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E129 SIP DESKPHONE USER MANUAL

AVAYA

E129 SIP DESKPHONE

Small-Medium Business IP Phone

Summary of content (59 pages)

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AVAYA E129 SIP DESKPHONE Small-Medium Business IP Phone E129 SIP DESKPHONE USER MANUAL
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E129 SIP DESKPHONE User Manual Index GNU GPL INFORMATION ........................................................................... 5 CHANGE LOG ............................................................................................. 6 FIRMWARE VERSION 1.0.5.2 .............................................................................................................. 6 WELCOME .................................................................................................. 7 PRODUCT OVERVIEW .....
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CUSTOMIZED LCD SCREEN & XML ................................................................................................. 25 CONFIGURATION GUIDE ......................................................................... 26 CONFIGURATION VIA KEYPAD ......................................................................................................... 26 CONFIGURATION VIA WEB BROWSER ........................................................................................... 31 DEFINITIONS ..............
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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...........................
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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: FIRMWARE VERSION 1.0.5.
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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. FIRMWARE VERSION 1.0.5.
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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.
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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.
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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.
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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 FIRMWARE VERSION 1.0.5.
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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.
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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. FIRMWARE VERSION 1.0.5.
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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.
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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.
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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.
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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.
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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.
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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).
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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.
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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.
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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 .
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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 Lo
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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.
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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.
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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.
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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.
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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.
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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, repla
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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).
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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".
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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.
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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.
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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. FIRMWARE VERSION 1.0.5.
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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.
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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.
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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.
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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.
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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.