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SIP-Phone User Manual

1. 4 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 21 SIP SIP Phone Administration Guide Chapter 3 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 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 22 SIP SIP Phone Administration Guide 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 23 SIP SIP Phone Administration Guide Step 2 Input the login name and password m 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 m Password The same with TELNET Null just press confirm no nee
2. Network Interface SIP Information System Configuration PPPoE Configure Voice Setting Phone Book Prefix Configuration DDNS Configuration DSCP Configuration Password ROM Configuration Flash Clean Reboot System 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 41 SIP SIP Phone Administration Guide 13 Reboot System SIP Phone Configuration It will take 40 seconds to reboot g Menu remember to COMMIT data before reboot REBOOT Network Interface SIP Information System Configuration PPPoE Configure Voice Setting Phone Book Prefix Configuration DDNS Configuration DSCP Configuration Password ROM Configuration Flash Clean 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 42 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 pass
3. 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 3 Specification of connector 1 Ethernet Port Ethernet port is for connecting SIP Phone to network transmit rate supports 10 100 Base T Ethernet connector LAN 2 RJ 11connector SIP SIP Phone Administration Guide RJ 11 connector is for connecting SIP Phone with PSTN NC N TP NIC z Z NA SIP SIP Phone Administration Guide 0000 or Hook Under neath BRGY RJ11connector 10 2 Software Features and Specification Application ISP ITSP Internet Telephony Service Provider IP PBX with office telephony services Multi nation enterprise communication SOHO Telephony Calling Features Call Hold Call Transfer Call Forward 10 sets last number Redial 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 1p1g VLAN Audio Features G 711 a u Law G 723 1 G 729 G 729a VAD CNG G 165 G 168 compliant echo cancellation Programmable Dynamic Jitter Buffer Bad Fram
4. 48 11 flash command 12 sysconf command 13 sip command 14 security command 15 voice command 16 tos command 17 ddns command 18 vlan command 19 bureau command 20 prefix command 21 rom command 22 passwd command Chapter 5 Upgrading Software on the SIP Phone 1 From LCD Phone Menu 2 Web Management 3 Remote Control Telnet SIP SIP Phone Administration Guide 48 49 50 51 52 53 55 56 57 58 58 59 61 61 62 63 Revision History Version Date Author Software Version Note LP_SIP_100 2004 03 Sabrina For test version only First released for software test version 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 2 3 4 5 Forward function Call list function LCD menu firmware upgrading section Add DDNS function Add VLAN function LP_SIP_102 2005 3 10 Sabrina lp201sip 102 Modify 1 2 3 4 PPPoE Echo EMS SIP Info DTMF 2 level access LP_SIP_103 2005 12 28 Jason lp201sip 104 Modify if LCD interface can upgrade new firmware successfully When LP201 is under P2P mode it can show SIP listen port Add secondary proxy Delete Session Expire function SIP SIP Phone Administration
5. 723 Silence enable disable PPPoE Configure Suppression Voice Setting Volume voice 32 ring 32 input 22 DTMF a1 Phone Book Echo C di Prefix Configuration Cancelor enable ieabia DDNS Configuration Jitter Buffer Min Delay oo Max Delay 150 DSCP Configuration Optimized Factor 9 Password Gitter ROM Configuration OK Flash Clean Reboot System 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 Frame Size User can set different packet size for each codec G 723 Silence Suppression Enable Disable sound compression and comfort noise generation It is only for codec G723 1 Volume Adjust the volume in Voice sending out Input receiving DTMF DTMF sending out Echo Cancelor Enable Disable suggested always Enable this function Jitter 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 33 SIP SIP Phone Administration Guide 6 Phone Book Please refer to chapter 4 5 pbook command SIP Phone Configuration Menu Network Interface SIP Information System Configuration PPPo
6. From 0 to 7 7 is the highest priority 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 SIP Phone will drop priority tag on packets sending out from PC port 56 SIP SIP Phone Administration Guide usr config vlan LAN information and configuration Usage lan dev enable vid number pcvid number L priority number pcpriority number pcdroptag enable print ud VLAN information and configuration dew Enable disable VLAN configuration O Disable 1 Enabl e vid VID number 1 4095 default 1 pcvid PC port VID number 1 4095 default l priority Priority number 0 3 Low priority 4 High priority pcpriority PC port priority number 0 3 Low priority 4 High prio pcdroptag Drop PC port tag O Disable 1 Enable Example vlan dev 1 vlan vid 2 vlan pcvid 2 vlan priority 4 vlan pcpriority O vlan pcdroptag 1 usr config 19 bureau command 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 bureau hotline IP address phone number of remote site Note To set SIP Phone as hotline mode user has to set SIP Phone as Peer
7. Guide Chapter 1 Overview of the SIP Phone SIP Phone is a full featured IP based telephone set via Ethernet base communication Over the office LAN connection it provides IP PBX solution such as station to station call IP call and local PSTN PBX Extension call via PSTN Gateway SIP Phone provides two 10 100BaseT switch hub RJ 45 ports allow connecting to office LAN and PC simultaneously It is compatible with ADSL or Cable Modem provided by ITSP ISP or Carrier Company to provide VoIP services to residential and SOHO application SIP Phone is also an integrated Analog Phone provides IP call or PSTN call selection When external power is down it can be a Plain Old Telephone set POTs It provides internal high guality 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 SIP SIP Phone Administration Guide 1 Hardware Overview 1 Front View and Keypad function o o ws oe TTT oe Sas THANNFEB APAL BPR RE LCD 2 lines 24 character Dot Matrix display C Jump out current LCD menu Move to previous selection or clear previous data gt Move to right or next selection OK Press OK to confirm the modification Direct Line DL Button 1 10 User press DL button after off hook to do speed dial according to phon
8. Phone Book Search Enable Disable DSCP Configuration Prefix String null Password Line Number 1001 ROM Configuration Line Account 1001 Flash Clean Line Password Reboot System SIP port 5060 RTP Port 16384 Expire 60 OK Run Mode Select SIP Phone to work under Peer to Peer mode or Proxy mode Primary Proxy IP Address Set Primary Proxy IP Address or URL address Domain Name Server must be configured Please refer to Network Interface Proxy port Set Proxy port for SIP Phone to send message default value is 5060 if there is no special request of Proxy server please don t change this value Secondary Proxy IP Address Set secondary proxy IP Address or URL address Domain Name Server must be configured Please refer to Network Interface When LP201 fail to register to primary proxy It will try to register to secondary Proxy when it fail again it will retry to register to primary Proxy Secondary Proxy port Set the secondary proxy port for every SIP massage default 29 SIP SIP Phone Administration Guide value is 5060 Outbound Proxy Set IP Address or URL address Domain Name Server must be configured Please refer to Network Configure of outbound Proxy server Outbount Proxy port Set outbound proxy port for LP201to send message default value is 5060 if there is no special request of proxy server please don t change this value Phone Book Search enable disable
9. TFTP server IP address Target File name Set file name prepared to upgrade Method 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 2 After upgrade Application please remember to execute Flash Clean 62 SIP SIP Phone Administration Guide which will clean all configurations become factory values except IP address 3 Remote Control Telnet Please refer to chapter 4 18 rom command o print show all current configurations and version information app boot dsptest dspcore dspapp upgrade main boot code main application code DSP testing code DSP kernel code DSP application code Ring Back Tone PCM file and Hold Tone Note After upgrade Application please remember to execute flash clean command which will clean all configurations become factory values except IP address boot2m to upgrade 2mb rom file which includes all firmware file mentioned in item 2 S it is necessary to prepare TFTP FTP server IP address for upgrading firmware rom file f the file name prepared for upgrading is necessary as well method specify download method to be TFTP or FTP O for TFTP 1 for FTP ftp specify user name and password for FTP down
10. bytes 64 bytes from 192 168 2 107 icmp_seq 6 time 5 ms 64 bytes from 192 168 2 167 icmp_seg 1 time 8 ms 64 bytes from 192 168 2 167 icmp_seg 2 time 8 ms 64 bytes from 192 168 2 107 icmp_seg 3 time 6 ms 192 168 2 187 PING Statistics 4 packets transmitted 4 packets received 6 packet loss round trip ms min aug max 6 1 5 usr config ping 192 168 99 99 PING 192 168 99 99 56 data bytes no answer from 192 168 99 99 usr config J 8 time command When SIP Phone enables SNTP function and be able to connect with SNTP server type time command will show the current time retrieved from SNTP server usr config time urrent time is TUE FEB 03 14 50 08 2004 usr config 9 ifaddr command Configure and display the SIP Phone IP information 1 print print out all current configurations of ifaddr command 2 ip mask gate Set SIP Phone IP Address subnet mask and default gateway respectively 5 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 If User set IP mode to be DHCP SIP Phone will search for DHCP server to capture IP address after reboot If user set IP mode to be PPPoE please remember to set related parameters under pppoe command 6 sntp When SNTP server is available enable SIP Phone SNTP function and assign SNTP server IP address so that SIP Phone can capture
11. current time from SNTP server ifaddr sntp 1 xxx xxx xxx xxx 7 dns User can set Domain Name Server IP address Once SIP Phone can connect with DNS server user can specify URL address instead of IP address for Proxy Server and phone book IP address etc 46 SIP SIP Phone Administration Guide 8 timezone User can set different time zone according to the location SIP Phone is For example in Taiwan the time zone should be set as 8 which means GMT 8 GMT 8 ifaddr timezone 8 9 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 O for disable and 1 for enable Note vendor 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 10 upnp enable disable UPnP function If the IP sharing or NAT device supports UPnP user can enable UPnP function so that SIP Phone will automatically connect with NAT device without configuration in SIP Phone and NAT device 11 server set EMS server IP address 12 id set EMS login user ID 13 pwd set EMS login user password 14 emstime set EMS server recycle time usr config ifaddr LAN information and configuration Usage ifaddr print dhcp used sntp mode server ifaddr ip ipaddress mask subnetmask
12. gate defaultgateway ifaddr dns index dns server address ifaddr autodns used print Display LAN information and configuration ip Specify ip address mask Set Internet subnet mask gate Specify default gateway ip address ipmode Set get IP mode O Fixed IP 1 DHCP 2 PPPOE sntp Set SNTP server mode and specify IP address autodns Specify the way to obtain DNS Server O Manual l Auto dns specify IP address of DNS Server timezone Set local timezone h h h ipsharing Specify usage of an IP sharing device and specify IP address upnp Specify the server Specify EMS id Specify EMS pwd Specify EMS emstime Specify EMS Note upnp mode of ipsharing O Off 1 On Server IP address Server ID Server password cycle time Range of ip address setting 0 0 0 0 255 255 255 255 SNTP mode O no update Example l specify server IP 2 broadcast mode ifaddr ip 210 59 163 202 mask 255 255 255 0 gate 210 59 163 254 ifaddr ipmode 1 ifaddr ap 1 210 59 163 254 a ifaddr ips ifaddr upnp 1 ifaddr autodns 1 ifaddr dns 1 168 95 1 1 ring 1 210 59 163 254 zi SIP SIP Phone Administration Guide 47 10 pppoe command print display all current configurations and information id to set PPPoE authentication user name pwd to set PPPoE authentication password reboot Choose enable or disable this function If user enables this function after PPPoE disconnected SIP Ph
13. hangs up B and C can be connected Note 1 Acannot press phone number of C before hearing dial tone 2 If Apress TRANSFER and number press TRANSFER again can cancel transferring and retrieve call with B 3 If Apress TRANSFER and number of C C is ringing press TRANSFER again C will return to standby mode and A can retrieve call with B 4 Before C picks up A cannot hangs up the phone 2 Change characters to be capital or lowercase when pressing TRANSFER before 7 SIP SIP Phone Administration Guide press letters can switch input type of letters REDIAL Redial the last outgoing call And Adjust the voice volume heard of communication SPEAKER Hand free mode User can talk without picking up handset Note 1 All function keys mentioned above except dialing keypad are effective only in IP Phone mode 2 When SIP Phone fail to register to Proxy server under Proxy mode when user wants to dial out SIP Phone will play busy tone and on LCD will display Register Failed SIP SIP Phone Administration Guide 2 Back View DC 9V DC 9V power input outlet LAN RJ 45 connector connected directly to the Hub through the straight CAT 5 cable PC RJ 45 connector connected directly to the PC through the straight CAT 5 cable Line RJ 11 connector connected directly to the PSTN analog line Note There are two LED indicated lights LINK ACT and 10 100 for LAN port and PC port
14. port of Secondary proxy outpx OlutBound Proxy server address IPvd address or dns name outpxport OutBound Proxy server port the port of OutBound proxy prefix Specify prefix string use it when UserID contains alphabets if UserID uses numerals specify as null line TEL Phone number pbsearch Search phone book Oioff 1 on expire The relative time after which the message expires O 2731 15 port SIP local UDP port number 5060 5070 Default 5060 rtp RTP receive port number 2326 65534 gt Default 16384 Example sip mode 1 sip px 210 59 163 171 line 70 4 usr config 14 security command 1 print display all current configurations 2 name set user ID of SIP Phone for registering User can set user name and password for registering If password is no need please set user name the same as line number or SIP Phone won t register successfully 3 pwd set account password for registering 51 SIP SIP Phone Administration Guide usr config security Secuirty information and configuration Usage security name username password password security print print Display system account information and configuration name Specify user name pwd specify password Example security name 1001 pwd 1001 usr config 15 voice command The voice command is associated with the voice codec setting information 1 print display voice codec information and configuration 2 send
15. shown as follows 1 High priority with DS field Expected Forwarding EF 101110 gt 46 Decimal System Assured Forwarding AF 001010 gt 10 Decimal System 010010 gt 18 Decimal System 011010 gt 26 Decimal System 100010 gt 34 Decimal System 2 Low Priority with DS field Assured Forwarding AF 001100 gt 12 Decimal System 010100 gt 20 Decimal System 011100 gt 28 Decimal System 100100 gt 36 Decimal System 001110 gt 14 Decimal System 010110 gt 22 Decimal System 53 SIP SIP Phone Administration Guide 011110 gt 30 Decimal System 100110 gt 38 Decimal System 000000 gt 0 Decimal System DROP Precedence Class 1 Class 2 Class 3 Class 4 Low Drop Precedence AF 11 AF21 AF31 AF 41 001010 010010 011010 100010 Medium Drop Precedence AF12 AF22 AF32 AF42 001100 010100 011100 100100 High Drop Precedence AF13 AF23 AF33 AF43 001110 010110 011110 100110 1 print display all current configurations 2 rtptype set DSCP value of signaling packets from 0 to 63 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 usr config tas IP Packet ToS type of service Differentiated Service configurat
16. 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 g723 711a g711u g729 g729a means SIP Phone can support only one codec up to five codecs 4 volume There are three types can be adjustable voice volume ring volume input gain and DTMF volume Voice volume means the volume user can hear input gain means the volume the other side can hear from SIP Phone DTMF means DTMF transmitting volume voice volume voice value of volume voice volume ring value of volume voice volume input value of volume voice volume dtmf value of volume Note If value of volume set as 0 means 32db 1 means 31db etc 5 nscng enable or disable sound compression and comfort noise generation It is only for codec G 723 1 0 for off 1 for on 6 echo enable or disable echo cancellation function 7 mindelay set minimum delay of jitter buffer 0 150 8 maxdealy set maximum delay of jitter buffer 0 150 Note It is for advanced administrator use only Please ask your distributor before changing any settings of this command 52 SIP SIP Phone Administration Guide oicea print a oice priority 6723 6729 G7llu G711A G729A beo print Display voice codec information
17. to Peer mode and sysconf service 2 command usr config bureau Bureau line setting information and configuration Usage bureau hotline Port DestIP TELnum bureau print print Display Bureau line information and configuration hotline Set Hot line information Port range 1 6 Note Hotline feature should be used together with sysconf service 2 HotLine service l sip mode 1 peer to peer mode Example ureau hotline 1 192 168 4 69 628 usr config 57 SIP SIP Phone Administration Guide 20 prefix command print show all current configurations add add a rule to drop or insert prefix digits of out going call prefix add prefix prefix number drop 0 1 insert insert number delete delete arule to drop or insert prefix digits of out going call prefix delete prefix prefix number modify modify a rule to drop or insert prefix digits of out going call prefix modify prefix prefix number drop 0 1 insert insert number prefix set which prefix number to implement prefix rule drop enable or disable drop function If this function is enable Lan Phone will drop prefix number on out going call insert set which digit to insert on out going call Print Screen usr config prefix Prefix Usage prefix prefix prefix prefix drop insert information and configuration add prefix numberlLdrop numberlLinsert digits delete index modify i
18. Delete Data If users want to delete the record form the table enter the index number first and press this button The record will be delete 35 SIP SIP Phone Administration Guide 8 DDNS Configuration Please refer to chapter 4 17 ddns command Configuration a F Status Enable Disale Menu E Network Interface Server wwwdyndnsorg v SIP Information Host Name j System Configuration ID PPPoE Configure Password Voice Setting Check IP On off Phone Book IP Check Server 1 checkip dyndns org Prefix Configuration DDNS Configuration IP Check Server 2 checkip dyndns org DSCP Configuration Check every jo C mintues hours Off Password DONE ROM Configuration Flash Clean Reboot System Status to enable disable DDNS function Server to choose one DDNS server on which user has already registered Now only one DDNS server is available www dyndns org Host Name to set the registered Domain Name of SIP Phone ID to set login ID of registered account to log in DDNS server Password set password of registered account to log in DDNS server Check IP to enable disable check IP function If SIP Phone is behind IP sharing when this function is enabled SIP Phone will check it s public IP address by asking IP address check server and send to DDNS server to update DDNS data If this function is disabled when SIP Phone is behind IP s
19. E Configure Voice Setting Prefix Configuration DDNS Configuration DSCP Configuration Password ROM Configuration Flash Clean IP Address Index Name E164 No Reboot System Add Data Delete Data Add Data 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 34 SIP SIP Phone Administration Guide 7 Prefix Configuration Please refer to chapter 4 20 prefix command Configuration Henu a Prati Drop men O Network Interface SIP Information System Configuration PPPoE Configure Voice Setting Phone Book Prefix Confiquration DDNS Configuration DSCP Configuration Password ROM Configuration Flash Clean E Prefix L O Index Prefix prop Enable Disable Insert Add Data Delete Data Index The list number of the Phone Book Prefix The prefix number of the whole numbers that could be into this lan phone Drop The drop function Enable this function by the Enable button Disable this function by the Disable button Insert The insert function Users could enter the digits that you want to insert in this number Add Data Press this button if users fill the entire information table above
20. M files updating commands Usage rom print app boot dsptest dspcore dspapp s TFTP FTP server ip f filename rom print print show versions of rom files optional app update main application code optional boot update main boot code optional boot2m update 2m code optional dsptest update DSP testing code optional dspcore update DSP kernel codeCoptional dspapp update DSP application code optional 5 IP address of TFTP FTP server mandatory f file name mandatory method download via TFTP FTP TFTP mode 0 FTP mode l ftp specify username and password for FTP Note This command can run select one option in app boot dsptest dspcore and dspapp Example rom method 1 rom ftp vwusr vwusr rom app s 192 168 4 101 f app bin usr config El xi Command rom print can show current version installed in SIP Phone usr config rom print Download Method TFTP Boot Rom 1p201_boot 101 pplication Rom 1p201_sip0202 BIN DSP Ap 48302ce3 140 DSP Kerne 48302ck 140 DSP Test Code 483cbit bin El usr config z 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 user users passwd set root user 59 SIP SIP Phone Administration Guide password 2 clean cle
21. P Phone will dial out IP address 1 print display phone book data User can print all data in phone book by command pbook print Furthermore user can also print only a section of data by indicate parameter start index and end index pbook print start index end index If parameter end index is omitted only record start index will be displayed pbook print start prefix 2 add add a new record in phone book table by giving name IP address and e 164 number of callee endpoint 44 SIP SIP Phone Administration Guide pbook add name X ip xxx xxx xXxx xxx e164 X 3 search search any record in the phone book according to IP address name or e164 number 4 delete delete a record of certain listed index in phone book table pbook delete index number 5 insert insert a record in specified index of phone book 6 modify modify record of a certain index in phone book Please notice that the name IP address and e164 number must be modified together user cannot just modify one parameter only pbook modify index name X ip xxx xxx xxx xxx e164 X usr config pbook Phonebook information and configuration Usage pbook C print Cstart_record Lend_recordll pbook C add ip ipaddress name Alias e164 phonenumberll pbook C search ip ipaddress name Alias Lei6d phonenumber J pbook C insert index ip ipaddress na
22. 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 4 If in communication with PSTN side user must hang up PSTN side before pick up IP side HOLD To hold a call after press HOLD button both sides will hear hold tone SPEED 1 Press SPEED and number Phone book index after off hook can do speed dial according to phone book data please refer to 3 LCD configuration 6 Phone Book or Advanced Configurations via 4 Telnet 10 pbook command 2 Switch input mode between character mode or digit mode e g when user wants to input phone number can press SPEED to switch input mode as digit mode when user wants to input name can also press SPEED to switch input mode as character mode FORWARD Forward an incoming call to another IP device Please refer to LCD configuration Forward Type MESSAGE and its indicated LED light When having missed incoming calls the MESSAGE LED will be flashing User can check the information of missed calls by pressing the MESSAGE button TRANSFER 1 Transfer a call to the third site When A and B are in communication A wants to transfer this call to C Acan press TRANSFER button now B will hear hold tone and A will hear dial tone then Acan press phone number of C after C picks up A can talk with C after A
23. SIP Phone User Manual Welltech Computer Co Ltd Chapter 1 Overview of the SIP Phone 1 Hardware Overview 1 Front View and Keypad function 2 Back View 3 Specification of connector O ON on A 2 Software Features and Specification 3 Physical specification amp Environment Chapter 2 Configuring the SIP Phone through LCD Phone menus 1 Initialize SIP Phone 2 LCD Menu Configuration Chapter 3 Configuring the SIP Phone through Web Pages Step 1 Browse the IP Address predefined via Keypad Step 2 Input the login name and password Step 3 Enter the web interface main screen Step 4 Start to configure 1 Network Interface 2 SIP Information 3 System Configuration 4 PPPoE Configure 6 Phone Book 7 Prefix Configuration 8 DDNS Configuration 9 DSCP Configuration 10 Password 11 Rom Configuration 12 Flash Clean 13 Reboot System Chapter 4 Configuring the SIP Phone through Telnet command lines 1 help command quit command debug command reboot command pbook command commit command ping command time command Oo AN A M FW WN ifaddr command 10 pppoe command SIP SIP Phone Administration Guide 13 14 14 16 22 23 24 25 25 26 29 31 32 34 35 36 37 39 40 41 42 43 43 44 44 44 44 45 46 46 46
24. TFTP Sever User has to offer one TFTP FTP server IP Address and set this IP Address via keypad The IP Address is necessary for upgrading SIP Phone new application rom file 3 FTP Account User has to input user name for FTP server login It is necessary for upgrading SIP Phone new application rom file via FTP method 4 FTP Password User has to input password for FTP server login lt is necessary for upgrading SIP Phone new application rom file via FTP method 5 File Name User has to press the file name of new application rom file prepared for upgrading 6 Application Version Show versions of application software Note Download via LCD command can only upgrade new application rom file 7 Start to Upgrade 61 SIP SIP Phone Administration Guide 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 Configuration pe FTP TFTP server IP Address Network Interface Target File name SIP Information SIP Information Method IFIP d System Configuration FTP Login name passwd Target File Type Application Image X PPPoE Configure Voice Setting Phone Book OK Prefix Configuration DDNS Configuration DSCP Configuration Password ROM Configuration Flash Clean Reboot System FTP TFTP Server IP Address Set
25. an up password restored before and user can login root user password press enter User who requests authorization to execute all configuration commands needs to login with root If a user login with user only commands below are functional ifaddr pbook password set user password commit reboot oh ge oe O usr config passwd Password setting information and configuration Usage passwd set Loginname Password passwd clean Note 1 Loginname can be only root or administrator 2 passwd clean will clear all passwd stored in flash please use it with care Example passwd set root Your_Passwd_setting ie usr config 60 SIP SIP Phone Administration Guide Chapter 5 Upgrading Software on the SIP Phone SIP Phone supports three methods to upgrade the new version 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 1 LCD Panel Control 2 Remote Control Telnet 3 Web Management 1 From LCD Phone Menu Please choose the 6 2 selection Firmware Upgrade Press OK to enter into the sub selection as below 1 Firmware Update 1 Download method There are two methods to download new version file please move the gt symbol by press or on the keypad to select TFTP or FTP method then press OK to confirm it 2 FTP
26. and configuration send Specify sending packet size G 723 30 60 90 ms G 729 20 40 60 ms G 7110 20 40 60 ms G 7llA 20 40 60 ms G 729A 20 40 60 ms priority Priority preference of installed codecs G 723 G 729 G 7llu G 711A G 7294 volume Specify the re gyng levels voice volume 0 45 efault 32 ring volume oar default 32 input gain 26 45 default 329 dtmf volume 0 31 default 31 nscng No sound compression and CNG G 723 1 only On l off 0 echo Setting of echo canceller On l Off 0 per port basis mindelay Setting of jitter buffer min delay 0 150 default 90 mer Setting of jitter buffer max delay 0 150 default 150 Example voice send g723 60 g729 60 g7llu 60 g7lla 60 g729a 60 voice volume voice 20 input 32 dtmf 27 voice echo 1 usr config voice volume ring 20 usr config commit his may take a few seconds please wait ommit to flash memory ok usr fconfig lal 16 tos command TOS DiffServ DS priority function can discriminate the Differentiated Service Code Point DSCP of the DS field in the IP packet header and map each Code Point to a corresponding egress traffic priority As per the definition in RFC2474 the DS field is 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
27. ask Gateway display current IP information 2 Network Mode Set network mode of SIP Phone to be Static Fixed IP DHCP or PPPoE 3 IP address Set IP address of SIP Phone 4 Subnet Mask Set subnet mask address of SIP Phone 5 Default Gateway Set default gateway address of SIP Phone 6 Domain Name Server Set IP address of Domain Name Server Once SIP Phone can connect to DNS server user can set URL address for Proxy server or Phone book instead of IP address 7 PPPoE Configuration 1 User Name Set PPPoE connection authentication user name 2 Password Set PPPoE connection authentication password 3 Auto Re connect Choose ON or OFF to 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 8 SNTP Configuration 1 SNTP Mode User can set SNTP function to be on or off which means SIP Phone will 18 SIP SIP Phone Administration Guide capture current time from SNTP server or not 2 SNTP Server User can specify a SNTP server for SIP Phone to capture current time 3 Time Zone User can set time zone via pressing or gt according to the location SIP Phone is For example in Taiwan the time zone should be set as GMT 8 00 9 B
28. d to key in password in default value Z http 192 168 2 103 Microsoft Internet Exp x KE me HW BAE IAM RAY El fs A de NS 2 x SEWA CREA Se BREE Fib 192 168 2 103 8 IP PHONE Web Configuration FRKA root EE V medr En eie S wa Note User can set password later in 8 PASSWORD via web interface 24 SIP SIP Phone Administration Guide Step 3 Enter the web interface main screen After enter login name and password user can see web interface main screen as below SIP Phone Configuration Menu Network Interface SIP Information System Configuration PPPoE Configure Voice Setting Phone Book Prefix Configuration DDNS Configuration DSCP Configuration Password ROM Configuration Flash Clean Reboot System Step 4 Start to configure Welcome to SIP Phone Web Configuration Server Please click the stems sn Jeft frame Most important items are Network Interface SIP Information and 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 25 1 Network Interface Please refer to chapter 4 9 ifaddr command srP Phone HDU y Configuration Menu IP Address 10 1 A 8 Networ
29. dress of Proxy This function is for special proxy server 10 line identify one number for the SIP Phone to register to the Proxy SIP line line number Note In proxy mode please remember to set user account information under security command 50 SIP SIP Phone Administration Guide Bl 11 pbsearch enable disable phone book search function under Proxy Mode 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 12 expire set expire time of registration SIP Phone will Keep re registering to proxy server before expire timed out 13 port set listening UDP port or SIP Phone 14 rtp set RTP port number SIP Phone will use this port to send and receive voice usr config sip SIP stack information and configuration Usage sip mode pxmodel sip px address L prefix prefixstringl L pxport ProxuPortlL outpx addresslL line number C expire t1 C port udpPortl L rtp rtpPortl sip print print Display SIP stack information and configuration mode Configure as Proxy mode or Peer to Peer mode px Proxy server address Proxy IPvd address or Proxy dns name pxport proxy server port the port of proxy px2 Secondary proxy server address Proxu IPvd address or dns name px2port Secondary proxy server port the
30. e Interpolation Gain Attenuation Settings SIP SIP Phone Administration Guide Provisioning and Configuration SIP RFC3261 compliance LCD configuration password protection Provide Proxy Mode or Peer to Peer Mode Non Proxy Server needed selection Ring tone Speaker and Handset volume adjustable Dial path selection PSTN or IP mode Support DNS server inguiry 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 3 Physical specification amp Environment Dimension 215mm W x 71mm H x 198mm D Weight 834 grams unit Power Supply DC 9V output adaptor AC 120Vac input Operation Environment Humidity 10 to 90 Non condensing Operational Temperature 0 to 40 C Storage Environment Humidity 10 to 90 Non condensing Storage Temperature 10 to 50 C Analog Specification 2wires RJ 11 1 Voltage 48V AC Impedance 600 O 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 1 Initialize SIP Phone 1 When power on t
31. e 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 gt gt w SIP SIP Phone Administration Guide Number 1 10 and ff The function is the same as the general phone set Corresponding list of keypad and symbol 1 Character mode 1 i be 2 A B C 2 3 DS ES caren 4 G H P 4 5 as a EES 6 M N O 6 7 Po QUR S a ke 8 T4 UND a 9 W X MON O eee ee a Se KB 0 Space O 2 Digit mode 1 1 oe g 4 5 6 7 g Q JOJOD INJO JOB P O _ O 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 in IP Phone mode PROXY characters will be displayed on LCD left bottom side Note SIP SIP Phone Administration Guide 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
32. ehind IP Sharing 1 If SIP Phone is behind IP sharing or NAT device on IP sharing must enable DMZ function or set Virtual Server to open ports UDP port 5060 and 16384 2 User must enter public IP address of IP sharing 6 Advanced Settings protected by password Please Enter Password User must key in password to enter this menu selections under this command are all important ones which can only be configured by advanced users Note 1 Default Password is empty user can enter this sub menu directly once password has been set in 3 LCD Menu password user must input password before enter this sub menu 2 If user forget password please contact with the distributor we will generate a specific password according to your MAC address of SIP Phone 3 User can also try to configure SIP Phone via Telnet or Web browser with default IP address 10 1 1 3 This only works when default IP address hasn t been changed 1 SIP Settings 1 Connect Mode Select SIP connection mode to be peer to peer mode or Proxy mode 2 Proxy A Proxy Set Proxy IP address or Domain Name B Outbound proxy SIP SIP Phone Administration Guide Set Outbound Proxy IP address or Domain Name C Proxy port Set Proxy port for SIP Phone to send messages D Expire in seconds 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 Se
33. haring it will send it s private IP address to DDNS server IP Check Server 1 to set 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 SIP Phone will re update its IP address in this time 36 SIP SIP Phone Administration Guide 9 DSCP Configuration Please refer to chapter 4 16 tos command Configuration 3 ignal Packet Menu z Delay Priority Wass z Drop Precedence L v C Network Interface Mam Forwarding AF SIP Information C Expedited Forwarding EF PHB System Configuration 2 amp Default PPPoE Configure Voice Setting C User Assign Special DSCP Code Phone Book RTP Packet H 7 H Prefix Contiguration wan sad 4 Prefix Configuration ase Forwarding AF Delay Priority Gesi y prof Bracedanced ow z DDNS Configuration C Expedited Forwarding EF PHB DSCP Configuration Password DRO ROM Configuration C User Assign Special DSCP Code Flash Clean EN OK Reboot System Set Signal or RTP Packet DSCP value Assured Forwarding AF PHB Select Delay priority and Drop Precedence Expedited Forwarding EF PHB Select TOS value as EF Default Select TOS value as 0 User Assign Special DSCP Code User can set other unspecified value here TOS DiffServ DS priority function can discriminate t
34. he Differentiated Service Code Point DSCP of the DS field in the IP packet header and map each Code Point to a corresponding egress traffic priority As per the definition in RFC2474 the DS field is 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 37 SIP SIP Phone Administration Guide DROP Precedence Class 1 Class 2 Class 3 Class 4 Low Drop Precedence AF 11 AF21 AF31 AF 41 001010 010010 011010 100010 Medium Drop Precedence AF12 AF22 AF32 AF 42 001100 010100 011100 100100 High Drop Precedence AF13 AF23 AF33 AF43 001110 010110 011110 100110 Please refer to RFC standard documents for more information about what is DSCP SIP SIP Phone Administration Guide 38 10 Password Please refer to chapter 4 22 password command Configuration T Menu Current Password Network Interface root y New Password SIP Information Confirm New Password System Configuration CHANGE ABORT PPPoE Configure Voice Setting Phone Book Prefix Configuration DDNS Configuration DSCP Configuration Passwor ROM Configuration Flash Clean Reboot System Change First select login name as root or user then enter current password new password and confirm new passwo
35. he SIP Phone the LCD screen shows as below Now SIP Phone is running Boot sector program IP Phone Board Start Booting 2 When SIP Phone finishes boot program initialization User can see flashing greeting as below System Initializing 3 Then SIP Phone get into standby mode SIP Phone Proxy 10 10 10 AM The main LCD screen would be shown as similar as above Proxy means the SIP Phone is in Proxy Mode and when SIP Phone is connected to SNTP server on LCD will show current time captured from SNTP server 4 When SIP Phone is under peer to peer mode on LCD will show P2P instead of Proxy SIP Phone P2P 10 10 10 AM 5 After pressing the PSTN button the Proxy or P2P will be replaced by PSTN SIP SIP Phone Administration Guide Please notice that user must plug PSTN line in RJ 11 port when SIP Phone is in PSTN mode SIP Phone will always stay in IP mode after a PSTN call is finished SIP Phone will automatically return to IP mode IP Phone PSTN 10 10 10 AM 6 Press or gt to enter configuration mode then press OK button to enter sub menus press C can jump out current menu to previous level Call List Forward Type Phone Book Ringer Settings Network Advanced Settings protected by password Reboot oe a a SIP SIP Phone Administration Guide 2 LCD Menu Configuration User can set the following config
36. ion OK DSCP Configuration Password ROM Configuration Flash Clean Reboot System User Name Set PPPoE authentication User Name Password Set PPPoE authentication password Reboot After Remote Host Disconnection Enable Disable auto reboot after PPPoE disconnection If user enables this function after PPPoE being disconnected SIP Phone will automatically reboot to re connect and after reboot if 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 PPPoE Echo Enable or Disable PPPoE Echo function If user enabled this feature SIP Phone will send out echo request to check PPPoE connection status Please notice that if user disables this function SIP Phone cannot detect if PPPoE connection is still alive or not Other items for reference only cannot allow to be configured 32 SIP SIP Phone Administration Guide 5 Voice Setting Please refer to chapter 4 15 voice command Configuration Menu Codec 1st 2nd 3rd 4th 5th 6th Prioirity G723 1 v G29 y G 71lmu Law G AlALaw Ga 6 729 d Network Interface 4 G 723 1 G 729a G 729 G 711mu G 711A G 729b SIP Information Frame Size aims Gms Ems v d ms v d ms v Gms System Configuration G
37. ion Usage os rtptype dscp os slgtype dscp os prin rrpr iiab mode os print Example tos rtptype 10 sigtype O usr config E 54 SIP SIP Phone Administration Guide 17 ddns command This function is for Dynamic Domain Name Server service Once user register to one DDNS server he can specify domain name for the SIP Phone When SIP Phone reboot it will automatically update it s IP address to DDNS server In this way even SIP Phone is using dynamic IP address other endpoint can locate this SIP Phone by its domain name 1 print display DDNS overall information and configuration 2 enable to enable disable DDNS function ddns enable 0 1 O for disable and 1 for enable 3 server to set IP address of DDNS login server Now only one DDNS server is available www dyndns org 4 hostname to set the registered Domain Name of SIP Phone Ex ddns lp001 ddns org 5 id to set login ID of registered account to log in DDNS server passwd to set password of registered account to log in DDNS server 7 checkip to enable disable check IP function If SIP Phone is behind IP sharing when this function is enabled SIP Phone will check it s public IP address by asking IP address check server and send to DDNS server to update DDNS data If this function is disabled when SIP Phone is behind IP sharing it will send it s private IP address to DDNS server 8 checkipsrv1 2 to set IP address of
38. ion Guide Deactivate No Response Forward function 3 Unconditional It is included the above two types Whether the SIP Phone is in which status calls will be automatically forwarded to the assigned phone number A Activate Enter a forwarded phone number to activate Unconditional Forward function B Deactivate Deactivate Unconditional Forward function 3 Phone Book 1 List List all records of name telephone number and IP address in the phone address book 2 Edit Delete Edit or delete a record of name telephone number and IP address of the phone address book 3 New Add a new record of name telephone number and IP address of the phone address book 4 Ringer Settings 1 Volume User can adjust ring volume by press or gt on the keypad to decrease or increase ringer volume 2 Style There are three tone styles for SIP Phone Move the gt symbol by press or on the keypad to select the tone style preferred then press OK to confirm it 5 Network 1 Information SIP SIP Phone Administration Guide User can press or gt to check current network status 1 Mode Display current network connection mode of SIP Phone to be Static Fixed IP DHCP or PPPoE Note When SIP Phone is under DHCP mode then change to Static mode the following items IP address Subnet Mask Default Gateway will display empty after reboot user can see information again 2 IP M
39. k Interface Subnet Mask oss fo jo i fo SIP Information Default routing gateway o J a Ez System Configuration Get IP Mode Fixed IP DHCP PPPoE PPPoE Configure SNTP enable disable Voice Setting p saree Address sen b 95 iss gt j Phone Book Prefix Configuration cmi a DDNS Configuration IP Sharing enable disable DSCP Configuration k a A UPnP enable 6 disable eS meee Hy Password IP Sharing Server Address lo lo fo fo ROM Configuration Primary DNS Server fies 195 192 1 Secondary DNS server 168 an 1 Ji OK Flash Clean Reboot System IP Address Set IP Address of SIP Phone Subnet Mask Set the Subnet Mask of SIP Phone Default routing gateway Set Default routing gateway of SIP Phone Get IP Mode User has to set SIP Phone to use which network mode 1 Fixed IP User has to assign a fixed IP to SIP Phone 2 DHCP 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 26 SIP SIP Phone Administration Guide SIP Phone Configuration Menu Network Interface SIP Information System Co
40. load 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 63 SIP SIP Phone Administration Guide usr config rom ROM files updating commands Usage rom print app boot dsptest dspcore dspapp s TFTP FTP server ip f filename rom print print show versions of rom files optional app update main application code optional boot update main boot code n Eaa boot2m update 2m code optional dsptest update DSP testing code optional dspcore update DSP kernel code optional dspapp update DSP application code optional 5 IP address of TFTP FTP server mandatory f file name mandatory method download via TFTP FTP TFTP mode 0 FTP mode l ftp specify username and password for FTP Note This command can run select one option in app boot _ dsptest dspcore and dspapp Example rom method 1 rom ftp vwusr vwusr rom app s 192 168 4 101 f app bin usr config zi Command rom print can show current version installed in SIP Phone Download Method TFTP Boot Rom 1p201_boot 101 pplication Rom 1p201_sip0202 BIN DSP Ap 48302ce3 140 DSP Kerne 48302ck 140 DSP Test Code 483cbit bin z usr config al 64 SIP SIP Phone Administration Guide
41. me Alias Ce164 phonenumberlJl pbook C delete index pbook C modify Cindex Cip ipaddress name Alias e164 phonenumber print Displau phonebook data add Add an record to phonebook search Search an record in phonebook delete Delete an record from phonebook insert Insert an record to phonebook in specified position modify Modify an exist record Note If parameter end_record is omited only record start_record will be disp lay If both parameters end_record and start_record are omited all records will be display Range of ip address setting 0 0 0 0 255 255 255 255 Range of index setting value 11005 Example phook print 1 10 pbook print 1 pbook print pbook add name Test ip 210 59 163 202 e164 1001 pbook insert 5 name Test ip 210 59 163 202 e164 1001 pbook delete 5 pbook search ip 192 168 4 99 pbook modify 3 name Test ip 210 59 163 202 e164 1001 J ka usr config 6 commit command Save any changes after configuring the SIP Phone usr config commit his may take a few seconds please wait ommit to flash memory ok usr config 45 SIP SIP Phone Administration Guide 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 usr config ping usr config ping 192 168 2 107 PING 192 168 2 1807 56 data
42. ndex prefix numberlLdrop numberlLinsert number print Prefix drop insert information prefix The prefix of dialed number clrop Drop prefix Enable i Disable O gt insert Insert digits Example prefix add prefix 100 drop 1 insert 2000 prefix add prefix 100 drop 1 prefix add prefix 100 drop 0 insert 200 prefix delete 1 prefix modify 1 prefix 100 drop insert 300 usr confie J 21 rom command print show all current configurations and version information app boot dsptest dspcore dspapp upgrade main boot code main application code DSP testing code DSP kernel code DSP application code Ring Back Tone PCM file and Hold Tone Note After upgrade Application please remember to execute flash clean command which will clean all configurations become factory values except IP address 3 boot2m to upgrade 2mb rom file which includes all firmware file mentioned in item 2 58 SIP SIP Phone Administration Guide 4 s it is necessary to prepare TFTP FTP server IP address for upgrading firmware rom file 5 f the file name prepared for upgrading is necessary as well 6 method specify download method to be TFTP or FTP O 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 usr config rom RO
43. nfiguration PPPoE Configure Voice Setting Phone Book Prefix Configuration DDNS Configuration DSCP Configuration Password ROM Configuration Flash Clean Reboot System SIP Phone Configuration Menu Network Interface SIP Information System Configuration PPPoE Configure Voice Setting Phone Book Prefix Configuration DDNS Configuration DSCP Configuration Password ROM Configuration Flash Clean Reboot System IP Address 210 59 163 242 Subnet Mask Default routing gateway 210 59 163 254 Get IP Mode Fixed IP ci HCP PPPoE SNTP 6 enable disable SNTP Server Address GMT Je IP Sharing enable disable UPnP enable disable IP Sharing Server Address Primary DNS Server Secondary DNS Server Fox IP Address Subnet Mask Default routing gateway Get IP Mode Fixed IP DHC PPPoE SNTP enable disable SNTP Server Address GMT EFE a IP Sharing enable disable UPnP IP Sharing Server Address C enable disable Primary DNS Server Secondary DNS Server SIP SIP Phone Administration Guide Lox 27 SNTP 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
44. one 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 Enable or Disable PPPoE Echo function If user enabled this feature SIP Phone will send out echo reguest to check PPPoE connection status Please notice that if user disables this function SIP Phone cannot detect if PPPoE connection is still alive or not pppoe echo 0 1 A 0O MND gt usr config pppoe PPPOE device information and configuration print L id username pwd password print Display PPPoE device information id Connection user name pwd Connection password ci ag Reboot after remote host disconnection echo usr config 11 flash command This command will clean the configuration stored in the flash rom to default value and reboot the SIP Phone Note 1 After user upgrade new software version suggested to execute this command to make sure new software work well on SIP Phone 2 To execute the command flash clean all configuration of SIP Phone stored in flash will be cleaned It is authorized for the user whose login name is root only 48 SIP SIP Phone Administration Guide usr config flash Flash memory information and configuration Usage lash clean Note This command will clean the c
45. onfiguration stored in the flash and reboot it Ea usr fconfig 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 service set SIP 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 Please refer to 19 bureau command 4 keypad set DTMF type User can select DTMF type SIP Phone receive and transmit sysconf keypad 0 1 O for in band 1 for RFC2833 2 for SIP Info 5 2833type change RFC2833 Payload type 6 eod select end of dialing key e g set end of dial key as gt button after finished pressing dialing number then press gt will dial out sysconf eod 0 1 2 O for no end of dial key 1 for x button 2 for ff button 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 49 SIP SIP Phone Administration Guide usr config sysconf System information and configuration idtime digit keypad dtmf 2833type type eod digit sysconf print print Di
46. 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 Prefix String set prefix string If user ID contains alphabets user can set it as prefix string here 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 Line Number identify one number for the SIP Phone to register to the Proxy Line Account set user name of SIP Phone for registering User can set user name and password for registering If password is no need please set user name the same as line number or SIP Phone won t register successfully Line Password set password for registering SIP Port set SIP UDP port RTP Port set RTP port for sending voice data Expire set expire time of registration SIP Phone will keep re registering to proxy server before expire timed out 30 SIP SIP Phone Administration Guide 3 System Configuration Please refer to chapter 4 12 sysconf command SiP Phone MEUN Y Configuration cenad E A id A eypa e In Ban RFC2833 INFO Menu _ yp wi laah i PN i N Network Interface RFC2833 Payload Type 96 SIP Info
47. primary and secondary IP address check server 9 delay to set the update interval time SIP Phone will re update its IP address in this time 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 SIP Phone will update DDNS data immediately ddns force IP address of SIP Phone o Note 1 2 For now we only support DDNS server as www dyndns org User must register to DDNS server first and specify user name and password in ddns id and ddns passwd The default IP address of DDNS login server is member dyndns org User has to specify domain name applied for SIP Phone in ddns hostname The default IP address of check IP server in is checkip dyndns org 55 SIP SIP Phone Administration Guide 18 vlan command print display current VLAN configuration settings dev enable disable VLAN configuration vid set CPU VLAN ID number pcvid set PC port VLAN ID number WD Note Only when vid numbers are the same network can connect For example if PC A connect to PC port of SIP Phone A and vid of pcvid are different this PC can t reach this SIP Phone however if PC B connect to SIP Phone B and pcvid of SIP Phone B is the same with vid of SIP Phone A PC B can reach SIP Phone A pE PCA eas BT PCVID 2 SIP PHONE A SIP PHONE B VID 2 5 priority set CPU packets priority number
48. rd again to set new password Abort Press abort will clean all inputs 39 SIP SIP Phone Administration Guide 11 Rom Configuration Please refer to chapter 4 21 rom command StP phone e Configuration Menu FTP TFTP server IP Address Network Interface Target File name SIP Information Method TFTP System Configuration FTP Login name passwd Voice Setting Target File Type Application Image X Phone Book OK Prefix Configuration PPPoE Configure DDNS Configuration DSCP Configuration Password ROM Configuration Flash Clean Reboot System FTP TFTP Server IP Address Set TFTP server IP address Target File name Set file name prepared to upgrade Method 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 2 After upgrade Application please remember to execute Flash Clean which will clean all configurations become factory values except IP address 40 SIP SIP Phone Administration Guide 12 Flash Clean Please refer to 4 11 flash command SIP Phone BE R LL u LLI Lu u d C lt LL i IS O R Configuration SIP Phone will be reseted to factory default values Menu CLEAN
49. rmation Inter Digit Time 3 PPPoE Configure Voice Setting Phone Book Prefix Configuration DDNS Configuration DSCP Configuration Password ROM Configuration Flash Clean Reboot System End of Dial Digit CNONE CX ox Keypad DTMF Type set DTMF type User can select DTMF type SIP Phone transmits 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 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 End of Dial Digit select end of dialing key e g set end of dial key as button after finished pressing dialing number then press will dial out 31 SIP SIP Phone Administration Guide 4 PPPoE Configure Please refer to chapter 4 10 pppoe command Configuration Menu User Name pppoe Network Interface Password Pieioioioioiook SIP Information it Af iot f F 1 a Wr Reboot After Remote Host on off a Disconnection System Configuration MY ty S N PPPoE Echo enable disable PPPoE Configure Phone Book Protocol TCP IP Prefix Configuration Device PPP PPPoE DDNS Configurat
50. 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 For example in Taiwan the time zone should be set as 8 which means GMT 8 IP Sharing Enable it if SIP Phone is behind IP Sharing router UPnP Enable it if IP sharing or NAT device supports UPnP function so that no need to configure IP sharing or SIP Phone when SIP Phone is behind NAT device IP Sharing Server Address Set Public IP Address of IP Sharing router for SIP Phone to work behind IP sharing Primary DNS Server Set Primary Domain Name Server IP address User can set Domain Name Server IP address Once SIP Phone can connect with DNS server user can specify URL address instead of IP address for Proxy and phone book IP address Secondary DNS Server Set Secondary Domain Name Server IP address 28 SIP SIP Phone Administration Guide 2 SIP Information Please refer to chapter 4 13 SIP command Configuration Menu Run Mode Peer 2 Peer Proxy Network Interface Primary Proxy Address 10 1 1 2 SIP Information Primary Proxy Port 5060 Syst Confi tion PN Re ee Secondary Proxy Address null PPPoE Configure E Secondary Proxy Port 5060 Voice Setting Outbound Proxy Address null Phone Book ka Prefix Configuration Outbound Proxy Port 5060 DDNS Configuration
51. splay system overall information and configuration idtime My ie time 1 10 sec service Specify lTanphone service type 0 Normal service 1 HotLine service keypad Select DTMF type O In band 1 RFC28 33 2833type RFC2833 au Type range 96 128 inter used 100 102 105 eod End of Dial Digit setting O NONE 1 2 8 privacy Privacy Number Type 0 NONE 1 Japan Example sysconf keypad O eod 2 usr config 13 sip command 1 print display all current configurations 2 mode configure SIP Phone as Proxy or Peer to Peer Mode Usage sip mode 0 1 1 for Proxy mode 0 for Peer to Peer mode 3 px set Primary Proxy server IP address or URL address sip px IP address or URL of Proxy server 4 pxport set listening port of Primary Proxy server 5 px2 set secondary proxy server IP address or URL address sip px IP address or URL of Proxy server 6 px2port set listening port of Secondarry Proxy server 7 outpx set IP address of outbound proxy server After user set outbound proxy all packets form SIP Phone will be sent to outbound proxy server 8 outpxport set listening port of Outbound Proxy server 9 prefix set prefix string If user ID contains alphabets user can set it as prefix string here 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 ad
52. t User Name of SIP Phone to register to Proxy Server If Proxy server doesn t reguest specific User name please enter Line number here B Line Number Set Line Number of SIP Phone to register to Proxy Server C Password Set User Password of SIP Phone to register to Proxy Server This configuration is not necessary if Proxy server doesn t reguest client to set password user only has to set User Name the same as Line Number 2 Firmware Update 1 Download method There are two methods to download new version file please move the gt symbol by press or on the keypad to select TFTP or FTP method then press OK to confirm it 2 FTP TFTP Sever User has to offer one TFTP FTP server IP Address and set this IP Address via keypad The IP Address is necessary for upgrading SIP Phone new application rom file 3 FTP Account User has to input user name for FTP server login It is necessary for upgrading SIP Phone new application rom file via FTP method 4 FTP Password User has to input password for FTP server login lt is necessary for upgrading SIP Phone new application rom file via FTP method 5 File Name User has to press the file name of new application rom file prepared for upgrading 6 Application Version Show current version of application 20 SIP SIP Phone Administration Guide 7 Start to Upgrade Select YES or NO to start upgrade Note Download via LCD command can only upgrade new application rom file
53. tems for user usage help command J usr confies J hd mr aas as des as SS r r 43 SIP SIP Phone Administration Guide 2 guit command 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 Most frequently used debug flag are sip fsm msg etc usr config debug Debug message information and configuration Usage debug C add typel type2 open close status status Display the enabled debug flags add Add debug flag delete Remove specified debug flag open Start to show debug messages close Stop showing debug messages Example debug add sip msg debug open usr confies J a 4 reboot command After typing commit command type reboot to restart the SIP Phone Sometimes after user type reboot on terminal screen will display Data modified commit to flash rom which means SIP Phone will record call history or not Ex REDIAL outgoing and incoming call data 5 Dbook command This command is functional both in Proxy mode and Peer to Peer mode In proxy mode use speed dial or 10 DL button will dial out e 164 number in phone book In the other hand in peer to peer mode SI
54. urations by LCD keypad Note 1 Press TRANSFER before input data can switch characters to be capital or lowercase 2 Press SPEED before input data can switch input mode as character mode or digit mode for example user wants to enter IP address after pressing SPEED can enter digits directly 3 When user is inputting data press C will jump out to menu list press will clear previous input data 1 Call List If there is an unanswered IP call it will be Kept in message box MESSAGE LED will be flashing until user press MESSAGE to check miss call and re press MESSAGE to return to main screen 1 Missed Calls to see all missed calls in message box 2 Received Calls to see all received calls in message box 3 Dialed Numbers to see all dialed calls in message box 2 Forward Type 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 B Deactivate Deactivate Busy Forward function 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 A Activate Enter a forwarded phone number to activate no response forward function B Deactivate SIP SIP Phone Administrat
55. word 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 2 User must input lower case command but contents of configurations such as SIP alias or user name etc user can set as capital case 3 After any change of configuration please remember to do commit command to save changes and then reboot command to reboot system 1 help command Type help or man or to display all the command lists The following figure is shown all commands of SIP Phone usr config help help help man Lcommandl quit guit exit close debug show debug message reboot reboot local machine pbook Phonebook information and configuration commit commit flash rom data ping test that a remote host is reachable time show current time ifaddr internet address manipulation pppoe PPPoE stack manipulation Flash clean configuration from flash rom susconf Sustem information manipulation sip SIP information manipulation security Security information manipulation voice Voice information manipulation tos IP Packet ToS Type of Service values ddns Dynamic DNS update manipulation vlan VLAN configuration and information bureau Bureau line information manipulation prefix Prefix drop insert information manipulation rom ROM file update passud Password setting information and configuration auth Set configuration i

SIP-Phone User Manual

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