Avaya E129V13 Sip Deskphone User Manual

AVAYA Sip Deskphone

User Manual

  E129 SIP DESKPHONE USER MANUAL                                               AVAYA   E129 SIP DESKPHONE   Small-Medium Business IP Phone
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 2 of 60                                   E129 SIP DESKPHONE User Manual Index GNU GPL INFORMATION ........................................................................... 5 CHANGE LOG ............................................................................................. 6 FIRMWARE VERSION 1.0.5.2 .............................................................................................................. 6 WELCOME .................................................................................................. 7 PRODUCT OVERVIEW ............................................................................... 8 FEATURE HIGHTLIGHTS ..................................................................................................................... 8 E129 SIP DESKPHONE TECHNICAL SPECIFICATIONS .................................................................... 8 INSTALLATION ......................................................................................... 10 EQUIPMENT PACKAGING ................................................................................................................. 10 CONNECTING YOUR PHONE ............................................................................................................ 10 SAFETY COMPLIANCES .................................................................................................................... 11 WARRANTY ......................................................................................................................................... 11 USING THE E129 SIP DESKPHONE ........................................................ 13 GETTING FAMILAR WITH THE LCD .................................................................................................. 13 GETTING FAMILAR WITH THE KEYPAD ........................................................................................... 14 MAKING PHONE CALLS..................................................................................................................... 15 HANDSET, SPEAKER AND HEADSET MODE ........................................................................... 15 2 CALLS WITH 1 SIP ACCOUNT ................................................................................................. 15 COMPLETING CALLS ................................................................................................................. 16 MAKING CALLS USING IP ADDRESSES ................................................................................... 17 ANSWERING PHONE CALLS ............................................................................................................ 19 RECEIVING CALLS...................................................................................................................... 19 DO NOT DISTURB ....................................................................................................................... 19 DURING A PHONE CALL .................................................................................................................... 20 CALL WAITING/CALL HOLD ....................................................................................................... 20 MUTE ............................................................................................................................................ 20 CALL TRANSFER ........................................................................................................................ 20 3-WAY CONFERENCING ............................................................................................................ 21 VOICE MESSAGES (MESSAGE WAITING INDICATOR) ........................................................... 23 CALL FEATURES ................................................................................................................................ 23
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 3 of 60                                   CUSTOMIZED LCD SCREEN & XML ................................................................................................. 25 CONFIGURATION GUIDE ......................................................................... 26 CONFIGURATION VIA KEYPAD ......................................................................................................... 26 CONFIGURATION VIA WEB BROWSER ........................................................................................... 31 DEFINITIONS ...................................................................................................................................... 31 STATUS PAGE DEFINITIONS ..................................................................................................... 32 ACCOUNT PAGE DEFINITIONS ................................................................................................. 32 SETTINGS/BASIC SETTINGS PAGE .......................................................................................... 40 SETTINGS/ADVANCED SETTINGS PAGE ................................................................................. 44 NAT SETTINGS ................................................................................................................................... 51 PUBLIC MODE .................................................................................................................................... 51 EDITING CONTACTS AND CLICK-TO-DIAL ...................................................................................... 52 UPGRADING AND PROVISIONING ......................................................... 56 UPGRADE VIA KEYPAD MENU ......................................................................................................... 56 UPGRAGE VIA WEB GUI .................................................................................................................... 56 NO LOCAL TFTP/HTTP SERVERS .................................................................................................... 57 CONFIGURATION FILE DOWNLOAD ................................................................................................ 57 RESTORE FACTORY DEFAULT SETTINGS ............................................ 59 EXPERIENCING THE E129 SIP DESKPHONE ........................................ 60     Table of Tables E129 SIP DESKPHONE User Manual  Table 1: E129 SIP DESKPHONE TECHNICAL SPECIFICATIONS ............................................................. 8 Table 2: E129 SIP DESKPHONE EQUIPMENT PACKAGING ................................................................... 10 Table 3: E129 SIP DESKPHONE CONNECTORS ..................................................................................... 11 Table 4: E129 SIP DESKPHONE DISPLAY DEFINITIONS ........................................................................ 13 Table 5: E129 SIP DESKPHONE LCD ICONS ........................................................................................... 13 Table 6: E129 SIP DESKPHONE KEYPAD DEFINITIONS ........................................................................ 14 Table 7: CALL FEATURES .......................................................................................................................... 24 Table 8: E129 SIP DESKPHONE CONFIGURATION MENU ..................................................................... 26
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 4 of 60                                   Table of Figures E129 SIP DESKPHONE User Manual  Figure 1: E129 SIP DESKPHONE Ports ..................................................................................................... 10 Figure 2: E129 SIP DESKPHONE Keypad MENU Flow ............................................................................. 30 Figure 3: E129 SIP DESKPHONE Web GUI - Contacts ............................................................................. 53 Figure 4: E129 SIP DESKPHONE Click-to-Dial .......................................................................................... 54     GUI Interface Examples E129 SIP DESKPHONE User Manual    1. Screenshot of Configuration Login Page 2. Screenshot of Status Page 3. Screenshot of Basic Setting Configuration Page 4. Screenshot of Advanced User Configuration Page 5. Screenshot of SIP Account Configuration Page 6. Screenshot of Saved Configuration Changes Page 7. Screenshot of Reboot Page
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 5 of 60                                   GNU GPL INFORMATION  E129 SIP DESKPHONE firmware contains third-party software licensed under the GNU General Public License (GPL). AVAYA uses software under the specific terms of the GPL. Please see the GNU General Public License (GPL) for the exact terms and conditions of the license.    AVAYA GNU GPL related source code can be downloaded from AVAYA web site from:
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 6 of 60                                   CHANGE LOG  This  section  documents  significant  changes  from  previous  versions  of  E129  SIP  DESKPHONE  user manuals.  Only  major  new  features  or  major  document  updates  are  listed  here.  Minor  updates  for corrections or editing are not documented here.  FIRMWARE VERSION 1.0.5.2    This is the initial version.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 7 of 60                                   WELCOME  Thank you for purchasing AVAYA E129 SIP DESKPHONE Small-Medium Business IP Phone. E129 SIP DESKPHONE is a next generation small-to-medium business IP phone that features single SIP account, up to 2 call appearances, a 128 x 40 graphical LCD, 3 XML programmable context-sensitive soft keys, dual network ports with integrated PoE (GXP1165 only), 3-way conference, and Electronic Hook Switch (EHS) with  Plantronics  headset.  The  GXP1160/1165  delivers  superior  audio  quality,  rich  and  leading  edge telephony  features,  personalized  information  and  customizable  application  service,  automated provisioning for easy deployment, advanced security protection for privacy, and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms. It is a perfect choice for small-to-medium businesses looking for a high quality, feature rich IP phone with highly affordable cost.     Caution:   Changes or modifications to  this  product not expressly approved by  Grandstream, or operation of this product in any way other than as detailed by this User Manual, could void your manufacturer warranty.  Warning:   Please do not use a different power adaptor with the E129 SIP DESKPHONE as it may cause damage to the products and void the manufacturer warranty.   This document is subject to change without notice. The latest electronic version of this user manual is available for download here:  Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose without the express written permission of AVAYA Networks, Inc. is not permitted.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 8 of 60                                   PRODUCT OVERVIEW                                   FEATURE HIGHTLIGHTS   128 x 40 pixel graphical LCD display;   Single SIP account, up to 2 call appearances, 3 XML programmable context-sensitive soft keys, 3-way conference;   Phonebook with up to 500 contacts and call history with up to 200 records;   Automated personal information service (e.g., local weather), personalized music sing tone/ring back tone;   Dual switched auto-sensing 10/100Mbps network ports, integrated PoE (GXP1165 only);   Automated provisioning using TR-069 or AES encrypted XML configuration file, SRTP and TLS for advanced security protection, 802,1x for media access control.  E129 SIP DESKPHONE TECHNICAL SPECIFICATIONS  Table 1: E129 SIP DESKPHONE TECHNICAL SPECIFICATIONS Protocols and Standards SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS (A record, SRV,  NAPTR),  DHCP,  PPPoE, TFTP, NTP, STUN, SIMPLE, TR-069, 802.1x, IPv6 Network Interfaces Dual switched 10/100Mbps ports, integrated PoE (GXP1165 only) Graphic Display 128 x 40 graphical LCD display Feature Keys 1  SIP  account,  3  XML  programmable  context  sensitive  soft  keys,  5 Navigation/Menu/Volume  keys,  9  dedicated  function  keys  for  PHONEBOOK, MESSAGE  (with  LED  indicator),  HOLD,  TRANSFER,  CONFERENCE,  FLASH, SPEAKERPHONE, VOLUME, SEND/REDIAL Voice Codec Support  for  G.723.1,  G.729A/B,  G.711u/a,  G.726-32,  G.722  (wide-band),  iLBC,   in-band and out-of-band DTMF (in audio, RFC2833, SIP INFO) Telephony Features Hold, transfer, forward, 3-way conference, downloadable phone book (XML, LDAP, up to 500 items), call waiting, call log (up to 200 records), off-hook auto dial, auto answer, click-to-dial, flexible dial plan, hot-desking, personalized music ringtones, server redundancy and fail-over Headset Jack RJ9, supporting Electronic Hook Switch (EHS) with Plantronics headsets Base Stand Yes, 1 angle position available
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 9 of 60                                   Wall Mountable Yes QoS Layer 2 (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS Security User  and  administrator  level  passwords,  MD5  and  MD5-sess  based authentication, AES encrypted configuration file, SRTP, TLS, 802.1x media access control Multi-language English,  German,  Italian,  French,  Spanish,  Portuguese,  Russian,  Croatian, Simplified and Traditional Chinese, Korean, Japanese and etc Upgrade and Provisioning Firmware  upgrade  via  TFTP/HTTP/HTTPS,  mass  provisioning  using  TR-069  or AES encrypted XML configuration file Power and Green Energy Efficiency Universal power adapter included Input: 100-240VAC 50-60Hz Output: +5VDC, 800mA Integrated Power-over-Ethernet (802.3af, GXP1165 only) Max power consumption 2.5W (universal power adapter) or 3W (PoE) Physical Unit dimension: 154mm (W) x 200mm (L) x 79mm (D) (handset onhook) Unit weight: 0.6kg Package weight: 1.03kg Temperature and Humidity Operating: 32-104oF / 0-40oC, 10-90% (non-condensing) Storage: 14-140oF / -10-60oC Package Content E129 SIP DESKPHONE phone, handset with cord, base stand, universal power supply, network cable, quick start guide Compliance FCC  Part  15  (CFR  47)  Class  B;  EN55022  Class  B,  EN55024,  EN61000-3-2, EN61000-3-3, EN60950-1; AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, RoHS; UL 60950 (power adapter)
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 10 of 60                                   INSTALLATION EQUIPMENT PACKAGING  Table 2: E129 SIP DESKPHONE EQUIPMENT PACKAGING Main Case Yes (1) Handset Yes (1) Phone Cord Yes (1) Power Adaptor Yes (1) Ethernet Cable Yes (1) Phone Stand Yes (1) Quick Start Guide Yes (1)  CONNECTING YOUR PHONE   Figure 1: E129 SIP DESKPHONE Ports
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 11 of 60                                    Table 3: E129 SIP DESKPHONE CONNECTORS Handset Port RJ9 handset connector port Headset Port RJ9  headset  connector  port,  supporting  EHS  (Electronic  Hook-Switch)  with Plantronics headsets LAN Port 10/100Mbps RJ-45 port connecting to Ethernet, integrated PoE (GXP1165 only) PC Port 10/100Mbps RJ-45 port for PC connection Power Jack 5V DC Power connector port   To set up the E129 SIP DESKPHONE, follow the steps below:  1.  Attach the phone stand to the back of the phone where there are slots; 2.  Connect the handset and main phone case with the phone cord; 3.  Connect the LAN port of the phone to the RJ-45 socket of a hub/switch or a router (LAN side of the router) using the Ethernet cable; 4.  Connect the  5V  DC  output  plug  to  the  power  jack  on  the  phone;  plug  the  power  adapter  into  an electrical outlet. If PoE switch is used on GXP1165 in step 3, this step could be skipped; 5.  The LCD will display provisioning or firmware upgrade information. Before continuing, please wait for the date/time display to show up; 6.  Using the keypad configuration menu or phone's embedded web server (Web GUI) by entering the IP address in web browser, you can further configure the phone.   SAFETY COMPLIANCES  The E129 SIP DESKPHONE phone complies with FCC/CE and various safety standards. The E129 SIP DESKPHONE power adapter is compliant with the UL standard. Use the universal power adapter provided with the E129 SIP DESKPHONE package only. The manufacturer’s warranty does not cover damages to the phone caused by unsupported power adapters.   WARRANTY  If the E129 SIP DESKPHONE phone was purchased from a reseller, please contact the company where the phone was  purchased for  replacement, repair or refund. If  the phone was purchased directly from Grandstream,  contact  the  AVAYA  Sales  and  Service  Representative  for  a  RMA  (Return  Materials
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 12 of 60                                   Authorization) number before the product is returned. AVAYA reserves the right to remedy warranty policy without prior notification.  Warning: Use the power adapter provided with the phone. Do not use a different power adapter as this may damage the phone. This type of damage is not covered under warranty.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 13 of 60                                   USING THE E129 SIP DESKPHONE GETTING FAMILAR WITH THE LCD  E129  SIP  DESKPHONE  has  a  dynamic  and  customizable  screen.  The  screen  displays  differently depending  on  whether  the  phone  is  idle  or  in  use  (active).  The  following  table  describes  the  items displayed on the E129 SIP DESKPHONE idle screen.  Table 4: E129 SIP DESKPHONE DISPLAY DEFINITIONS DATE AND TIME Displays the current date and time. It can be synchronized with Internet time servers. LOGO NAME Displays company logo name. This logo name can be customized via xml screen customization.  The  maximum  size  for  logo  name  is  26  characters  in  English (approximately). NETWORK STATUS Shows the status of network in the middle of the screen. It will indicate whether the network is down or starting. STATUS BAR Shows the status of the phone for registration status, call features and etc, using icons as shown in the next table. SOFTKEYS The softkeys are context sensitive and will change depending on the status of the phone. Typical functions assigned to softkeys are:  NextScr: Toggles among idle screen, weather information, IP Address and extension number;  Headset: Onhook/offhook using headset; or toggle to headset mode;  FwdAll: Unconditionally forwards the calls to another number;  Missed: Shows unanswered calls to this phone;  Redial: Redials the last dialed out number.  Table 5: E129 SIP DESKPHONE LCD ICONS  Registration Status: Registered.  Registration Status: Not Registered.  Handset Status.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 14 of 60                                   OFF - handset on hook                                                                 ON - handset off hook  Speaker Status. OFF - speaker off                                                               ON - speaker on  Headset Status. OFF - headset off ON - headset on  DND Status. OFF - Do Not Disturb disabled ON - Do Not Disturb enabled  Call Forward Status. OFF - Call Forward feature disabled ON - Call Forward feature enabled  MUTE Status. OFF - The active call is not muted ON - The active call is muted  SRTP Status. OFF - SRTP is not used ON - SRTP is used  GETTING FAMILAR WITH THE KEYPAD  The following table describes the buttons used on the E129 SIP DESKPHONE keypad.  Table 6: E129 SIP DESKPHONE KEYPAD DEFINITIONS  Place active call on hold, or resume the call on hold.  Transfer an active call to another number.  Establish 3-way conference with other 2 parties.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 15 of 60                                    Bring up a new line; or answer the second incoming call.  Speaker.  Send/Redial.     Send. Enter the digits and then press Send to dial out the number;  Redial. Redial when there is a previously dialed call.  Voicemail. Press to retrieve voice mails.  Phonebook. Brings phonebook on screen.  Navigation Keys/Menu.     Press the 4 navigation keys to move up/down/left/right;   Press  the  round  button  in  the  center  to  enter  Keypad  Configuration MENU when phone is in idle;   The  round  button  "MENU"  can  also  be  used  as  ENTER  key  when  in Keypad Configuration.  Volume. Press "-" or "+" to adjust the volume. 0 - 9, *, # Standard phone keypad.  MAKING PHONE CALLS HANDSET, SPEAKER AND HEADSET MODE  The E129 SIP DESKPHONE allows users to switch among handset, speaker or headset when making calls. Press the Hook Switch to switch to handset; press the Headset softkey to switch to headset; or press the Speaker button    to switch to speaker.  2 CALLS WITH 1 SIP ACCOUNT  E129 SIP DESKPHONE can support up to two lines "virtually" mapped to one SIP account. By picking up the
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 16 of 60                                   handset, the E129 SIP DESKPHONE will be in off hook state and the dial tone will be heard. To make a call,   dial out the number with the current line.  During the call, users can press the FLASH key to hold the current call and make/answer another call. If   they are 2 calls established, users can switch the two lines by pressing the FLASH key.  COMPLETING CALLS  There are several ways to complete a call on E129 SIP DESKPHONE.   On hook dialing. Enter the number when the phone is on hook and then send out.   When the phone is in idle, enter the number to be dialed out;   Take handset off hook; or Press Speaker button; or Press Headset softkey with headset plugged in;   The call will be dialed out.   Off hook and dial. Off hook the phone, enter the number and send out.   Take handset off hook; or Press Speaker button; or Press Headset softkey with headset plugged in;   You shall hear dial tone after off hook;   Enter the number;   Press SEND key    or # to dial out.   Redial. Redial the last dialed number.   Take handset off hook; or Press Speaker button; or Press Headset softkey with headset plugged in; or When the phone is in idle;   Press SEND key  , or the REDIAL softkey.   Via Call History. Dial the number logged in phone's call history.   Press MENU button to bring up the main menu;   Enter Call History and select "Answered Calls", "Missed Calls", "Transferred Calls" or "Forwarded Calls";
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 17 of 60                                     Select the entry you would like to call using the navigation "UP" and "DOWN" arrow keys;   Press SEND key    to dial out.   Via Phonebook. Dial the number from the phonebook.   Press MENU button to bring up the main menu;   Select and enter Phonebook;   Select the phonebook entry you would like to call using the navigation "UP" and "DOWN" arrow keys;   Press SEND key    to dial out.   Via Page/Intercom.   Take handset off hook; or Press Speaker button; or Press Headset softkey with headset plugged in;   You shall hear dial tone after off hook;   Press MENU button to switch the call screen from "Line x: Caller DIAL" to "Line x: Caller Paging";   Enter the number;   Press SEND key    or # to dial out.  Note:     After entering the number, the phone waits for the No Key Entry Timeout (Default timeout is 4 seconds, configurable via Web GUI) before dialing out. Press SEND key    or # key to override the No Key Entry Timeout;   If digits have been entered after handset is off hook, the SEND key will works as SEND instead of REDIAL;   By default, # can be used as SEND to dial the number out. Users could disable it by setting "Use # as Dial Key" to "No" from Web GUI->Account page;   For  Paging/Intercom, if  the  SIP Server/PBX  supports  the  feature and  has  Paging/Intercom feature code set up already, users might not necessarily need toggle to paging mode in the call screen on E129 SIP DESKPHONE. Simply dial the feature code with extension as a normal call.  MAKING CALLS USING IP ADDRESSES
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 18 of 60                                   Direct IP Call allows two phones to talk to each other in an ad-hoc fashion without a SIP proxy. VoIP calls can be made between two phones if:    Both phones have public IP addresses; or   Both phones are on the same LAN/VPN using private or public IP addresses; or   Both phones can be connected through a router using public or private IP addresses (with necessary port forwarding or DMZ).  To make a direct IP call, please follow the steps below:    Press MENU button to bring up main menu;   Select "Direct IP Call" using the navigation arrow keys;   Press MENU to enter the Direct IP Call mode;   Input the 12-digit target IP address (Please see example below);   Press  the  "More"  softkey to  make  sure  the  softkey selection  "IPv4"  or  "IPv6"  is  correctly  selected depending on your network environment;   Press "OK" softkey to dial.    For example:  If the target IP address is 192.168.1.60 and the port is 5062 (i.e., 192.168.1.60:5062), input the following: 192*168*1*60#5062. The * key represents the  dot (.), the # key represents colon (:). Wait for about 4 seconds and the phone will initiate the call.  Quick IP Call Mode:  The E129 SIP DESKPHONE also supports Quick IP Call mode. This enables the phone to make direct IP calls using only the last few digits (last octet) of the target phone's IP address. This is possible only if both phones are under the same LAN/VPN. This simulates a PBX function using the CMSA/CD without a SIP server. Controlled static IP usage is recommended.  To enable Quick IP Call Mode, go to E129 SIP DESKPHONE Web GUI->Advanced Setting page, set "Use Quick IP Call Mode" to "Yes". Click on "Update" on the bottom of the Web GUI page to take the change. To make Quick IP Call, take the phone off hook first. Then dial #xxx where x is 0-9 and xxx<255. Press # or SEND and a direct IP call to aaa.bbb.ccc.XXX will be completed. "aaa.bbb.ccc" is from the local IP address regardless of subnet mask. The number #xx or #x are also valid. The leading 0 is not required (but it's OK).  For example:    192.168.0.2 calling 192.168.0.3 -- dial #3 followed by # or “SEND”;
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 19 of 60                                     192.168.0.2 calling 192.168.0.23 -- dial #23 followed by # “SEND”;   192.168.0.2 calling 192.168.0.123 -- dial #123 followed by # “SEND”;   192.168.0.2: dial #3 and #03 and #003 results in the same call -- call 192.168.0.3.  Note:      The # will represent colon ":" in direct IP call rather than SEND key as in normal phone call;   If you have a SIP server configured, direct IP call still works. If you are using STUN, direct IP call will also use STUN;   Configure the "Use Random Port" to "No" when completing direct IP calls.  ANSWERING PHONE CALLS RECEIVING CALLS   Single incoming call. Phone rings with selected ring tone. Answer call by taking handset off hook, or using Speaker/Headset;  Multiple  incoming  calls.  When  another  call  comes  in  while  having  an  active  call,  the  phone  will produce a Call Waiting tone (stutter tone). Answer the incoming call by pressing the FLASHING key. The current active call will be put on hold automatically.  DO NOT DISTURB  Do Not Disturb can be enabled/disabled in Menu->Preference.    Press the Menu button and select "Preference" using navigation keys;   Press Menu button again to get into Preference options;   Select "Do Not Disturb" and press Menu button;   Use "UP"  and "DOWN" arrow keys to select and press  Menu button to  enable or disable "Do  Not Disturb" feature.  When Do Not Disturb feature is turned on, the DND icon will appear on the right side of the LCD. The incoming call will not be accepted or directly go into voicemail.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 20 of 60                                   DURING A PHONE CALL CALL WAITING/CALL HOLD   Hold. Place a call on hold by pressing the HOLD key  ;  Resume. Resume call by pressing the HOLD key   again;  Multiple calls. Automatically place active call on hold or switch between two calls by pressing the FLASH key  . Call waiting tone (stutter tone) will be audible on incoming call during the active call.  MUTE  During an active call, press the MUTE softkey to mute/unmute the microphone. The LCD will show "LINEx: TALKING" or "LINEx: MUTE" to indicate the mute status, with Mute icon displayed on the right side of the screen.  CALL TRANSFER  E129 SIP DESKPHONE supports Blind Transfer, Attended Transfer and Auto-Attended Transfer.   Blind Transfer.   During the first active call, press TRANSFER key    and dial the number to transfer to;   Press SEND key    or # to complete transfer of active call.   Attended Transfer.   During the first active call, press FLASH key  . The first call will be put on hold;   Enter the number for the second call and establish the call;
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 21 of 60                                     Press TRANSFER key  ;   Press FLASH key    to transfer the call.   Auto-Attended Transfer.   Set "Auto-Attended Transfer" to "Yes" under Web GUI->Advanced Settings page. And then click "Update" on the bottom of the page;   Establish one call first;   During the call, press TRANSFER key  . A new line will be brought up and the first call will be automatically placed on hold;   Enter the number and press SEND key    to establish the second call;   After  the  second  call  is  established,  press  TRANSFER  key    again.  The  call  will  be transferred;   If users press the SPLIT softkey before the call is transferred in the step above, the second call will be resumed.  Note:   To transfer calls across SIP domains, SIP service providers must support transfer across SIP domains.  3-WAY CONFERENCING  E129 SIP DESKPHONE can host 3-way conference call with another 2 parties.   Initiate a conference call.    Establish 2 calls with 2 parties respectively;   Press CONFERENCE key  ;   Press FLASH key  . 3-way conference will be established.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 22 of 60                                     Cancel Conference.    If after press the CONFERENCE key  , the user decides not to conference, press Cancel softkey;   This will resume the 2-way conversation with the current line.     Split and Re-conference.    During the 3-way conference, press HOLD key  . The conference call will be split and both calls will be put on hold separately;   Press FLASH key    to resume the 2-way conversation with the second established call;  If users would like to re-establish conference call, press the ReConf softkey.   End Conference.    Press HOLD key    to split the conference call. The conference call will be ended with both calls on hold; Or   Users could press the EndCall softkey or simply hang up the call to terminate the conference call.  E129  SIP  DESKPHONE  supports  Easy  Conference  Mode,  which  can  be  used  combined  with  the traditional way to establish the conference.   Initiate a conference call.    Establish 1 call;   Press CONFERENCE key    and a new line will be brought up using the same account;   Dial the number and press SEND key    to establish the second call;   Press CONFERENCE key    or press the ConfCall softkey to establish the conference.   Split and Re-conference.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 23 of 60                                     During the 3-way conference, press HOLD key  . The conference call will be split and both calls will be put on hold separately;   Press FLASH key    to resume the 2-way conversation with the second established call;   If users would like to re-establish conference call, press the ReConf softkey.     Cancel Conference.    If users decides not to conference after establishing the second call, press EndCall softkey;   This will end the second call and the screen will show the first call on hold.   End Conference.    Press HOLD key    to split the conference call. The conference call will be ended with both calls on hold; Or   Users could press the EndCall softkey or simply hang up the call to terminate the conference call.  Note:      The party that starts the conference call has to remain in the conference for its entire duration, you can put the party on mute but it must remain in the conversation. Also, this is not applicable when the feature "Transfer on call hangup" is turned on;   The  option  "Disable  Conference"  under  E129  SIP  DESKPHONE  Web  GUI->Settings->Advanced Settings has to be set to "No" to establish conference.  VOICE MESSAGES (MESSAGE WAITING INDICATOR)  A blinking red MWI (Message Waiting Indicator) indicates a message is waiting. Dial into the voicemail box   to retrieve the message by entering the voice mail number of the server or pressing the MESSAGE key   (Voice  Mail  User  ID  has  to  be  properly  configured  as  the  voice  mail  number  under  Web GUI->Account page). An IVR will prompt the user through the process of message retrieval.  CALL FEATURES
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 24 of 60                                   The E129 SIP DESKPHONE  supports traditional and advanced telephony features  including caller ID, caller ID with caller Name, call forward and etc.  Table 7: CALL FEATURES *30 Block Caller ID (for all subsequent calls)   Off hook the phone;   Dial *30. *31 Send Caller ID (for all subsequent calls)   Off hook the phone;   Dial *31. *67 Block Caller ID (per call)   Off hook the phone;   Dial *67 and then enter the number to dial out. *82 Send Caller ID (per call)   Off hook the phone;   Dial *82 and then enter the number to dial out. *70 Disable Call Waiting (per Call)   Off hook the phone;   Dial *70 and then enter the number to dial out. *71 Enable Call Waiting (per Call)   Off hook the phone;   Dial *71 and then enter the number to dial out. *72 Unconditional Call Forward. To set up unconditional call forward:   Off hook the phone;   Dial *72 and then enter the number to forward the call;   Press OK softkey or SEND key. *73 Cancel Unconditional Call Forward. To cancel the unconditional call forward:   Off hook the phone;   Dial *73;   Hang up the call. *90 Busy Call Forward. To set up busy call forward:
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 25 of 60                                     Off hook the phone;   Dial *90 and then enter the number to forward the call;   Press OK softkey or SEND key. *91 Cancel Busy Call Forward. To cancel the busy call forward:   Off hook the phone;   Dial *91;   Hang up the call. *92 Delayed Call Forward. To set up delayed call forward:   Off hook the phone;   Dial *92 and then enter the number to forward the call;   Press OK softkey or SEND key. *93 Cancel Delayed Call Forward. To cancel the delayed call forward:   Off hook the phone;   Dial *93;   Hang up the call.  CUSTOMIZED LCD SCREEN & XML  The  E129  SIP  DESKPHONE  IP  phone  supports  the  following  XML  applications.  Please  refer  to  the corresponding link for documentation and templates.    XML custom idle screen (customize idle screen logo, softkey layout, and etc.)    XML downloadable phonebook
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 26 of 60                                   CONFIGURATION GUIDE  The E129 SIP DESKPHONE can be configured via two ways:    LCD Configuration Menu using the phone's keypad;   Web GUI embedded on the phone using PC's web browser.  CONFIGURATION VIA KEYPAD  To configure via the LCD configuration menu using phone's keypad, follow the instructions below:   Enter  MENU  options.  When  the  phone  is  in  idle,  press  the  round  MENU  button  to  enter  the configuration menu;  Navigate  in  the  menu  options.  Press  the  arrow  keys  up/down/left/right  to  navigate  in  the  menu options;  Enter/Confirm selection. Press the round MENU button to enter the selected option;  Exit. Press LEFT arrow key to exit to the previous menu;   The phone automatically exits MENU mode with an incoming call, when the phone is off hook or the MENU mode if left idle for more than 20 seconds.  The MENU options are listed in the following table.  Table 8: E129 SIP DESKPHONE CONFIGURATION MENU Call History Displays  call  logs  for  answered  calls,  dialed  calls,  missed  calls,   transferred calls and forwarded calls. Status Displays  network  status,  account  registration  status,  software  version number, MAC address, hardware version number, P/N number.  Network status.   Press  to  enter  the  sub  menu  for  IP  setting  information (DHCP/Static IP/PPPoE), IPv4 address, IPv6 address,  Subnet Mask, Gateway and DNS server. Phone Book Displays phonebook. Users could add, edit, search and delete contacts here, or download phonebook XML to the phone. LDAP Directory Configures  LDAP  directory  options,  displays  LDAP  directory  by
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 27 of 60                                   searching. Instant Messages Displays received instant messages. Direct IP Call Makes direct IP call. Preference Preference sub menu includes the following options:  Do Not Disturb Enables/disables Do Not Disturb on the phone.  Forward Call Configures  call  forward  feature  on  selected  account,  forward type and number.  Ring Tone Configures different ring tones for incoming call.  Ring Volume Adjusts ring volume by pressing left/right arrow key.  LCD Contrast Adjusts LCD contrast by pressing left/right arrow key.  Download SCR XML Triggers  the  phone  to  download  the  XML  idle  screen  file immediately. The XML idle screen server path and downloading method  need  to  be  set  up  correctly  in  Web  GUI->Advanced Settings.  Erase Custom SCR Erases custom XML idle screen previously loaded on the phone. After erasing it, the phone will show default idle screen.  Display Language Selects the language to be displayed on the phone. Users could select  Automatic  for  local  language  based  on  IP  location  if available.  Time Settings Configures date and time on the phone. Config Config sub menu includes the following options:  SIP Configures SIP Proxy, Outbound Proxy, SIP User ID, SIP Auth ID, SIP Password, SIP Transport and Audio information to
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 28 of 60                                   register SIP account on the phone.  Upgrade Configures firmware server and config server for upgrading and provisioning the phone.    Factory Reset Resets the phone to factory default settings.  Layer 2 QoS Configures 802.1Q/VLAN Tag and priority value. Factory Functions Factory Functions sub menu includes the following options:  Audio Loopback Speak to the phone using speaker/handset/headset. If you can hear your voice, your audio is working fine. Press Menu button to exit audio loopback mode.  Diagnostic Mode All LEDs will light up. Press any key (except MENU key) on the keypad to display the button name in the LCD. Lift and put back the handset or press Menu button to exit diagnostic mode.  Keyboard Diagnostic Press all the available keys on the phone. The LCD will display the  name  for  the  keys  to  be  pressed  to  finish  the  keyboard diagnostic mode. Network Selects IP mode (DHCP/Static IP/PPPoE); Configures PPPoE account ID  and  password;  Configures  IP  address,  Netmask,  Gateway,  DNS Server 1 and DNS Server 2; Configures 802.1x mode. Call Features Configures call forward features for Forward All, Forward Busy, Forward No Answer and No Answer Timeout. Voice Mails Displays voicemail message information in the format below: new messages/all messages (urgent messages/all urgent messages) Reboot Reboot the phone. Exit Exit from this menu.   The following picture shows the keypad MENU configuration flow on E129 SIP DESKPHONE.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 30 of 60                                     Figure 2: E129 SIP DESKPHONE Keypad MENU Flow  Call History  Status  Phone Book  LDAP Directory  Instant Messages  Direct IP Call  Preference    Config  Factory Functions  Network  Call Features  Voice Mails  Reboot  Exit  MENU Answered Calls Dialed Calls Missed Calls Transferred Calls Forwarded Calls Clear All Back  Groups New Entry Search   Download Phonebook XML Delete All Entries Back First Name Last Name Number Acct Groups Confirm Add Cancel & Return Search LDAP Configuration Back Server Address Port Base User Name Password LDAP Number Filter LDAP Name Filter LDAP Version ... Do Not Disturb Forward Call Ring Tone Ring Volume LCD Contrast Download SCR XML Erase Custom SCR Display Language Time Settings Back SIP Upgrade Factory Reset Layer 2 QoS Back  Audio Loopback Diagnostic Mode Keyboard Diagnostic Back Enable DND Disable DND Back Default Ring Ring1 Ring2   Ring 3 Back Account SIP Proxy Outbound Proxy SIP User ID SIP Auth ID SIP Password SIP Transport Audio Save   Cancel  Firmware Server Config Server Upgrade Via Back  802.1Q/VLAN Tag Priority value Reset Vlan Config Back IP Setting PPPoE Settings IP Netmask Gateway DNS Server 1 DNS Server 2 802.1X Back  Account 1 Forward All Forward Busy Forward No Answer No Answer Timeout
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 31 of 60                                    CONFIGURATION VIA WEB BROWSER  The  E129  SIP  DESKPHONE  embedded  Web  server  responds  to  HTTP/HTTPS  GET/POST  requests. Embedded HTML pages allow a user to configure the IP phone through a Web browser such as Microsoft’s IE, Mozilla Firefox and Google Chrome.  To access the E129 SIP DESKPHONE Web GUI:  1.  Connect the computer to the same network as the phone; 2.  Make sure the phone is turned on and shows its IP address. You may check the IP address by pressing NextScr softkey or go to MENU->Status; 3. Open a Web browser on your computer; 4. Enter the phone’s IP address in the address bar of the browser; 5. Enter the administrator’s login and password to access the Web Configuration Menu.  Note:    The computer has to be connected to the same sub-network as the phone. This can be easily done by     connecting  the  computer  to  the  same  hub  or  switch  as  the  phone  connected  to.  In  absence  of  a hub/switch (or free ports on the hub/switch), please connect the computer directly to the PC port on the back of the phone;   If the phone is properly connected to a working Internet connection, the IP address of the phone will display  in  MENU->Status.  This  address  has  the  format:  xxx.xxx.xxx.xxx,  where  xxx  stands  for  a number from 0-255. Users will need this number to access the Web GUI. For example, if the phone has  IP  address  192.168.40.154,  please  enter  “http://192.168.40.154”  in  the  address  bar  of  the browser;   The default administrator password is set to "admin". The default user password is set to "123".  When changing any settings, always SUBMIT them by pressing the UPDATE button on the bottom of the  page.  After  submitting  the  changes  in  all  the  Web  GUI  pages,  reboot  the  phone  to  have  the changes take effect if necessary. All the options under Basic Setting and Account Setting, and most of the  options  under  Advanced  Settings  do  not  require  reboot  after  submitting  the  changes.  Under Advanced Setting, the parameters on network configuration require reboot after update.  DEFINITIONS  This section describes the options in the E129 SIP DESKPHONE Web GUI. As mentioned, you can log in
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 32 of 60                                   as an administrator or an end user.   Status: Displays the Account status, Network status, and System Info of the phone;  Account: To configure the SIP account;  Basic Settings: To configure basic network settings, time settings, Line keys, and etc;  Advanced Settings: To configure advanced network settings, upgrading and provisioning, language settings, call features, and etc.  STATUS PAGE DEFINITIONS MAC Address Global unique ID of device, in HEX format. The  MAC address will be used for provisioning and can be found on the label coming with original box and on the label located on the back of the device. IPv4 Address The IPv4 address obtained on the phone. IPv6 Address The IPv6 address obtained on the phone. Product Model Product model of the phone. Part Number Product part number. Software Version  boot: boot version number;   core: core version number;   base: base version number;   prog:  program  version  number.  This  is  the  main  firmware  release number, which is always used for identifying the software system of the phone;   dsp: DSP version number. System Up Time System up time since the last reboot. System Time Current system time on the phone system. Registered SIP account registration status. PPPoE Link Up PPPoE connection status. Service Status GUI and Phone service status: running or stopped. Core Dump Core dump file that could be downloaded for troubleshooting purpose.  ACCOUNT PAGE DEFINITIONS Account Name The name associated with the SIP account. SIP Server The URL or IP address, and port of the SIP server. This is provided by your VoIP service provider (ITSP).
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 33 of 60                                   Secondary SIP Server The URL or IP address, and port of the SIP server. This will be used when the primary SIP server fails. Outbound Proxy IP  address  or  Domain  name  of  the  Primary  Outbound  Proxy,  Media Gateway,  or  Session  Border  Controller.  It's  used  by  the  phone  for Firewall  or  NAT  penetration  in  different  network  environments.  If  a symmetric NAT is detected, STUN will not work and ONLY an Outbound Proxy can provide a solution. SIP User ID User  account  information,  provided  by  your  VoIP  service  provider (ITSP).  It's  usually  in  the  form  of  digits  similar  to  phone  number  or actually a phone number. Authenticate ID SIP service subscriber's Authenticate ID used for authentication. It can be identical to or different from the SIP User ID. Authenticate Password The account password required for the phone to authenticate with the ITSP (SIP) server before the account can be registered. After it is saved, this will appear as hidden for security purpose. Name The SIP server subscriber's name (optional) that will be used for Caller ID display. DNS Mode This  parameter  controls  how  the  Search  Appliance  looks  up  IP addresses  for  hostnames.  There  are  four  modes:  A  Record,  SRV, NATPTR/SRV, Use Configured IP. The default setting is "A Record". If the user wishes to locate the server by DNS SRV, the user may select "SRV" or "NATPTR/SRV". If "Use Configured IP" is selected, please fill in the three fields below:  Primary IP: The primary IP address where the phone sends DNS query to;  Backup IP 1;  Backup IP 2. TEL URI If the phone has an assigned PSTN telephone number, this field should be  set  to  "User=Phone".  Then  a  "User=Phone"  parameter  will  be attached  to  the  Request-Line  and  "TO"  header  in  the  SIP  request  to indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP request. The default setting is "Disable". SIP Registration Selects whether or not the phone will send SIP Register messages to the proxy/server. The default setting is "Yes". Unregister On Reboot If  set  to  "Yes",  the  SIP  user's  registration  information  will  be  cleared when  the  phone  reboots.  The  SIP  Contact  header  will  contain  "*"  to notify the server to unbind the connection. The default setting is "No".
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 34 of 60                                   Register Expiration Specifies  the  frequency  (in  minutes)  in  which  the  phone  refreshes  its registration with the specified registrar. The default value is 60 minutes. The maximum value is 64800 minutes (about 45 days). Reregister Before Expiration Specifies  the  time  frequency  (in  seconds)  that  the  phone  sends   re-registration request before the Register Expiration. The default value is 0. Local SIP Port Defines the local SIP port used to listen and transmit. The default value is 5060 for Account 1 and 5062 for Account 2. SIP Registration Failure Retry Wait Time Specifies  the  interval  to  retry  registration  if  the  process  is  failed.  The default value is 20 seconds. SIP T1 Timeout SIP T1 Timeout. The default setting is 0.5 seconds. SIP T2 interval SIP T2 Interval. The default setting is 4 seconds. SIP Transport Determines the network protocol used for the SIP transport. Users can choose from TCP, UDP and TLS. SIP URI Scheme when using TLS Specifies if "sip:" or "sips:" will be used when TLS/TCP is selected for SIP Transport. The default setting is "sips:". Use Actual Ephemeral Port in Contact with TCP/TLS Defines whether the actual ephemeral port in contact with TCP/TLS will be used or not. This is used when TLS/TCP is selected for SIP Transfer. The default setting is "No". Check Domain Certificates Defines  whether  the  domain  certificates  will  be  checked  or  not  when TLS/TCP is used for SIP Transport. The default setting is "No". Remove OBP from route Configures to remove outbound proxy from route. This is used for the SIP  Extension  to  notify  the  SIP  server  that  the  device  is  behind  a NAT/Firewall. Validate Incoming Messages Defines whether the incoming messages will be validated or not. The default setting is "No". Support SIP Instance ID Defines whether SIP Instance ID is supported or not. The default setting is "Yes". NAT Traversal This  parameter  configures  whether  the  NAT  traversal  mechanism  is activated.  Users  could  select  the  mechanism  from  No,  STUN, Keep-Alive, UPnP, Auto or VPN. If set to "STUN" and STUN server is configured, the phone will route according to the STUN server. If NAT type is Full Cone, Restricted Cone or Port-Restricted Cone, the phone will try to use public IP addresses and port number in all the SIP&SDP messages. The phone  will send empty SDP packet  to the SIP server periodically  to  keep  the  NAT  port  open  if  it  is  configured  to  be "Keep-Alive". Configure  this  to  be  "No"  if  an  outbound  proxy  is  used.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 35 of 60                                   "STUN" cannot be used if the detected NAT is symmetric NAT. SUBSCRIBE for MWI When set to "Yes", a SUBSCRIBE for Message Waiting Indication will be  sent  periodically.  The  phone  supports  synchronized  and non-synchronized MWI. The default setting is "No". SUBSCRIBE for Registration When  set  to  "Yes",  a  SUBSCRIBE  for  Registration  will  be  sent  out periodically. The default setting is "No". Feature Key Synchronization This feature is used for Broadsoft call feature synchronization. When it's enabled,  DND  and  Call  Forward  features  can  be  synchronized  with Broadsoft server. The default setting is "Disabled". Proxy-Require A  SIP  Extension  to  notify  the  SIP  server  that  the  phone  is  behind  a NAT/Firewall.  Do  not  configure  this  parameter  unless  this  feature  is supported on the SIP server. Voice Mail UserID Allows you to access voice messages by pressing the MESSAGE button on  the  phone.  This  ID  is  usually  the  VM  portal  access  number.  For example, in Asterisk server, 8500 could be used. Send DTMF Specifies  the  mechanism  to  transmit  DTMF  digits.  There  are  3 supported modes: in audio which means DTMF is combined in the audio signal (not very reliable with low-bit-rate codecs), via RTP (RFC2833), or via SIP INFO. DTMF Payload Type Configures  the  payload  type  for  DTMF  using  RFC2833.  The  default value is 101. Early Dial Selects whether or not to enable early dial. If it's set to "Yes", the SIP proxy must support 484 response. The default setting is "No". Dial Plan Prefix Sets the prefix added to each dialed number. Dial Plan A dial plan establishes the expected number and pattern of digits for a telephone number. This parameter configures the allowed dial plan for the phone.  Dial Plan Rules: 1.  Accepted Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d; 2.  Grammar: x - any digit from 0-9; a)  xx+ - at least 2 digit numbers b) xx. - only 2 digit numbers c) ^ - exclude d) [3-5] - any digit of 3, 4, or 5 e)  [147] - any digit of 1, 4, or 7 f)  <2=011> - replace digit 2 with 011 when dialing
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 36 of 60                                   g) | - the OR operand    Example 1: {[369]11 | 1617xxxxxxx} Allow 311, 611, and 911 or any 10 digit numbers with leading digits 1617;    Example 2: {^1900x+ | <=1617>xxxxxxx} Block  any  number  of  leading  digits  1900  or  add  prefix  1617  for  any dialed 7 digit numbers;    Example 3: {1xxx[2-9]xxxxxx | <2=011>x+} Allows  any  number  with  leading  digit  1  followed  by  a  3  digit  number, followed  by  any  number  between  2  and  9,  followed  by  any  7  digit number OR Allows any length of numbers with leading digit 2, replacing the 2 with 011 when dialed.  Example of a simple dial plan used in a Home/Office in the US: {  ^1900x.  |  <=1617>[2-9]xxxxxx  |  1[2-9]xx[2-9]xxxxxx  |  011[2-9]x.  | [3469]11 }  Explanation of example rule (reading from left to right):   ^1900x. - prevents dialing any number started with 1900;   <=1617>[2-9]xxxxxx  -  allows  dialing  to  local  area  code  (617) numbers by dialing 7 numbers and 1617 area code will be added automatically;   1[2-9]xx[2-9]xxxxxx |- allows dialing to any US/Canada Number with 11 digits length;  011[2-9]x - allows international calls starting with 011;   [3469]11 - allows dialing special and emergency numbers 311, 411, 611 and 911.  Note: In some cases where the user wishes to dial strings such as *123  to activate  voice  mail  or  other  applications  provided  by  their  service provider,  the  *  should  be  predefined  inside  the  dial  plan  feature.  An example dial plan will be: { *x+ } which allows the user to dial * followed by any length of numbers. Delayed Call Forward Wait Time Defines  the  timeout  (in  seconds)  before  the  call  is  forwarded  on  no answer. The default value is 20 seconds. Enable Call Features When enabled, Do No Disturb, Call Forward and other call features will
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 37 of 60                                   be supported locally provided ITSP support those features. The default setting  is  "Yes".  If  set  to  "No",  ForwardAll  softkey  will  be  hidden  for Account 1. Call Log Configures Call Log setting on the phone. You can log all calls, only log incoming/outgoing calls or disable call log. The default setting is "Log All Calls". Session Expiration The  SIP  Session  Timer  extension  that  enables  SIP  sessions  to  be periodically "refreshed"  via  a  SIP  request  (UPDATE,  or  re-INVITE). If there is no refresh via an UPDATE or re-INVITE message, the session will be terminated once the session interval expires. Session Expiration is  the  time  (in  seconds)  where  the  session  is  considered  timed  out, provided no successful session refresh transaction occurs beforehand. The default value is 180 seconds. Min-SE The minimum session expiration (in seconds). The default value is 90 seconds. Caller Request Timer If set to "Yes" and the remote party supports session timers, the phone will use a session timer when it makes outbound calls. Callee Request Timer If set to "Yes" and the remote party supports session timers, the phone will use a session timer when it receives inbound calls. Force Timer If Force Timer is set to "Yes", the phone will use the session timer even if the remote party does not support this feature. If Force Timer is set to "No", the phone will enable the session timer only when the remote party supports this feature. To turn off the session timer, select "No". UAC Specify Refresher As a Caller, select UAC to use the phone as the refresher; or select UAS to use the Callee or proxy server as the refresher. UAS Specify Refresher As a Callee, select UAC to use caller or proxy server as the refresher; or select UAS to use the phone as the refresher. Force INVITE The Session Timer can be refreshed using the INVITE method or the UPDATE method. Select "Yes" to use the INVITE method to refresh the session timer. Enable 100rel The use of the PRACK (Provisional Acknowledgment) method enables reliability  to  SIP  provisional  responses  (1xx  series).  This  is  very important in order to support PSTN internetworking. To invoke a reliable provisional  response,  the  100rel  tag  is  appended  to  the  value  of  the required header of the initial signaling messages. Account Ring Tone Allows users to configure the ringtone for the account. Users can choose from different ringtones from the dropdown menu.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 38 of 60                                   Matching Incoming Caller ID Specifies matching rules with number, pattern or Alert Info text. When the incoming caller ID or Alert Info matches the rule, the phone will ring with selected distinctive ringtone. Matching rules:     Specific caller ID number. For example, 8321123;   A defined pattern with certain length using x and + to specify, where x could be any digit from 0 to 9. Samples: xx+ : at least 2-digit number; xx : only 2-digit number; [345]xx: 3-digit number with the leading digit of 3, 4 or 5; [6-9]xx: 3-digit number with the leading digit from 6 to 9.   Alert Info text Users could configure the matching rule as certain text (e.g., priority) and select the custom ring tone mapped to it. The custom ring tone will be used if the phone receives SIP INVITE with Alert-Info header in the following format: Alert-Info: <http://127.0.0.1>; info=priority Distinctive Ringtones Selects  the  distinctive  ring  tone  for  the  matching  rule.  When  the incoming caller ID or Alert Info matches the rule, the phone will ring with the selected ring. Ring Timeout Defines the timeout (in seconds) for the rings on no answer. The default setting is 60 seconds. Send Anonymous If set to "Yes", the "From" header in outgoing INVITE messages will be set to anonymous, essentially blocking the Caller ID to be displayed. Anonymous Call Rejection If set to "Yes", anonymous calls will be rejected. The default setting is "No". Auto Answer If set to "Yes", the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep. Allow Auto Answer by Call-Info If set to "Yes", the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep, based on the SIP info header sent from the server/proxy. The default setting is "No". Refer-To Use Target Contact If  set  to  "Yes",  the  "Refer-To"  header  uses  the  transferred  target's Contact header information for attended transfer. The default setting is "No". Transfer on Conference Hangup Defines whether or not the call  is  transferred  to  the  other party if  the initiator of the conference hangs up. The default setting is "No". Check SIP User ID for incoming INVITE If set to "Yes", SIP User ID will be checked in the Request URI of the incoming INVITE. If it doesn't match the phone's SIP User ID, the call will be rejected. The default setting is "No".
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 39 of 60                                   Authenticate Incoming INVITE Defines whether the phone will challenge INVITE requests or not. When set to "Yes", the phone will challenge the INVITE for authentication with SIP 401 Unauthorized response. The PBX will need resend the SIP INVITE request with authentication credentials. The default setting is "No". Preferred Vocoder 7 different vocoder types are supported on the phone, including G.711 U-law (PCMU), G.711 A-law (PCMA), G.723.1, G.729A/B, G.722 (wide band), iLBC and G72-32. Users can configure vocoders in a preference list that is included with the same preference order in SDP message. SRTP Mode Enables the SRTP mode based on your selection. The default setting is "Disabled". Symmetric RTP Defines whether symmetric RTP is supported or not. The default setting is "No". Silence Suppression Controls the silence suppression/VAD feature of the audio codec G.723 and G.729. If set to "Yes", when silence is detected, a small quantity of VAD packets (instead of audio packets) will be sent during the period of no talking. If set to "No", this feature is disabled. The default setting is "No". Voice Frames Per TX Configures the  number of voice frames transmitted per packet. When configuring this, it should be noted that the "ptime" value for the SDP will change  with  different  configurations  here.  This  value  is  related  to  the codec used and the actual frames transmitted during the in payload call. For end users, it is recommended to use the default setting, as incorrect settings may influence the audio quality. No Key Entry Timeout (s) Defines the timeout (in seconds) for no key entry. If no key is pressed after  the  timeout,  the  digits  will  be  sent  out.  The  default  value  is  4 seconds. Use # as Dial Key Allows users to configure the "#" key as the "Send" key. If set to "Yes", the "#" key will immediately dial out the input digits. In this case, this key is essentially equivalent to the "Send" key. If set to "No", the "#" key is included as part of the dialing string. G723 Rate Selects encoding rate for G723 codec. The default value is 5.3kbps. G.726-32 Packing Mode Selects "ITU" or "IETF" for G726-32 packing mode. iLBC Frame Size Selects iLBC packet frame size. The default value is 30ms. iLBC Payload Type Specifies iLBC Payload type. The default value is 97. The valid range is between 96 and 127. Jitter Buffer Type Selects  either  Fixed  or  Adaptive  based  on  network  conditions.  The
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 40 of 60                                   default setting is "Adaptive". Jitter Buffer Length Selects Low, Medium, or High based on network conditions. The default setting is "Medium". Conference URI Configures  the  conference  URI  when  using  Broadsoft  N-way  calling feature. DND Call Feature On Configures DND feature code to turn on DND. DND Call Feature Off Configures DND feature code to turn off DND. Use Privacy Header Controls  whether  the  Privacy  Header  will  present  in  the  SIP  INVITE message or not. The default setting is "default", which is when "Huawei IMS" special feature is on, the Privacy Header will not show in INVITE. If set to "Yes", the Privacy Header will always show in INVITE. If set to "No", the Privacy Header will not show in INVITE. Use P-Preferred-Identity Header Controls whether the P-Preferred-Identity Header will present in the SIP INVITE message or not. The default setting is "default", which is when "Huawei IMS" special feature is on, the P-Preferred-Identity Header will not show in INVITE. If set to "Yes", the P-Preferred-Identity Header will always show in INVITE. If set to "No", the P-Preferred-Identity Header will not show in INVITE. Special Feature Different soft switch vendors have special requirements. Therefore users may need select special features to meet these requirements. Users can choose from Standard, Nortel MCS, Broadsoft, CBCOM, RNK, Sylantro or  Huawei  IMS  depending  on  the  server  type.  The  default  setting  is "Standard".  SETTINGS/BASIC SETTINGS PAGE  End User Password Allows  the  administrator  to  set  the  password  for  user-level  web  GUI access.  This  field  is  case  sensitive  with  a  maximum  length  of  30 characters. Confirm Password Confirms the end user password field to be the same as above. Internet Protocol Selects Prefer IPv4 or Prefer IPv6. IPv4 Address Type Allows users to configure the appropriate network settings on the phone to  obtain  IPv4  address.  Users  could  select  "DHCP",  "Static  IP"  or "PPPoE". By default, it is set to "DHCP". DHCP Host name (Option 12) Specifies  the  name  of  the  client.  This  field  is  optional  but  may  be required by some Internet Service Providers.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 41 of 60                                   DHCP Vendor Class ID   (Option 60) Used by clients and servers to exchange vendor class ID. Allow DHCP Option 120 to override SIP Server Enables DHCP Option 120 from local server to override the SIP Server on the phone. The default setting is "No". PPPoE Account ID Enter the PPPoE account ID. PPPoE Password Enter the PPPoE Password. PPPoE Service Name Enter the PPPoE Service Name. IPv4 Address Enter the IP address when static IP is used. Subnet Mask Enter the Subnet Mask when static IP is used for IPv4. Gateway Enter the Default Gateway when static IP is used for IPv4. DNS Server 1 Enter the DNS Server 1 when static IP is used for IPv4. DNS Server 2 Enter the DNS Server 2 when static IP is used for IPv4. Preferred DNS Server Enter the Preferred DNS Server for IPv4. IPv6 Address Type Allows users to configure the appropriate network settings on the phone to  obtain  IPv6  address.  Users  could  select  "Auto-configured"  or "Statically configured" for the IPv6 address type. Static IPv6 Address Enter  the  static  IPv6  address  when  Full  Static  is  used  in  "Statically configured" IPv6 address type. IPv6 Prefix Length Enter  the  IPv6  prefix  length  when  Full  Static  is  used  in  "Statically configured" IPv6 address type. IPv6 Prefix Enter the IPv6 Prefix (64 bits) when Prefix Static is used in "Statically configured" IPv6 address type. DNS Server 1 Enter the DNS Server 1 for IPv6. DNS Server 2 Enter the DNS Server 2 for IPv6. Preferred DNS server Enter the Preferred DNS Server for IPv6. 802.1x mode Allows the user to set 802.1x mode on the phone. The default value is disabled. Identity Enter  the  Identity  for  the  802.1x  mode  (EAP-MD5, EAP-PEAPv0/MSCHAPv2). 802.1x Secret/Private Key Password Enter  the  Secret/Private  Key  Password  for  802.1x  mode.  It  won't  be displayed for security protection purpose. 802.1x CA Certificate Upload the CA Certificate file for 802.1x mode. 802.1x Client Certificate Upload the Client Certificate for 802.1x mode. HTTP Proxy Specifies the HTTP proxy URL for the phone to send packets to. The proxy  server  will  act  as  an  intermediary  to  route  the  packets  to  the
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 42 of 60                                   destination. HTTPS Proxy Specifies the HTTPS proxy URL for the phone to send packets to. The proxy  server  will  act  as  an  intermediary  to  route  the  packets  to  the destination. Time Zone Configures the date/time used on the phone according to the specified time zone. Self-Defined Time Zone This parameter allows the users to define their own time zone.   The syntax is: std offset dst [offset], start [/time], end [/time]   Default is set to: MTZ+6MDT+5,M4.1.0,M11.1.0    MTZ+6MDT+5 This indicates a time zone with 6 hours offset with 1 hour ahead which is U.S central time. If it is positive (+) if the local time zone is west of the Prime  Meridian  (A.K.A:  International  or  Greenwich  Meridian)  and negative (-) if it is east.   M4.1.0,M11.1.0   The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec)   The 2nd number indicates the nth iteration of the weekday: (1st Sunday, 3rd Tuesday…)   The  3rd  number  indicates  weekday:  0,1,2,..,6(  for  Sun,  Mon, Tues, ... ,Sat)   Therefore, this example is the DST which starts from the First Sunday of April to the 1st Sunday of November. Enable Weather Update Configures  to  enable  or  disable  weather  update  on  the  phone.  The default setting is "Yes". If set to "No", the weather information screen will not show. City Code Configures  weather  city  code  for  the  phone  to  look  up  the  weather information.  The  default  setting  is  "Automatic"  and  the  weather information  will  be  obtained  based  on  the  IP  location  of  the  phone  if available.  Otherwise,  specify  the  self-defined  city  code.  For  example, USCA0638 is the city code for Los Angeles, CA, United States. Update Interval Specifies the weather update interval (in minutes). The default value is 15 minutes. Degree Unit Specifies the degree unit for the weather information to display on the phone. LCD Contrast Configures the LCD contrast level (from 0 to 20). The default value is 10. Date Display Format Configures the date display format on the LCD. The following formats are supported:
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 43 of 60                                    yyyy-mm-dd: 2012-07-02  mm-dd-yyyy: 07-02-2012  dd-mm-yyyy: 02-07-2012 Time Display Format Configures the time display in 12-hour or 24-hour format on the LCD. The default setting is in 12-hour format. Disable in-call DTMF Display When it's set to "Yes", the DTMF digits entered during the call will not display. The default setting is "No". Always Ring Speaker Configures  to  enable  or  disable  the  speaker  to  ring  when  headset  is used  on  "Toggle  Headset/Speaker"  mode.  If  set  to  "Yes",  when  the phone is in Headset "Toggle Headset/Speaker" mode, both headset and speaker will ring on incoming call. The default setting is "No". Headset Key Mode When headset is connected to the phone, users could use the HEADSET button in "Default Mode" or "Toggle Headset/Speaker".    Default Mode:   When the phone is in idle, press HEADSET button to off hook the phone and making calls by using headset. Headset icon will display on the left side of the screen in dialing/talking status.   When there is an incoming call, press HEADSET button to pick up the call using headset.   When there is an active call using headset, press HEADSET button to hang up the call.   When Speaker/Handset is being used in dialing/talking status, press HEADSET button to switch to headset. Press it again to hang up the call. Or press speaker/Handset to switch back to the previous mode.    Toggle Headst/Speaker:   When the phone is in idle, press HEADSET button to switch to Headset mode. The headset icon will display on the left side of the screen. In this mode, if pressing Speaker button or Line key to off hook the phone, headset will be used.   When there is an active call, press HEADSET button to toggle between Headset and Speaker. Write Timeout Defines the interval (in seconds) to save the call history to phone's flash. The default value is 300 seconds. Max Unsaved Log Defines the number of unsaved logs before written to phone's flash. The default value is 200 entries.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 44 of 60                                   Headset TX gain Configures the transmission gain of the headset. The default value is 0dB. Headset RX gain Configures the receiving gain of the headset. The default value is 0dB. Handset TX gain Configures the transmission gain of the handset. The default value is 0 dB.  SETTINGS/ADVANCED SETTINGS PAGE Admin Password Allows  users  to  change  the  admin  password.  The  password  field  is purposely hidden after clicking the Update button for security purpose. This field is case sensitive with a maximum length of 30 characters. Confirm Password Confirms the admin password field to be the same as above. Layer 3 QoS Defines  the  Layer  3  QoS  parameter.  This  value  is  used  for  IP Precedence, Diff-Serv or MPLS. The default value is 12. Layer 2 QoS 802.1Q/VLAN Tag Assigns the VLAN Tag of the Layer 2 QoS packets. The default value is 0. Layer 2 QoS 802.1p Priority Value Assigns the priority value of the Layer2 QoS packets. The default value is 0. Local RTP Port This parameter defines the local RTP port used to listen and transmit. It is the base RTP port for channel 0. When configured, channel 0 will use this port _value for RTP; channel 1 will use port_value+2 for RTP. Local RTP port ranges from 1024 to 65400  and  must be  even.  The  default value is 5004. Use Random Port When set to "Yes", this parameter will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple phones are behind the same full cone NAT. The default setting is "Yes" (This parameter must be set to "No" for Direct IP Calling to work). Keep-alive Interval Specifies how often  the  phone sends a blank UDP packet to  the  SIP server in order to keep the "ping hole" on the NAT router to open. The default setting is 20 seconds. Use NAT IP The NAT IP address used in SIP/SDP messages. This field is blank at the default settings. It should ONLY be used if it's required by your ITSP. STUN Server The IP address or Domain name of the STUN server. STUN resolution results  are  displayed  in  the  STATUS  page  of  the  Web  GUI.  Only non-symmetric NAT routers work with STUN. Firmware Upgrade and Provisioning Specifies how firmware upgrading and provisioning request to be sent: Always Check for New Firmware, Check New Firmware only when F/W
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 45 of 60                                   pre/suffix changes, Always Skip the Firmware Check. XML Config File Password The password for encrypting the XML configuration file using OpenSSL. This  is  required  for  the  phone  to  decrypt  the  encrypted  XML configuration file. HTTP/HTTPS User Name The user name for the HTTP/HTTPS server. HTTP/HTTPS Password The password for the HTTP/HTTPS server. Upgrade Via Allows users to choose the firmware upgrade method: TFTP, HTTP or HTTPS. Firmware Server Path Defines the server path for the firmware server. It could be different from the configuration server for provisioning. Config Server Path Defines the server path for provisioning. It could be different from the firmware server for upgrading. Firmware File Prefix Enables  your  ITSP  to  lock  firmware  updates.  If  configured,  only  the firmware  with  the  matching  encrypted  prefix  will  be  downloaded  and flashed into the phone. Firmware File Postfix Enables  your  ITSP  to  lock  firmware  updates.  If  configured,  only  the firmware  with  the  matching  encrypted  postfix  will  be  downloaded and flashed into the phone. Config File Prefix Enables your ITSP to lock configuration updates. If configured, only the configuration file with the matching encrypted prefix will be downloaded and flashed into the phone. Config File Postfix Enables your ITSP to lock configuration updates. If configured, only the configuration file with the matching encrypted postfix will be downloaded and flashed into the phone. Allow DHCP Option 43 and Option 66 Override Server If DHCP option 66 is enabled on the LAN side, the TFTP server can be redirected. The default setting is "Yes". Automatic Upgrade Enables automatic upgrade and provisioning. The default setting is "No". Hour of the Day (0-23) When "Automatic Upgrade" is set to "Yes, check for upgrade every day", configure the hour of the day when the upgrading/provisioning starts. Day of the Week (0-6) When  "Automatic  Upgrade"  is  set  to  "Yes,  check  for  upgrade  every week", configure the day of the week when the upgrading/provisioning starts. Authenticate Conf File Authenticates configuration file before acceptance. The default setting is "No". Enable TR-069 Enables TR-069. The default setting is "No". ACS URL URL for TR-069 Auto Configuration Servers (ACS).
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 46 of 60                                   TR-069 Username ACS username for TR-069. TR-069 Password ACS password for TR-069. Periodic Inform Enable Enables periodic inform. If set to "Yes", device will send inform packets to the ACS. The default setting is "No". Periodic Inform Interval Sets up  the periodic inform interval to send the inform  packets to the ACS. Connection Request Username The user name for the ACS to connect to the phone. Connection Request Password The password for the ACS to connect to the phone. Connection Request Port The port for the ACS to connect to the phone. CPE SSL Certificate The Cert File for the phone to connect to the ACS via SSL. CPE SSL Private Key The Cert Key for the phone to connect to the ACS via SSL. Phonebook XML Download Configures  to  enable  phonebook  XML  download.  Users  could  select HTTP/HTTPS/TFTP to download the phonebook file. The default setting is "No".   Phonebook XML Server Path Configures the server path to download the phonebook XML. This field could be IP address or URL, with up to 256 characters. Phonebook Download Interval Configures the phonebook download interval (in minutes). If it's set to 0, the  automatic  download  will  be  disabled.  The  default  value  is  0.  The valid range is 5 to 720 minutes. Remove Manually-edited Entries on Download If set to "Yes", when XML phonebook is downloaded, the entries added manually will be automatically removed. The default setting is "Yes". LDAP Directory: Server Address Configures the IP address or DNS name of the LDAP server. LDAP Directory: Port Configures the LDAP server port. LDAP Directory: Base Configures the LDAP search base. This is the location in the directory where the search is requested to begin.   Example:   dc=grandstream, dc=com   ou=Boston, dc=grandstream, dc=com   LDAP Directory: User Name Configures  the  bind  "Username"  for  querying  LDAP  servers.  Some LDAP servers allow anonymous binds in which case the setting can be left blank. LDAP Directory: Password Configures the  bind  "Password" for  querying  LDAP servers.  The  field can be left blank if the LDAP server allows anonymous binds. LDAP Number Filter Configures the filter used for number lookups.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 47 of 60                                   Examples: (|(telephoneNumber=%)(Mobile=%)  returns  all  records  which  has  the "telephoneNumber" or "Mobile" field starting with the entered prefix;  (&(telephoneNumber=%)  (cn=*))  returns  all  the  records  with  the "telephoneNumber" field  starting  with  the entered prefix and "cn" field set. LDAP Name Filter Configures the filter used for name lookups.   Examples:   (|(cn=%)(sn=%))  returns  all  records  which  has  the  "cn"  or  "sn"  field starting with the entered prefix;  (!(sn=%)) returns all the records which do not have the "sn" field starting with the entered prefix;  (&(cn=%)  (telephoneNumber=*))  returns  all  the  records  with  the  "cn" field starting with the entered prefix and "telephoneNumber" field set. LDAP Version Selects the protocol version for the phone to send  the bind requests. The default setting is "Version 3". LDAP Name Attributes Specify the "name" attributes of each record which are returned in the LDAP  search  result.  This  field  allows  the  users  to  configure  multiple space separated name attributes. Example: gn   cn sn description LDAP Number Attributes Specifies the "number" attributes of each record which are returned in the LDAP search result. This field allows the users to configure multiple space separated number attributes. Example: telephoneNumber   telephoneNumber Mobile  LDAP Display Name Configures the entry information to be shown on phone's LCD. Up to 3 fields can be displayed. Example:   %cn %sn %telephoneNumber  Max. Hits Specifies the maximum number of results to be returned by the LDAP server. If set to 0, server will return all search results. The default setting is 50.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 48 of 60                                   Search Timeout Specifies the interval (in seconds) for the server to process the request and client waits for server to return. The default setting is 30 seconds.   Sort Results Specifies  whether  the  searching  result  is  sorted  or  not.  The  default setting is "No". LDAP Lookup Configures  to  enable  LDAP  number  searching  when  dialing  and receiving calls. Lookup Display Name Configures  the  display  name  when  LDAP  looks  up  the  name  for incoming call or outgoing call. This field must be a subset of the LDAP Name Attributes. Example: gn   cn sn description Use Phonebook Key for LDAP Search If set to "Yes", the Phonebook Key   pressing will bring up LDAP search screen. Idle Screen XML Download Configures  to  enable  idle  screen  XML  download.  Users  could  select HTTP/HTTPS/TFTP to download the XML idle screen file. The default setting is "No". Download Screen XML At Bootup If set to  "Yes", the idle screen XML file will be downloaded when the phone boots up. The default setting is "No". User Custom Filename Specifies the custom file for the idle screen XML file to be downloaded. Idle Screen XML Server Path Configures the server path to download the idle screen XML file. This field could be IP address or URL, with up to 256 characters. Offhook Auto Dial Configures a User ID/extension to dial automatically when the phone is off hook.  The  phone  will  use  the  first  account  to  dial out. The  default setting is "No". Auto Recover From Abnormal Configures  whether  auto  recover  or  not  when  the  phone  is  running abnormal. The default setting is "Yes". Syslog Server The URL or IP address of the syslog server for the phone to send syslog to. Syslog Level Selects the level of logging for syslog. The default setting is None. There are 4 levels: DEBUG, INFO, WARNING AND ERROR. Syslog messages are sent based on the following events:     product model/version on boot up (INFO level);   NAT related info (INFO level);   sent or received SIP message (DEBUG level);     SIP message summary (INFO level);
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 49 of 60                                     inbound and outbound calls (INFO level);   registration status change (INFO level);  negotiated codec (INFO level);   ethernet link up (INFO level);   SLIC chip exception (WARNING and ERROR levels);   memory exception (ERROR level). Send SIP Log Configures whether the SIP log will be included in the syslog messages or not. The default setting is "No". NTP Server Defines the URL or IP address of the NTP server. The phone may obtain the date and time from the server. Allow DHCP Option 42 Override NTP Server Defines whether DHCP Option 42 should override NTP server or  not. When enabled, DHCP Option 42 will override the NTP server if it's set up on the LAN. The default setting is "Yes". Public Mode Configures  to  turn  on/off  public  mode  for  hot  desking  feature  on  the phone. If set to "Yes", users would need fill in the SIP Server address for account 1 as well. Then reboot the phone. When the phone boots up, users will need enter SIP User ID and Password on the LCD to login and use the phone.  Note: When the phone is in public mode login screen, press HOLD button will have the IP address of the phone displayed. SSL Certificate SSL Certificate used for SIP Transport in TLS/TCP. SSL Private Key SSL Private key used for SIP Transport in TLS/TCP. SSL Private Key Password SSL Private key password used for SIP Transport in TLS/TCP. System Ring Tone Configures  system  ring  tone.  The  default  value  is  North  American standard.  Users  could  adjust  system  ring  tone  frequencies  and cadences based on local telecom standard. Call Progresses Tones: Dial Tone Message Waiting Ring Back Tone Call-Waiting Tone Busy Tone Reorder Tone Configures  ring  or  tone  frequencies  based  on  parameters  from  local telecom.  The  default  value  is  North  American  standard.  Frequencies should  be  configured  with  known  values  to  avoid  uncomfortable  high pitch sounds.  Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]];   (Frequencies are in Hz and cadence on and off are in 10ms)   ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of silence. In order to set a continuous ring, OFF should be zero. Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 50 of 60                                   Up to three cadences are supported. Disable Call-Waiting Disables the call waiting feature. The default setting is "No". Disable Call-Waiting Tone Disables the call waiting tone when call waiting is on. The default setting is "No". Disable Direct IP Calls Disables Direct IP Call. The default setting is "No". Use Quick IP-Call mode When  set  to  "Yes",  users  can  dial  an  IP  address  under  the  same LAN/VPN segment by entering the last octet in the IP address. To dial quick  IP  call,  off  hook  the  phone  and  dial  #XXX  (X  is  0-9  and  XXX <=255),  phone  will  make  direct  IP  call  to  aaa.bbb.ccc.XXX  where aaa.bbb.ccc comes from the local IP address REGARDLESS of subnet mask. #XX or #X are also valid so leading 0 is not required (but OK). No SIP server is required to make quick IP call. The default setting is "No". Disable Conference Disables the Conference function. The default setting is "No". Disable Transfer Disables the Transfer function. The default setting is "No". Auto-Attended Transfer If  set  to  "Yes",  the  phone  will  use  attended  transfer  by  default.  The default setting is "No". In-call dial number on pressing transfer key Configures the  number for the phone  to  dial as DTMF during the call using TRANSFER button. Configuration via Keypad Menu Configures the  access  control  for  the  users  to  configure from  keypad Menu. There are three different options:  Unrestricted. All the options can be accessed in keypad Menu.  Basic settings only. The CONFIG option will not display for users to access in keypad Menu.  Constraint  Mode.  CONFIG,  FACTORY  FUNCTIONS  and NETWORK options will not display for users  to  access in keypad menu. Enable STAR key Keypad locking If set to "Yes", the keypad can be locked by pressing and holding the STAR  *  key  for  about  4  seconds.  A  lock  icon  will  show  indicating the keypad is locked. The default setting is "Yes".  Note: When the keypad is locked, users would need press and hold the STAR * key for about 4 seconds again and then enter the password to unlock it. Password to lock/unlock Configures the password to lock/unlock the keypad. The password field allows number with up to 32 characters. Offhook timeout If configured, when the phone is on hook, it will go off hook after the
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 51 of 60                                   timeout (in seconds). The default value is 30 seconds. China Telecom Mode Enables/Disables China Telecom Mode to use China Telecom  special features on the phone. Do Not Escape # as %23 in SIP URI Specifies whether to replace # by %23 or not for some special situations. The default setting is "No". Disable Telnet Disables Telnet access. The default setting is "No". PC Port Mode Configures the PC port mode. The default setting is "Enabled". When set to "Disabled", the PC port is turned off. When set to "Mirrored", the traffic in the LAN port will go through PC port as well so users could capture phone's trace by connecting a PC to the phone's PC port.     Display Language Selects display language on the phone. Download Device Configuration Click to download the device configuration file in .txt format.  NAT SETTINGS  If the devices are kept within a private network behind a firewall, we recommend using STUN Server. The following settings are useful in the STUN Server scenario:   STUN Server (under Advanced Settings page) Enter a STUN Server IP (or FQDN) that you may have, or look up a free public STUN Server on the internet and enter it on this field. If using Public IP, keep this field blank.   Use Random Ports (under Advanced Settings page) This setting depends on your network settings. When set to "Yes", it will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple GXPs are behind the same NAT. If using a Public IP address, set this parameter to "No".   NAT Traversal (under Account Setting page) Default setting is "No". Enable the device to use NAT traversal when it is behind firewall on a private network. Select Keep-Alive, Auto, STUN (with STUN server path configured too) or other option according to the network setting.  PUBLIC MODE
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 52 of 60                                   The E129 SIP DESKPHONE supports hot desking using public mode. Under public mode, users could login the phone with the SIP account User ID and password. Please follow the steps below to configure the phone for public mode:    Under Web GUI->Account 1 setting page, fill up the SIP server address for account 1. Click "Update" on the bottom of the page;   Under Web GUI->Advanced setting page, set Public Mode option to "Yes". Click "Update" and reboot the phone;  When phone boots up, SIP User ID and Password to register to the configured SIP server in account 1 will  be  required.  Enter  the  correct  account  information  to  log  in  to  the  phone.  When  entering  the account information, press softkey "123"/"abc" to toggle input method;   In login page, pressing HOLD button on the phone will show phone's IP address;   After using the phone, go to LCD MENU->LogOut to log off the public mode.  EDITING CONTACTS AND CLICK-TO-DIAL  From  E129  SIP  DESKPHONE  Web  GUI,  users  could  view  contacts,  edit  contacts,  or  dial  out  with Click-to-Dial feature    on the top of the Web GUI. In the following figure, the Contact page shows all the added contacts (manually or downloaded via XML phonebook). Here users could add new contact, export phonebook XML, import phonebook XML, search contact, filter contacts by group, edit selected contact, or dial the contact/number.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 53 of 60                                                     Figure 3: E129 SIP DESKPHONE Web GUI - Contacts  When clicking on the   icon on the top menu of the Web GUI, a new dialing window will show for you to enter the number. Once Dial is clicked, the phone will go off hook and dial out the number from selected account.   Click  to add  new contacts. Click to export phonebook  in XML format. Click to import phonebook XML file. Click to call this contact from  the phone. Click  to  download the  contact information in .vcf format   Click  to  edit  this contact. Click  to  select group  in  the dropdown menu. Click  to search  in phonebook. Click  to  input  number and  dial  from  available lines.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 54 of 60                                    Figure 4: E129 SIP DESKPHONE Click-to-Dial  Additionally, users could directly send the command for the phone to dial out by specifying the following URL in PC's web browser, or in the field as required in other call modules.  http://ip_address/cgi-bin/api-make_call?phonenumber=1234&account=0&password=admin  In the above link, replace the fields with   ip_address:                  Phone's IP Address.  phonenumber=1234:     The number for the phone to dial out  account=0:     The account index for the phone to make call. The index is 0 for account 1, 1 for account 2, 2 for account 3, and etc.  password=admin:      The admin login password of phone's Web GUI.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 55 of 60                                     SAVING THE CONFIGURATION CHANGES  After users makes changes to the configuration, press the Update button on the bottom of the Web GUI page. We recommend rebooting or powering cycle the IP phone after saving changes.   REBOOTING FROM REMOTE LOCATIONS  Press the  Reboot button on the bottom of the  web GUI page to reboot the phone remotely. The web browser will then display a reboot page with message "The device is rebooting now...". Wait for about 1 minute to log in again.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 56 of 60                                   UPGRADING AND PROVISIONING  The E129 SIP DESKPHONE can be upgraded via TFTP/HTTP/HTTPS by configuring the URL/IP Address for the TFTP/HTTP/HTTPS server and selecting a download method. Configure a valid URL for TFTP or HTTP; the server name can be FQDN or IP address.  Examples of valid URLs: firmware.grandstream.com fw.ipvideotalk.com/gs  There are two ways to setup a software upgrade server: The IVR Menu or the Web Configuration Interface.  UPGRADE VIA KEYPAD MENU  Follow the steps below to configure the upgrade server path via phone's keypad menu:    Press MENU button and navigate using Up/Down arrow to select Config;   In the Config options, select Upgrade;   Enter the firmware server path and select upgrade method. The server path could be in IP address format or FQDN format;   Press the "OK" softkey. A reboot message window will be prompt.   Reboot the phone to have the change take effect.  When  upgrading  starts, the  screen  will  show  upgrading  progress.  When  done  you  will  see  the  phone restarts again. Please do not interrupt or power cycle the phone when the upgrading process is on.  UPGRAGE VIA WEB GUI  Open a web browser on PC and enter the IP address of the phone. Then, login with the administrator username and password. Go to Settings->Advanced Settings page, enter the IP address or the FQDN for the upgrade server in "Firmware Server Path" field and choose to upgrade via TFTP or HTTP/HTTPS. Update the change by clicking the "Update" button. Then "Reboot" or power cycle the phone to update the new firmware.  When  upgrading  starts, the  screen  will  show  upgrading  progress.  When  done  you  will  see  the  phone restarts again. Please do not interrupt or power cycle the phone when the upgrading process is on.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 57 of 60                                    Firmware upgrades take around 60 seconds in a controlled LAN or 5-10 minutes over the Internet. We recommend completing firmware upgrades in a controlled LAN environment whenever possible.  NO LOCAL TFTP/HTTP SERVERS  For  users  that  would  like  to  use  remote upgrading  without  a local  TFTP/HTTP server, AVAYA  offers a NAT-friendly HTTP server. This enables users to download the latest software upgrades for their phone via this server. Please refer to the webpage:    Alternatively, users can download a free TFTP or HTTP server and conduct a local firmware upgrade. A free windows version TFTP server is available for download from : .    Instructions for local firmware upgrade via TFTP: 1.  Unzip the firmware files and put all of them in the root directory of the TFTP server; 2.  Connect the PC running the TFTP server and the phone to the same LAN segment; 3.  Launch the TFTP server and go to the File menu->Configure->Security to change the TFTP server's default setting from "Receive Only" to "Transmit Only" for the firmware upgrade; 4. Start the TFTP server and configure the TFTP server in the phone’s web configuration interface; 5.  Configure the Firmware Server Path to the IP address of the PC; 6.  Update the changes and reboot the phone.  End users can also choose to download a free HTTP server from http://httpd.apache.org/ or use Microsoft IIS web server.  Note:   When the phone boots up, it will send a TFTP or HTTP request to download the configuration file   "cfgxxxxxxxxxxxx" where "xxxxxxxxxxxx" is the MAC address of the phone. If it is a normal TFTP or HTTP upgrade, the following messages “TFTP Error from [IP ADRESS] requesting cfg000b82023dd4 : File does not exist. Configuration File Download” can be ignored in the TFTP/HTTP server log.  CONFIGURATION FILE DOWNLOAD  AVAYA SIP Devices can be configured via the Web Interface as well as via a Configuration File (binary or XML) through TFTP or HTTP/HTTPS. The "Config Server Path" is the TFTP or HTTP/HTTPS server path for the  configuration file. It  needs to be set  to a valid URL, either in  FQDN or IP address format. The "Config Server Path" can be the same or different from the "Firmware Server Path".
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 58 of 60                                    A  configuration  parameter  is  associated  with  each  particular  field  in  the  web  configuration  page.  A parameter consists of a Capital letter P and 2 to 3 (Could be extended to 4 in the future) digit numeric numbers. i.e., P2 is associated with the “Admin Password” in the Web GUI->Settings->Advanced Settings. For a detailed parameter list, please refer to the corresponding firmware release configuration template.  When a AVAYA Devices boots up or reboots, it will issue a request for a configuration XML file named "cfgxxxxxxxxxxxx.xml"  followed  by  a  file  named  "cfgxxxxxxxxxxxx",  where  "xxxxxxxxxxxx"  is  the  MAC address  of  the  device,  i.e.,  "cfg000b820102ab".  The  configuration  file  name  should  be  in  lower  case letters.  For more details on XML provisioning, please refer to:
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 59 of 60                                   RESTORE FACTORY DEFAULT SETTINGS  Warning:   Restoring  the  Factory  Default  Settings  will  delete  all  configuration  information  on  the  phone.  Please backup or print all the settings before you restore to the factory default settings. AVAYA is not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider.  Please follow the instructions below to reset the phone:    Press MENU button to bring up the keypad configuration menu;   Select "Config" and enter;   Select "Factory Reset";   A warning window will pop out to make sure a reset is requested and confirmed;   Press the "OK" softkey to confirm and the phone will reboot. To cancel the Reset, press Cancel softkey instead.
  FIRMWARE VERSION 1.0.5.2                          E129 SIP DESKPHONE USER MANUAL           Page 60 of 60                                   EXPERIENCING THE E129 SIP DESKPHONE  Please visit our website: to receive the most up- to-date updates on firmware releases, additional features, FAQs, documentation and news on new products.   We encourage you to browse our and for answers to your general questions.   If you have purchased our products  through  a  AVAYA  Certified  Partner  or  Reseller,  please  contact  them  directly  for  immediate support.    Our technical support staff is trained and ready to answer all of your questions. Contact a technical support member or to receive in-depth support. Thank you again for purchasing AVAYA IP phone, it will be sure to bring convenience and color to both your business and personal life.   FCC Caution:  Any Changes or modifications not expressly approved by the party responsible for compliance could void the user's   authority to operate the equipment.     This  device  complies  with  part  15  of  the  FCC  Rules.  Operation  is  subject  to  the  following  two conditions: (1) This device may not cause harmful interference, and (2) this device must accept any interference received, including   interference that may cause undesired operation.     Note: This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to   part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference in   a residential installation. This equipment generates, uses and can radiate radio frequency energy and, if not installed   and used in accordance with the instructions, may cause harmful interference to radio communications. However,   there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful   interference to radio or television reception, which can be determined by turning the equipment off and on, the user is   encouraged to try to correct the interference by one or more of the following measures:     —Reorient or relocate the receiving antenna.     —Increase the separation between the equipment and receiver.     —Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.     —Consult the dealer or an experienced radio/TV technician for help.

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