 61.220.2.2) But you must take notice when you use this function, you must enable “ipsharing” function. 15. –stunport: Input port of STUN server.
4. –reboot: Select enable or disable this function. If user enables this function, after PPPoE disconnected, SIP Phone will automatically reboot to re-connect, and after reboot, if SIP Phone still can’t connect with server, SIP Phone will keep trying to connect. On the other hand, if user disables this function, SIP Phone won’t reboot and keep trying to connect. (pppoe –reboot 0/1) 5. –echo: to set PPPoE send echo request function or not.
6.12 [sysconf] command 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. -keypad: set DTMF type .User can select DTMF type SIP Phone receive and transmit.(sysconf –keypad 0/1 , 0 for in band ,1 for RFC2833.) 4. -2833type: change RFC2833 Payload type. 5. -eod: select end of dialing key, e.g.
6.13 [sip] command 1. –print: display all current configurations. 2. –px: set proxy server IP address or URL address (sip –px “IP address or URL of Proxy server”). 3. –px2: set alternative proxy server IP address or URL address. If phone failed to register first proxy, it will try to register this alternative proxy. 4. –pxport: set listening port of Proxy server. 5. –outpx: set IP address of outbound proxy server.
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. 12. –sexpire: set the session timer. 13. –minse: set the mini session timer. 6.14 [security] command 1. –print: display all current configurations. 2. -name: set user ID of SIP Phone SP5101 for registering.
6.15 [line] command 1. –print: display all current configurations. 2. –fwd: set forward function. There are 3 selections in Forward type, user must select under which condition to forward calls. 6.15.1 Busy Forward When SIP-Phone is in busy status, the incoming call will be forwarded to the assigned phone number. 6.15.2 No Response When SIP-Phone hasn’t been picked up for around 10 seconds, the incoming call will be forwarded to the assigned phone number. 6.15.
6.16 [voice] command The voice command is associated with the voice codec setting information. 1. -print: display voice codec information and configuration. 2. -send: three voice packet size can be configured as 20 ms, 40 ms or 60 ms.(only 30 and 60 ms for G..723.1) 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 g729 711a g711u means SIP Phone can support only one codec or four codecs. 4.
tone, rbt: ring back tone, bt: busy tone, dt: dial tone). This parameter should be accompanied with tone type. 6.18 2. -ring: To set RING tone value. The played tone type, when Phone ha incoming call. 3. -rbt: To set Ring Back Tone value. The played tone type, when Phone dials out a call. 4. -bt: To set Busy Tone value. The played tone type, when Phone dials to a destination that is busy. 5. -dt: To set Dial Tone value. The played tone type, when hook off the Phone.
1. High priority with DS-field. Expected Forwarding (EF) 101110 Assured Forwarding 001010 ====> 10 (Decimal System) 010010 ====> 18 (Decimal System) 011010 ====> 26 (Decimal System) 100010 ====> 34 (Decimal System) (AF) ====> 46 (Decimal System) 2.
z drop: Enable or disable drop function. If this function is enabled, Gateway will drop prefix number on incoming call. z insert: Set which digit to insert on incoming call. Example: prefix -add prefix 100 drop 1 insert 2000 2. -modify: Modify a rule to drop or insert prefix digits of incoming call. Example: prefix –modify 100 drop 0 insert 200 3. -delete: Delete a rule to drop or insert prefix digits of incoming call. Example: prefix –delete modify 100 drop 0 insert 200 6.20 [rom] command 1.
the TFTP server is 192.168.1.1, User has to input command as below: rom –app –s 192.168.1.1 –f lpsip.100 Command rom –print can show current version installed in SIP Phone SP5101. 6.21 [auth] command For security concern, the “root” user can customize some configurable items for “administrator” user. 1. -“item name”: Assign the configurable item for “administrator” user. Example: auth -ifaddr 1 auth -sip 1 auth -voice 1 Now the Administrator can use these command which Root user assign to them. 2.
6.22 [passwd] command For security protection, user has to input the password before entering application user/config mode. Two configurations of login name/password are supported by the system. 1. –set: set password of “root” users or “administrator” users. (passwd –set root/administrator “password”) 2. –clean: clean up password restored before, and user can login :”root/administrator”, password: ”press enter”.
