User Manual
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| Document ID | 2801393 |
| Application ID | j/ohEOu7vdEarSX/665cPA== |
| Document Description | User Manual |
| Short Term Confidential | No |
| Permanent Confidential | No |
| Supercede | No |
| Document Type | User Manual |
| Display Format | Adobe Acrobat PDF - pdf |
| Filesize | 78.4kB (979971 bits) |
| Date Submitted | 2015-11-03 00:00:00 |
| Date Available | 2015-11-06 00:00:00 |
| Creation Date | 2015-11-02 15:37:18 |
| Document Lastmod | 2015-11-03 11:44:36 |
| Document Title | User Manual |
| Document Author: | Grandstream |
AVAYA
E129 SIP DESKPHONE
Small-Medium Business IP Phone
E129 SIP DESKPHONE USER MANUAL
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
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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
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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
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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:
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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.
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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.
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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
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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)
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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
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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
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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.
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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.
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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.
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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
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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";
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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
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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”;
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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.
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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
Multiple calls. Automatically place active call on hold or switch between two calls by pressing the
FLASH key
again;
. 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
Press SEND key
and dial the number to transfer to;
or # to complete transfer of active call.
Attended Transfer.
During the first active call, press FLASH key
Enter the number for the second call and establish the call;
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. The first call will be put on hold;
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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
After the second call is established, press TRANSFER key
to establish the second call;
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
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. 3-way conference will be established.
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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
If users would like to re-establish conference call, press the ReConf softkey.
to resume the 2-way conversation with the second established call;
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
Dial the number and press SEND key
Press CONFERENCE key
and a new line will be brought up using the same account;
to establish the second call;
or press the ConfCall softkey to establish the conference.
Split and Re-conference.
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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
If users would like to re-establish conference call, press the ReConf softkey.
to resume the 2-way conversation with the second established call;
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
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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
*31
*67
*82
*70
*71
*72
*73
*90
Block Caller ID (for all subsequent calls)
Off hook the phone;
Dial *30.
Send Caller ID (for all subsequent calls)
Off hook the phone;
Dial *31.
Block Caller ID (per call)
Off hook the phone;
Dial *67 and then enter the number to dial out.
Send Caller ID (per call)
Off hook the phone;
Dial *82 and then enter the number to dial out.
Disable Call Waiting (per Call)
Off hook the phone;
Dial *70 and then enter the number to dial out.
Enable Call Waiting (per Call)
Off hook the phone;
Dial *71 and then enter the number to dial out.
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.
Cancel Unconditional Call Forward. To cancel the unconditional call forward:
Off hook the phone;
Dial *73;
Hang up the call.
Busy Call Forward. To set up busy call forward:
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*91
*92
*93
Off hook the phone;
Dial *90 and then enter the number to forward the call;
Press OK softkey or SEND key.
Cancel Busy Call Forward. To cancel the busy call forward:
Off hook the phone;
Dial *91;
Hang up the call.
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.
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
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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
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Configures LDAP directory options, displays LDAP directory by
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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
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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.
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MENU
Call History
Status
Answered Calls
Dialed Calls
Missed Calls
Transferred Calls
Forwarded Calls
Clear All
Back
Groups
New Entry
Search
Download Phonebook XML
Delete All Entries
Back
Phone Book
LDAP
Search
LDAP Configuration
Back
Directory
Instant
Messages
Direct IP Call
Preference
Config
Factory
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
Functions
Network
Audio Loopback
Diagnostic Mode
Keyboard Diagnostic
Back
Call Features
Voice Mails
Reboot
Exit
IP Setting
PPPoE Settings
IP
Netmask
Gateway
DNS Server 1
DNS Server 2
802.1X
Back
Account 1
First Name
Last Name
Number
Acct
Groups
Confirm Add
Cancel & Return
Server Address
Port
Base
User Name
Password
LDAP Number Filter
LDAP Name Filter
LDAP Version
...
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
Forward All
Forward Busy
Forward No Answer
No Answer Timeout
Figure 2: E129 SIP DESKPHONE Keypad MENU Flow
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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
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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
Global unique ID of device, in HEX format. The MAC address will be
MAC Address
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
SIP Server
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The name associated with the SIP account.
The URL or IP address, and port of the SIP server. This is provided by
your VoIP service provider (ITSP).
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Secondary SIP Server
The URL or IP address, and port of the SIP server. This will be used
when the primary SIP server fails.
IP address or Domain name of the Primary Outbound Proxy, Media
Gateway, or Session Border Controller. It's used by the phone for
Outbound Proxy
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.
User account information, provided by your VoIP service provider
SIP User ID
(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.
The account password required for the phone to authenticate with the
Authenticate Password
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.
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
DNS Mode
"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.
If the phone has an assigned PSTN telephone number, this field should
be set to "User=Phone". Then a "User=Phone" parameter will be
TEL URI
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".
If set to "Yes", the SIP user's registration information will be cleared
Unregister On Reboot
when the phone reboots. The SIP Contact header will contain "*" to
notify the server to unbind the connection. The default setting is "No".
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Specifies the frequency (in minutes) in which the phone refreshes its
Register Expiration
registration with the specified registrar. The default value is 60 minutes.
The maximum value is 64800 minutes (about 45 days).
Specifies the time frequency (in seconds) that the phone sends
Reregister Before Expiration
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
Specifies the interval to retry registration if the process is failed. The
Wait Time
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
Specifies if "sip:" or "sips:" will be used when TLS/TCP is selected for
TLS
SIP Transport. The default setting is "sips:".
Use Actual Ephemeral Port in
Contact with TCP/TLS
Check Domain Certificates
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".
Defines whether the domain certificates will be checked or not when
TLS/TCP is used for SIP Transport. The default setting is "No".
Configures to remove outbound proxy from route. This is used for the
Remove OBP from route
SIP Extension to notify the SIP server that the device is behind a
NAT/Firewall.
Validate Incoming Messages
Support SIP Instance ID
Defines whether the incoming messages will be validated or not. The
default setting is "No".
Defines whether SIP Instance ID is supported or not. The default setting
is "Yes".
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
NAT Traversal
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.
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"STUN" cannot be used if the detected NAT is symmetric NAT.
When set to "Yes", a SUBSCRIBE for Message Waiting Indication will
SUBSCRIBE for MWI
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".
This feature is used for Broadsoft call feature synchronization. When it's
Feature Key Synchronization
enabled, DND and Call Forward features can be synchronized with
Broadsoft server. The default setting is "Disabled".
A SIP Extension to notify the SIP server that the phone is behind a
Proxy-Require
NAT/Firewall. Do not configure this parameter unless this feature is
supported on the SIP server.
Allows you to access voice messages by pressing the MESSAGE button
Voice Mail UserID
on the phone. This ID is usually the VM portal access number. For
example, in Asterisk server, 8500 could be used.
Specifies the mechanism to transmit DTMF digits. There are 3
Send DTMF
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
Early Dial
Dial Plan Prefix
Configures the payload type for DTMF using RFC2833. The default
value is 101.
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".
Sets the prefix added to each dialed number.
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;
Dial Plan
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)
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<2=011> - replace digit 2 with 011 when dialing
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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
Defines the timeout (in seconds) before the call is forwarded on no
Time
answer. The default value is 20 seconds.
Enable Call Features
When enabled, Do No Disturb, Call Forward and other call features will
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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.
Configures Call Log setting on the phone. You can log all calls, only log
Call Log
incoming/outgoing calls or disable call log. The default setting is "Log All
Calls".
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
Session Expiration
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
Caller Request Timer
Callee Request Timer
The minimum session expiration (in seconds). The default value is 90
seconds.
If set to "Yes" and the remote party supports session timers, the phone
will use a session timer when it makes outbound calls.
If set to "Yes" and the remote party supports session timers, the phone
will use a session timer when it receives inbound calls.
If Force Timer is set to "Yes", the phone will use the session timer even if
Force Timer
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
UAS 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.
As a Callee, select UAC to use caller or proxy server as the refresher; or
select UAS to use the phone as the refresher.
The Session Timer can be refreshed using the INVITE method or the
Force INVITE
UPDATE method. Select "Yes" to use the INVITE method to refresh the
session timer.
The use of the PRACK (Provisional Acknowledgment) method enables
reliability to SIP provisional responses (1xx series). This is very
Enable 100rel
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
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Allows users to configure the ringtone for the account. Users can choose
from different ringtones from the dropdown menu.
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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;
Matching Incoming Caller ID
[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: ; info=priority
Selects the distinctive ring tone for the matching rule. When the
Distinctive Ringtones
incoming caller ID or Alert Info matches the rule, the phone will ring with
the selected ring.
Ring Timeout
Send Anonymous
Anonymous Call Rejection
Auto Answer
Defines the timeout (in seconds) for the rings on no answer. The default
setting is 60 seconds.
If set to "Yes", the "From" header in outgoing INVITE messages will be
set to anonymous, essentially blocking the Caller ID to be displayed.
If set to "Yes", anonymous calls will be rejected. The default setting is
"No".
If set to "Yes", the phone will automatically turn on the speaker phone to
answer incoming calls after a short reminding beep.
If set to "Yes", the phone will automatically turn on the speaker phone to
Allow Auto Answer by Call-Info
answer incoming calls after a short reminding beep, based on the SIP
info header sent from the server/proxy. The default setting is "No".
If set to "Yes", the "Refer-To" header uses the transferred target's
Refer-To Use Target Contact
Contact header information for attended transfer. The default setting is
"No".
Transfer on Conference
Defines whether or not the call is transferred to the other party if the
Hangup
initiator of the conference hangs up. The default setting is "No".
Check SIP User ID for
incoming INVITE
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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".
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Defines whether the phone will challenge INVITE requests or not. When
set to "Yes", the phone will challenge the INVITE for authentication with
Authenticate Incoming INVITE
SIP 401 Unauthorized response. The PBX will need resend the SIP
INVITE request with authentication credentials. The default setting is
"No".
7 different vocoder types are supported on the phone, including G.711
Preferred Vocoder
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
Symmetric RTP
Enables the SRTP mode based on your selection. The default setting is
"Disabled".
Defines whether symmetric RTP is supported or not. The default setting
is "No".
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
Silence Suppression
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".
Configures the number of voice frames transmitted per packet. When
configuring this, it should be noted that the "ptime" value for the SDP will
Voice Frames Per TX
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.
Defines the timeout (in seconds) for no key entry. If no key is pressed
No Key Entry Timeout (s)
after the timeout, the digits will be sent out. The default value is 4
seconds.
Allows users to configure the "#" key as the "Send" key. If set to "Yes",
Use # as Dial Key
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
Jitter Buffer Type
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Specifies iLBC Payload type. The default value is 97. The valid range is
between 96 and 127.
Selects either Fixed or Adaptive based on network conditions. The
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default setting is "Adaptive".
Jitter Buffer Length
Conference URI
Selects Low, Medium, or High based on network conditions. The default
setting is "Medium".
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.
Controls whether the Privacy Header will present in the SIP INVITE
message or not. The default setting is "default", which is when "Huawei
Use Privacy Header
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.
Controls whether the P-Preferred-Identity Header will present in the SIP
INVITE message or not. The default setting is "default", which is when
Use P-Preferred-Identity
"Huawei IMS" special feature is on, the P-Preferred-Identity Header will
Header
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.
Different soft switch vendors have special requirements. Therefore users
may need select special features to meet these requirements. Users can
Special Feature
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
Allows the administrator to set the password for user-level web GUI
End User Password
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.
Allows users to configure the appropriate network settings on the phone
IPv4 Address Type
to obtain IPv4 address. Users could select "DHCP", "Static IP" or
"PPPoE". By default, it is set to "DHCP".
DHCP Host name (Option 12)
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Specifies the name of the client. This field is optional but may be
required by some Internet Service Providers.
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DHCP Vendor Class ID
(Option 60)
Used by clients and servers to exchange vendor class ID.
Allow DHCP Option 120 to
Enables DHCP Option 120 from local server to override the SIP Server
override 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.
Allows users to configure the appropriate network settings on the phone
IPv6 Address Type
to obtain IPv6 address. Users could select "Auto-configured" or
"Statically configured" for the IPv6 address type.
Static IPv6 Address
IPv6 Prefix Length
IPv6 Prefix
Enter the static IPv6 address when Full Static is used in "Statically
configured" IPv6 address type.
Enter the IPv6 prefix length when Full Static is used in "Statically
configured" IPv6 address type.
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
Identity
Allows the user to set 802.1x mode on the phone. The default value is
disabled.
Enter
the
Identity
for
the
802.1x
mode
(EAP-MD5,
EAP-PEAPv0/MSCHAPv2).
802.1x Secret/Private Key
Enter the Secret/Private Key Password for 802.1x mode. It won't be
Password
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
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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
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destination.
Specifies the HTTPS proxy URL for the phone to send packets to. The
HTTPS Proxy
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.
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
Self-Defined Time Zone
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.
Configures to enable or disable weather update on the phone. The
Enable Weather Update
default setting is "Yes". If set to "No", the weather information screen will
not show.
Configures weather city code for the phone to look up the weather
information. The default setting is "Automatic" and the weather
City Code
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
Degree Unit
LCD Contrast
Date Display Format
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Specifies the weather update interval (in minutes). The default value is
15 minutes.
Specifies the degree unit for the weather information to display on the
phone.
Configures the LCD contrast level (from 0 to 20). The default value is 10.
Configures the date display format on the LCD. The following formats
are supported:
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Time Display Format
Disable in-call DTMF Display
yyyy-mm-dd: 2012-07-02
mm-dd-yyyy: 07-02-2012
dd-mm-yyyy: 02-07-2012
Configures the time display in 12-hour or 24-hour format on the LCD.
The default setting is in 12-hour format.
When it's set to "Yes", the DTMF digits entered during the call will not
display. The default setting is "No".
Configures to enable or disable the speaker to ring when headset is
Always Ring Speaker
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".
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.
Headset Key Mode
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
Max Unsaved Log
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Defines the interval (in seconds) to save the call history to phone's flash.
The default value is 300 seconds.
Defines the number of unsaved logs before written to phone's flash. The
default value is 200 entries.
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Headset TX gain
Headset RX gain
Handset TX gain
Configures the transmission gain of the headset. The default value is
0dB.
Configures the receiving gain of the headset. The default value is 0dB.
Configures the transmission gain of the handset. The default value is 0
dB.
SETTINGS/ADVANCED SETTINGS PAGE
Allows users to change the admin password. The password field is
Admin Password
purposely hidden after clicking the Update button for security purpose.
This field is case sensitive with a maximum length of 30 characters.
Confirm Password
Layer 3 QoS
Confirms the admin password field to be the same as above.
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
Assigns the VLAN Tag of the Layer 2 QoS packets. The default value is
Tag
0.
Layer 2 QoS 802.1p Priority
Assigns the priority value of the Layer2 QoS packets. The default value
Value
is 0.
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
Local RTP Port
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.
When set to "Yes", this parameter will force random generation of both
Use Random Port
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).
Specifies how often the phone sends a blank UDP packet to the SIP
Keep-alive Interval
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.
The IP address or Domain name of the STUN server. STUN resolution
STUN Server
results are displayed in the STATUS page of the Web GUI. Only
non-symmetric NAT routers work with STUN.
Firmware Upgrade and
Specifies how firmware upgrading and provisioning request to be sent:
Provisioning
Always Check for New Firmware, Check New Firmware only when F/W
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pre/suffix changes, Always Skip the Firmware Check.
The password for encrypting the XML configuration file using OpenSSL.
XML Config File Password
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
Firmware Server Path
Config Server Path
Allows users to choose the firmware upgrade method: TFTP, HTTP or
HTTPS.
Defines the server path for the firmware server. It could be different from
the configuration server for provisioning.
Defines the server path for provisioning. It could be different from the
firmware server for upgrading.
Enables your ITSP to lock firmware updates. If configured, only the
Firmware File Prefix
firmware with the matching encrypted prefix will be downloaded and
flashed into the phone.
Enables your ITSP to lock firmware updates. If configured, only the
Firmware File Postfix
firmware with the matching encrypted postfix will be downloaded and
flashed into the phone.
Enables your ITSP to lock configuration updates. If configured, only the
Config File Prefix
configuration file with the matching encrypted prefix will be downloaded
and flashed into the phone.
Enables your ITSP to lock configuration updates. If configured, only the
Config File Postfix
configuration file with the matching encrypted postfix will be downloaded
and flashed into the phone.
Allow DHCP Option 43 and
If DHCP option 66 is enabled on the LAN side, the TFTP server can be
Option 66 Override Server
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.
When "Automatic Upgrade" is set to "Yes, check for upgrade every
Day of the Week (0-6)
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).
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TR-069 Username
ACS username for TR-069.
TR-069 Password
ACS password for TR-069.
Periodic Inform Enable
Periodic Inform Interval
Connection Request
Username
Enables periodic inform. If set to "Yes", device will send inform packets
to the ACS. The default setting is "No".
Sets up the periodic inform interval to send the inform packets to the
ACS.
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.
Configures to enable phonebook XML download. Users could select
Phonebook XML Download
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.
Configures the phonebook download interval (in minutes). If it's set to 0,
Phonebook Download Interval
the automatic download will be disabled. The default value is 0. The
valid range is 5 to 720 minutes.
Remove Manually-edited
If set to "Yes", when XML phonebook is downloaded, the entries added
Entries on Download
manually will be automatically removed. The default setting is "Yes".
LDAP Directory: Server
Address
LDAP Directory: Port
Configures the IP address or DNS name of the LDAP server.
Configures the LDAP server port.
Configures the LDAP search base. This is the location in the directory
where the search is requested to begin.
LDAP Directory: Base
Example:
dc=grandstream, dc=com
ou=Boston, dc=grandstream, dc=com
Configures the bind "Username" for querying LDAP servers. Some
LDAP Directory: User Name
LDAP servers allow anonymous binds in which case the setting can be
left blank.
LDAP Directory: Password
LDAP Number Filter
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Configures the bind "Password" for querying LDAP servers. The field
can be left blank if the LDAP server allows anonymous binds.
Configures the filter used for number lookups.
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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.
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;
LDAP Name Filter
(!(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".
Specify the "name" attributes of each record which are returned in the
LDAP search result. This field allows the users to configure multiple
LDAP Name Attributes
space separated name attributes.
Example:
gn
cn sn description
Specifies the "number" attributes of each record which are returned in
the LDAP search result. This field allows the users to configure multiple
LDAP Number Attributes
space separated number attributes.
Example:
telephoneNumber
telephoneNumber Mobile
Configures the entry information to be shown on phone's LCD. Up to 3
LDAP Display Name
fields can be displayed.
Example:
%cn %sn %telephoneNumber
Specifies the maximum number of results to be returned by the LDAP
Max. Hits
server. If set to 0, server will return all search results. The default setting
is 50.
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Search Timeout
Sort Results
LDAP Lookup
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.
Specifies whether the searching result is sorted or not. The default
setting is "No".
Configures to enable LDAP number searching when dialing and
receiving calls.
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
Lookup Display Name
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.
Configures to enable idle screen XML download. Users could select
Idle Screen XML Download
HTTP/HTTPS/TFTP to download the XML idle screen file. The default
setting is "No".
Download Screen XML At
If set to "Yes", the idle screen XML file will be downloaded when the
Bootup
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.
Configures a User ID/extension to dial automatically when the phone is
Offhook Auto Dial
off hook. The phone will use the first account to dial out. The default
setting is "No".
Auto Recover From Abnormal
Syslog Server
Configures whether auto recover or not when the phone is running
abnormal. The default setting is "Yes".
The URL or IP address of the syslog server for the phone to send syslog
to.
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:
Syslog Level
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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);
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Send SIP Log
NTP Server
Allow DHCP Option 42
Override NTP Server
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).
Configures whether the SIP log will be included in the syslog messages
or not. The default setting is "No".
Defines the URL or IP address of the NTP server. The phone may obtain
the date and time from the 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".
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
Public Mode
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.
Configures system ring tone. The default value is North American
System Ring Tone
standard. Users could adjust system ring tone frequencies and
cadences based on local telecom standard.
Configures ring or tone frequencies based on parameters from local
Call Progresses Tones:
Dial Tone
Message Waiting
Ring Back Tone
Call-Waiting Tone
Busy Tone
Reorder Tone
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.
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Up to three cadences are supported.
Disable Call-Waiting
Disable Call-Waiting Tone
Disable Direct IP Calls
Disables the call waiting feature. The default setting is "No".
Disables the call waiting tone when call waiting is on. The default setting
is "No".
Disables Direct IP Call. The default setting is "No".
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
Use Quick IP-Call mode
<=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
Configures the number for the phone to dial as DTMF during the call
transfer key
using TRANSFER button.
Configures the access control for the users to configure from keypad
Menu. There are three different options:
Configuration via Keypad Menu
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.
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".
Enable STAR key Keypad
locking
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
Offhook timeout
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Configures the password to lock/unlock the keypad. The password field
allows number with up to 32 characters.
If configured, when the phone is on hook, it will go off hook after the
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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
Specifies whether to replace # by %23 or not for some special situations.
SIP URI
The default setting is "No".
Disable Telnet
Disables Telnet access. The default setting is "No".
Configures the PC port mode. The default setting is "Enabled". When set
PC Port Mode
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
Download Device
Configuration
Selects display language on the phone.
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
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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.
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Click
to
group
in
select
the
dropdown menu.
Click
to
Click to input number
search
in
and dial from available
phonebook.
lines.
Click to edit this
contact.
Click
to
Click to export
Click to import
Click to call
Click to download
add new
phonebook in
phonebook
this contact
the
contacts.
XML format.
XML file.
from
information in .vcf
phone.
the
contact
format
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.
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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.
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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.
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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.
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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".
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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:
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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.
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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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