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1. Introduction

This is the User Guide for the I Series VoIP phone. 
This  phone  is  a  standard  SIP  (Session  Initiation 
Protocol)  phone,  and  some  configuration  is 
necessary before it is ready for use. This manual 
will  illustrate  how  to  set  up  the  phone  through 
keyboard and Web User Interface (UI).

1.1 Overview of Hardware

1.1.1 

The  two  RJ-45  network  interfaces  support 
10/100M Ethernet. The default WAN interface is a 
DHCP Client. The user connects the WAN interface 
to a Power Over Ethernet (POE) enabled switch 
to obtain power and an IP address. (Alternatively, 
the phone can be assigned a static IP address.) 
The  LAN  port  is  commonly  used  for  bridging  a 
guest computer to the network. You can use the 
default  administrator’s  user  name  “admin”  and 
password “admin” to login and setup the device’s 
configuration.

1.1.2

Only the WAN interface supports POE (802.3af).

1.2 Overview of Software

Network Protocol

  SIP  v1 (RFC2543)

  SIP  v2 (RFC3261)

  IP/TCP/UDP/RTP/RTCP

  IP/ICMP/ARP/RARP/SNTP

  TFTP  Client/DHCP  Client/PPPOE  Client

  Telnet/HTTP  Server

  DNS  Clients

Codec

  G.711:  64K  bit/s (PCM)

  G.723.1:  63k/5.3k  bit/s

  G.726:  16k/24k/32k/40k  bit/s (ADPCM)

  G.729A:  8k bit/s (CS-ACELP)

  G.729B:  Adds VAD &  CNG  to G.729

Voice Quality

  VAD: Voice Activity Detection

  CNG:  Comfortable  Noise  Generator

  LEC:  Line  Echo  Canceller

  Packet  Loss  Compensation

  Adaptive Jitter  Buffer

Call Function

  Call  Hold 

  Call  Forward

  Call  Waiting 

  Three-way  Conference

Tone

  Ring  Tone 

  Dial  Tone

  Ring  Back  Tone 

  Busy  Tone

Phone Function

  Volume Adjustment 

  Phone  Book

  Speed Dial  Key

IP Assignment

  IP  (Static  IP)  

 

  DHCP  

 

  PPPoE

Security

 

 

HTTP  1.1 Basic/Digest  Authentication  Web 
Setup

 

 

MD5 for SIP  Authentication  (RFC2069/
RFC2617)

QoS

  QoS Field

NAT Traversal

  STUN

Configuration

  Web Browser  

  Console/Telnet  

  Keypad

DTMF

  DTMF  RELAY 

  DTMF  SIP  Info

  DTMF  RFC  2833

SIP Server

 

 

Support  two SIP  servers  working  at the 
same time

  Provide  a backup SIP  server

Firmware Upgrade

  TFTP  

 

  HTTP  

 

  FTP

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www.teledexeurope.com    

I SERIES V oIP USER GUIDE

Summary of Contents for I SERIES

Page 1: ...I SERIES VoIP USER GUIDE For ND Series NDC Series and RediDocks ...

Page 2: ...hen the appliance is subsequently used 9 Unplug this product from the wall outlet and refer servicing to qualified service personnel under the following conditions When the power supply cord or plug is damaged or frayed If liquid has been spilled into the product If the product has been exposed to rain or water If the product does not operate normally by following the operating instructions Adjust...

Page 3: ... users should ensure that it is permissible to be connected to the facilities of the local telecommunications company The equipment must also be installed using an acceptable method of connection The customer should be aware that compliance with theaboveconditionsmaynotpreventdegradation of service in some situations Repairs to certified equipment should be coordinated by a representative designat...

Page 4: ...tain of the number of devices that may be connected to a line as determined by the total RENs contact the local telephone company ForproductsapprovedafterJuly23 2001 theREN for this product is a part of the product identifier thathastheformatUS AAAEQ TXXXX Thedigits represented by are the REN without a decimal point e g 03isaRENof0 3 Forearlierproducts the REN is separately shown on the label If t...

Page 5: ...ephone above the speaker grill is the wall desk mount clip Remove this clip by firmly pushing upward towards the hookswitch 2 Flip the clip over top to bottom so that the protruding edge is towards the top of the phone and reinsert into cutout This protruding edge will hold the handset 3 Turn the telephone over so the bottom is up facing you Place the telephone on a non abrasive surface to prevent...

Page 6: ...igital Map Configuration 19 3 5 8 Call Service 20 3 5 9 Memory Key Programming 21 3 5 10 MMI Filter 22 3 5 11 DSP Configuration Audio Settings 22 3 6 Configuration Management 22 3 7 Update Firmware 23 3 7 1 Update Configuration 23 3 7 2 Autoprovision Firmware Update 23 3 8 System Manage 24 3 8 1 Account Manage 24 3 8 2 Syslog Configuration 24 3 8 3 Phone Book 24 3 8 4 Time Set via Simple Network T...

Page 7: ...I SERIES V oIP USER GUIDE www teledexeurope com 7 I Series ND and NDC Series 1 Line VoIP Phone Map I Series ND and NDC Series 2 Line VoIP Phone Map ...

Page 8: ...8 www teledexeurope com I SERIES V oIP USER GUIDE I Series RediDock VoIP Phone Map ...

Page 9: ... DND Do Not Disturb Black List Limit List Auto answer Program settings through a standard web browser Remote management functionality via Telnet Classification management for common user s password and superuser s password StandardsandProtocols IEEE 802 3 802 3 u 10 Base T 100 Base TX PPPoE DHCP Client and Server Support G 711a u G729 G7231 5 3 6 3 audio Codec SIP RFC3261 RFC 2543 Support IAX2 TCP...

Page 10: ...are Network Protocol SIP v1 RFC2543 SIP v2 RFC3261 IP TCP UDP RTP RTCP IP ICMP ARP RARP SNTP TFTP Client DHCP Client PPPOE Client Telnet HTTP Server DNS Clients Codec G 711 64K bit s PCM G 723 1 63k 5 3k bit s G 726 16k 24k 32k 40k bit s ADPCM G 729A 8k bit s CS ACELP G 729B Adds VAD CNG to G 729 Voice Quality VAD Voice Activity Detection CNG Comfortable Noise Generator LEC Line Echo Canceller Pac...

Page 11: ...mber INDICATION The Messages key serves as an indicator that messages are present as well as a speed dial key to dial in and retrieve those messages If messagesarepresent presstheflashingredMWI key labeled Messages to go off hook and dial the programmed voicemail number First program the speed dial to the correct voice mail port of your PBX in the Web UI found under Advanced Memory Key MWI Number ...

Page 12: ...and guest guest 3 2 Current Status The Current Status page shows the working state of the VoIP phone The network part shows the connection state of the WAN interface and the LAN interface as well as the network setting DHCP or Static the MAC address of the phone and the Firmware version Here you can also see whether the phone is registered to the IP PBX server or not The Phone Number section shows...

Page 13: ... for the LAN DHCP Service Enable DHCP service on the LAN port After the user changes LAN IP the phone will automatically modify the DHCP Lease Table and save the configuration according to the IP and netmask information The DHCP server configuration won t take effect until you reboot the device NAT Enable Network Address Translation NAT for navigating through firewalls Bridge Mode Enable this opti...

Page 14: ...it is different from the SIP server Forward Type Call forwarding has three types Always Busy and No Answer You may select the type of forwarding here and then you must complete the configuration by Disabling Call Waiting Found in Advanced Call Service No answer If no answer after a set period of time NoAnswerTimeisfoundinAdvanced Call Service it will forward to the appointed phone Always The calle...

Page 15: ...ospitality environments The default is unchecked NAT Keep Alive Usually not applicable in Hospitality environments The default is unchecked KeepAlive isatermusedtodescribe the act of keeping a persistent connection Ports are opened and closed in IP communications A keep alive prompt command ensures the connection is NOT dropped but is instead kept open When the client sends another request it uses...

Page 16: ...alid Netmask Subnet mask of the lease table Gateway Default gateway of the lease table DNS Default DNS Domain Name Server of the lease table DNS Relay Enables the Domain Name Server relay function DHCP Lease Table Shows the IP MAC corresponding table assigned by the DHCP server Note This setting won t take effect unless you save the configuration and reboot the device 3 5 2 NAT CONFIGURATION ADVAN...

Page 17: ...ces of the SIP line UseSTUN TheSTUNsettingthatallows enables or forbids use 3 5 4 NET SERVICE HTTP Port Configures the Hyper Text Transfer Port the default HTTP port is 80 User may change this port to enhance the system s security the number must be between 1024 and 65000 For example if you were to change to port 34687 then when this port is changed you would use a generic IP address of format htt...

Page 18: ...oS is based on the 802 1p protocol The QoS is used to mark the network communication priority in the data link MAC sub layer An I Series phone will sort the packets using QoS and send it to the destination VLAN Enable If enabled the traffic over the WAN port and the bridged LAN port will separate voice signaling and data transmission This is an essential security feature in hospitality environment...

Page 19: ...ialing and when to send the string of numbers to the server I Series phones support the following digital map End With Use as the end of dialing Fixed Length When the length of the dialing matches the call will be sent Timeout Specify the timeout of the last digit dialed The call will be sent after the timeout represents the range of digits used It can be a range such as 1 4 or it can use commas s...

Page 20: ...wer call forward time setting Default is 20 seconds before a call forward P2P IP Prefix Peer to Peer IP Prefix Not applicable for Hospitality environments Sets the Prefix for Peer to Peer IP calls For example if you want to dial the number 192 168 1 119 you define the P2P IP Prefix as 192 168 1 So you dial only 119 to reach the number 192 168 1 119 Default is If there is no set the result disables...

Page 21: ...n the settings below in Memory 3 the transfer to 1107 will occur after the connection is established a pause will happen then digits are sent P equals a 1 second pause so PPP equals a 3 second pause then the digits 12345 are sent Call Park Retrieve allows the phone s user to send a call to a Park location setup within the PBX where another user may then Retrieve the call from a different phone It ...

Page 22: ...alerting ring as heard by the called party G729 Payload Length G729 payload length G 729 is an audio data compression algorithm for voice that compresses digital voice in packets of 10 milliseconds duration It is officially described as Coding of speech at 8 kbit s The payload is the user data being transmitted and the length is measured in milliseconds Signal Standard Signal standard for differen...

Page 23: ...on File Export Exports the configuration file Configuration File Import Imports the configuration file The I Series phone supports FTP and TFTP auto updating The gateway will automatically obtain the configuration file from your updated server if it is configured To obtain the original configuration file you can use the FTP TFTP back up as described above Put the configuration file in the root dir...

Page 24: ...capture on This information is highly technical and we recommend using the free software Wireshark for capturing network data packets instead of Syslog Warning if the power is cycled on the phone Syslog is set back to its default off state Click Apply after setting this up 3 8 3 PHONE BOOK Cetis VoIP phones have a standard base firmware that is used across several models Not all of the models have...

Page 25: ... require a reboot in order for the new settings to work Always save the configuration before the reboot otherwise the settings will return to the previous settings 4 Operating Method for Dialing 4 1 How to Dial an IP Phone You can make a call after your phone is properly set up Please confirm whether all the network wires are connected correctly and the phone credentials have been added under VOIP...

Page 26: ... into the VoIP SIP Configuration window to set up the SIP account information Server Address This is the IP address or FQDN Fully Qualified Domain Name of the IP PBX SIP Registration server Server Port This is the Internet Socket Port for SIP information The common UDP port for a SIP server is 5060 Default port is 5060 Account Name Username of your SIP account Often the same as the phone number Pa...

Page 27: ... enter 87 123 301 When a VLAN ID is set the phone should set VLAN Enabled and the Voice VLAN ID If necessary the LAN should be set to Bridge Mode also Show Firmware Version 39 FW Description When 39 is entered the phone will display or audibly respond with the firmware version number such as 1 7 240 235 Show TFTP Server Address 83 TF Description When entered the phone responds with the IP address ...

Page 28: ...dials In our example the SIP phone is programmed as extension 1001 The firmware version of the phone is circled in red Program Your Connection Under the VoIP SIP Configuration Screen The Server Address of the IP PBX Extension Number Account Name Phone Number Display Name Enable Register and Enable Message Waiting indicators are all programmed here Speed Dial and OneTouch Message Keys Are Programme...

Page 29: ...of Class 2 The auxiliary handset stations are powered via standard electrical plugs AutoSync Handset Registration Handsets can be registered most simply by performing AutoSync which is to place the handset into the cradle and the pairing will happen automatically Register Up To 4 Additional Handsets It is possible to register up to 4 additional remote handset kits to a single base station Repeat t...

Page 30: ... 2017 teledexeurope com Product specifications and descriptions in this document subject to change without notice Telephone 31 23 5670000 Fax 31 23 5670009 www teledexeurope com info stheurope com ...

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