November 23, 2015 at 14:24

Overview of the compact Escene ES206 phone with advanced features

    The Escene ES206 is an improved version of the Escene ES205, which we wrote about in a previous review. The presence of a speed dial panel makes this model a good alternative to the familiar Escene US102YN . This model is part of the line of corporate ES phones, which means it retains its features - a molded handset, separate plastic round buttons, a stand, two Ethernet ports, optional PoE (Power over Ethernet) support and voice support in HD quality, but all for less money than older models. The compact office phone ES206 is made in strict corporate style lines, all elements of the phone use high-quality plastic, a large clear LCD screen and two independent lines.

    The new model has something that some users lacked in the previous model - additional multifunctional programmable buttons and a separate headset jack. This was one of the reasons Escene added this model to the ES line of compact office IP phones. Almost any company has categories of employees (for example, secretaries) who, in addition to the basic functions of the phone, need a few more special ones, for example, monitoring the status of BLF lines or speed dialing. For convenient access and control of such functions, the Escene ES206 phone has a built-in panel of 8 LED LED buttons, while the price difference compared to the Escene ES205 will be minimal.

    The phone is available in two models: Escene ES206-PN  with PoE support (power over Ethernet) and  Escene ES206-N  (without PoE support - comes with an Escene AD200 power supply). For the PoE model, the power supply is not included, but if necessary, it can be purchased separately. 

    Positive features

    • Compact design, housing dimensions of only 21.3 x 15.7 x 3.9 centimeters.
    • High quality body materials.
    • Large and clear graphic screen with backlight.
    • Headset jack
    • High ergonomics.
    • On-off stand.
    • 5 programmable buttons
    • Panel of 8 LED programmable buttons
    • Easy setup through a clear interface.
    • Russified web-interface and on-screen menu.
    • The ability to fully customize the phone using the screen and buttons, including SIP accounts.
    • The ability to adapt the phone to work with SIP-compatible equipment.
    • Functionality is greater than most IP-PBXs and telecom operators currently support.

    Functionality

    • Direct SIP connection to Virtual IP PBXs (e.g. Broadworks, Switchray (MFI) RTU, Metaswitch, Alcatel-Lucent) and to office IP PBXs (e.g. Asterisk, 3CX IP PBX, Avaya IP Office, Huawei).
    • Two Ethernet (PC / LAN) ports with VLAN support and the ability to work in switching or routing mode.
    • Easy installation and operation, the possibility of advanced settings (including SIP and DVO functions) through the on-screen menu or via the web interface.
    • Support for two simultaneous calls on two independent SIP accounts.
    • Full duplex speakerphone, caller ID, call hold, call transfer and call forwarding, as well as other additional functions.
    • Support for high definition audio Voice HD (G.722 codec).
    • Built-in VPN client.
    • Encryption of signal SIPS and SRTP media traffic.
    • Support for corporate notebook using LDAP or XML or personal notebook.
    • Russified OSD and web-based phone.
    • Auto Tuning over HTTP / TFTP / FTP, TR069

    Specifications


    VoIP

    • RFC 3261 standard SIP server, Asterisk, Avaya, Cisco, Broadsoft, Switchray RTU, 3CX IP PBX, Panasonic SIP-PBX, Huawei, Metaswitch, Alcatel-Lucent, Yeastar and others.
    • Encryption of SIPS signaling traffic and SRTP media traffic.
    • Audio codecs: G.711 u / a, G.722 (HD Voice), G.729a, G.723.
    • DTMF: In-Band, RFC2833, SIP Info, Auto
    • QoS: TOS, Jiffer Buffer, VAD, CNG, G.168 (32ms).
    • Support for DNS SRV.
    • Two SIP accounts with the ability to register on two independent SIP servers and the ability to automatically switch in case of loss of registration.
    • Two simultaneous phone calls from either of two SIP accounts.

    Data transfer

    • 2 * RJ45 10 / 100M Ethernet Interfaces (LAN / PC)
    • Modes Bridge / Router PC Port
    • Support VLAN QoS (802.1pq) / QoS.
    • IP Addressing: DHCP client or static IP assignment.
    • NAT Traversal: STUN mode
    • Built-in VPN client L2TP or OpenVPN (SSL VPN).
    • Network protocols HTTP, BOOTP, FTP, TFTP, IEEE 802.1Q, IEEE 802.1X.

    Physical parameters

    • Monochrome LCD screen with backlight and a size of 128 * 64 characters.
    • Line status indicator (two-color LED).
    • Full-duplex speaker and speakerphone (Full-duplex).
    • Two buttons for selecting line 1 and line 2 with a light indication of the status of the line.
    • Built-in panel of 8 programmable multi-function buttons.
    • Buttons for adjusting the volume of the telephone / ringtone.
    • 4 multi-function buttons under the screen.
    • 5 navigation multi-function buttons (4 navigation buttons and a button to delete the “C” symbol).
    • Redial button.
    • Handsfree button with light indication.
    • Mute button.
    • RJ9 headset jack.
    • Connector for connecting the RJ9 handset.

    Additional types of service (additional functions)

    • Waiting for a second call, queue (if it supports IP PBX), transferring a call, transferring a call, holding a call, intercepting a call, returning a call, retrying a call, answering automatically.
    • Speed ​​dialing, button to start recording a conversation using the old code (if it supports IP PBX).
    • Multilateral conference (if supports IP PBX), 3-way conference on the phone.
    • Do Not Disturb (DND).
    • Voicemail (if the function is supported by IP PBX).
    • Personal notebook, corporate notebook (LDAP or XML).

    Control

    • Protocol update: FTP / HTTPS / HTTP / TFTP / PnP auto-provision (PnP auto-provision).
    • Configuration: via the on-screen menu of the phone / web-interface / auto-provision (auto-provision)
    • SNMP V1 / 2, TR069
    • Debugging: telnet / phone screen / web-interface.

    Nutrition

    • Adapter model AD200 (AV 220/110 Volts, output DC 5 Volts / 1 A).
    • LAN port Power Over Ethernet (802.3af, class 0) for ES206-PN
    • Power Consumption 1.5 W

    Scope of delivery, appearance and packaging


    The phone comes in a cardboard box, on the side of the package there is a sticker with the model number and barcode of the device. Inside, each element is neatly packed in an individual soft film, there is nothing superfluous in the box. Obviously, this configuration reduces the cost of the phone. The brief user manual is in Russian, it contains the contact numbers of the technical support of the distributor. This guide is enough to perform basic phone setup.



    Phone Package Contents

    • Telephone set
    • Phone stand
    • Handset
    • Handset cord
    • RJ45 patch cord for connecting to a network
    • Instruction and warranty card
    • Power supply Escene AD200 (for model ES206-N)

    The bundled model  Escene ES206-PN  there is no power supply Escene AD200 (at 5 volts), it must be ordered separately.





    Front panel and hardware buttons



    The device takes up a minimum of space in the user's workspace and makes its presence almost imperceptible, but the phone will always be at hand at the right time. Moreover, the small dimensions of the IP phone were achieved without compromising on functionality and ergonomics, except that on the case you can miss a couple of standard buttons, but those that can be reprogrammed for any functions. Moreover, in this model, in comparison with the Escene ES205, a panel of eight programmable LED buttons is located above the screen. The screen itself was left large enough to easily read everything that is written on it.

    Conventionally, there are four button blocks:

    Multi-function on-screen buttons- these are four soft buttons under the phone’s screen, each of which displays the function that is currently active, for example, “New call”, “End call”, “Do not disturb”, “Transfer call” and others.

    When navigating the menu, these buttons are also used for navigation, for example, “Back”, “Enter” and others, in addition, the block has two navigation buttons “Up” and “Down” and a multi-function button for deleting the “C” symbol.

    Line control buttons- The phone has two independent SIP accounts (two SIP lines). By default, outgoing calls are established from line 1, unless of course it is configured, if necessary make a call from line 2, you need to press the line button, then dial the number - the phone will send the call through a second SIP account. The phone can accept two simultaneous calls. On the “Line 1” and “Line 2” buttons there is a light indication, when a call comes in, the diode of the line to which the call arrives flashes in red. If the line is busy, the line button lights up in red. If the line lights up green - an active call is on the line, if it blinks green - the call is held on the line.

    Service buttons - Redial redial button, two buttons for adjusting the volume, and a speakerphone enable button.

    Programmable multi-function buttons - a panel of 8 buttons, in the center of which is a transparent pocket with an insert card. On it you can sign the function of each of the 8 buttons. The functions of the buttons can be different, for example, line status indication (BLF), speed dialing, paging, parking, interception and others.

    Phone



    back panel A standard sticker with the model number, serial number and MAC address is on the back of the phone. If it is necessary to bring the wires of the handset or power wires to the bottom of the phone, then they can be laid along the body of the device, for this there are two grooves on the panel.



    The phone stand is very easy to install - you need to combine the lower guides of the stand with the grooves in the phone body, then leaning on the lower guides to raise the stand counterclockwise until it clicks. As can be seen from the figure above, the stand can be fixed in two positions.

    Phone Interfaces and Connectors



    The photo above shows the interface block. For AC power, the panel has a socket with a constant input voltage of 5 volts for connecting a network adapter. Two Ethernet interfaces - PC for connecting the phone to a computer and LAN for connecting to a local area network and PoE power supply.

    In the bottom row are two RJ9 connectors - for connecting a handset and for connecting a headset.
    The following image shows a view of the rear panel with the wires connected, I did not begin to remove them. The wires do not interfere, the phone stands flat on the table and does not move after picking up the phone, during dialing, talking and other actions.


    So the phone looks like an assembly, a solid weighty handset, good quality plastic, bright backlight allows you to read the screen in any light, and the contrast of the text and the screen makes it easy to make out everything that is displayed on the screen even when the backlight is off. When the phone first appears on the table, it feels how compact it is.


    Phone screen


    It should be noted separately a nice screen with good resolution. The phone has a monochrome LCD screen with a backlight size of 128 * 64, small, but its size is enough to easily read information from the screen.



    This is how the phone screen looks in Russian with a registered line. “Line1” and “Line 2” - an arbitrary label, which is configured in the menu “SIP Accounts” and is called “Label”. 

    Enter and dial. When entering a number, the numbers on the screen are large and read well. When you enter the first digits of the number, the phone shows the most similar numbers by the dialed mask. When dialing a number, the line button through which the number is dialed lights up in red.





    Missed Calls. Information about missed calls is immediately displayed on the screen. The line button to which the call has gone unanswered is lit in yellow. The line diode will light up and the display will remain on until the user views the number of the missed call.



    Incoming call. In addition to the sound signal and indication on the screen, when an incoming call, the button of the line to which the call has arrived flashes.



    The state of the conversation. During a call, the line button lights up in green. There is a talk timer on the screen.



    Call logs.



    View of the menu on the phone screen.



    Phone setup


    The phone can be configured either using the on-screen menu, or using the web-based interface. Unlike most phones from other manufacturers that leave a minimum of settings in the phone’s menu and a larger number of them only through the web interface, Escene developers decided to make the settings related to SIP accounts available from the phone’s menu in addition to the standard settings. That is, the phone can be fully configured using the on-screen menu.

    This step is justified, in some cases, you can set up your phone faster. In addition, sometimes there may be problems with accessing the phone via the web interface or it may be necessary to remotely explain to the employee how to reconfigure his phone. It will be easier for an untrained person to use the phone menu than the web-based interface. 

    Initial setup using the phone buttons

    So, we turned on the phone, connected the LAN port to a local network that has access to the IP PBX. The employee’s computer was connected via cable to the PC port.

    Now we need to include Russian in the menu. Press the softkey “Menu”, it is located on the left under the screen, the menu will open. To move through the menu, use the Up or Down navigation buttons, to select a menu item, press the corresponding button on the phone’s dialer or softkey (for example, “Enter”) to return to the previous item, use the “Back” button.

    Next, press the number 3, which corresponds to the choice of the “System Settings” menu, then select “Phone Setting” (number 1), then “Language” (number 1), using the navigation buttons “Up” or “Down” select “Russian” and press “Save”
    Then press the “C” button to exit the menu.

    Now you need to configure the network settings:

    Press “Menu”, then select the “System Settings” menu (or press number 3), number 2 - “Advanced Settings”, the password is empty by default, just click “OK”. If you need to configure VLAN (menu item 2 - "Network", 3 - VLAN), go to the corresponding menu and configure its ID and priority. Next, select “Network”, then “LAN port”, by default, after the phone boots up, a DHCP client is turned on, which is trying to get an IP address, so there must be a DHCP server in the network where the IP phone is located. If all the settings are made correctly, the phone will receive an IP address and will be ready for further configuration. 

    If you need to use a static IP address, in the “LAN port” menu, select “Static” and click “OK”. By default, IP 192.168.0.200 is configured on the phone, to change the IP address, mask, gateway and DNS settings, use the menu buttons and navigation keys, after saving the settings, the phone will reboot. I draw your attention to the fact that in this menu “LAN port” you can configure the port for access to the web interface, by default it is 80, and also the port for access to the phone via telnet. 

    The PC port setting deserves special attention (Menu -> Settings -> Advanced Settings -> Network -> PC Port). Here you can configure the network mode between the PC and LAN ports. In bridge mode, it is a two-port switch with support for a separate tagged or untagged VLAN for a LAN or PC port. If you set the “Router” mode, then the IP port and mask are assigned to the PC port, NAT address translation is enabled between the LAN and PC, you can also enable the DHCP server. Thus, the phone also becomes a NAT-enabled router.

    Now you need to check the correctness of the network settings and see the IP address that was assigned by DHCP, for this, press "Menu", press the number 1 - "View Status", using the navigation buttons "Up" or "Down", find the IP assigned to the phone, in in my case, the IP address assigned by DHCP: 192.168.1.41

    Configuring additional telephone functions

    All these settings are performed in the "Menu" -> "Functions" (number 2) 

    • “Auto Answer” allows you to set up an automatic answer to a call without picking up the handset.
    • “DND” allows you to reject all calls in case of busy subscriber.
    • “Hot line” allows you to set the automatic dialing of a given number immediately or with a set timeout.
    • “Call Forwarding" allows you to set conditional and unconditional call forwarding to the specified numbers.
    • “Call Waiting” allows you to enable or disable the ability to receive a second call during a call.

    Support for additional services (TWO) and programmable buttons


    The phone supports two independent SIP accounts, that is, registration on two different IP PBXs. When registering both lines at the same time, by default, the first line will be used. To switch to the second line (it must be configured) and return to the first, use the “Line 1” and “Line 2” buttons. 

    Please note that the phone supports two simultaneous calls, therefore, to use simultaneous SIP registration on both lines in the settings of SIP accounts for each line, you need to set the "Number of lines used by the account" parameter to 1 (the default value is 2). That is, the device supports only two lines, you can distribute them at your discretion: either assign both lines to the first SIP account, or distribute one line to each SIP account and register both at the same time. 

    As for the Far East Region, they all work correctly:

    • The ConFr softkey allows you to transfer a call; call transfer is implemented using the contact information in the SIP 302 Moved Temporarily message. Almost all IP PBXs on the market today support this message.
    • The "Transfer" softkey - transferring a call with consultation and blind, also uses SIP 302 Moved Temporarily.
    • The “Hold” softkey (also Pickup) allows you to either put a call on hold during a call or intercept a call. By default, when you click on this button, the standard combination 123 works, it can be reassigned via the web interface in the menu "Advanced Settings" -> "Phone Settings" -> "Basic" -> "Calls", the parameter "Call Pickup Code".
    • The “Redial” button allows you to redial the last number.
    • The “Speakerphone” button allows you to turn the speakerphone on or off, answer a call with the speakerphone turned on, or end the call if the conversation has taken place through the speakerphone.
    • Button “Mute” - allows you to mute the microphone during a call. The indication appears on the phone screen.

    To access call logs:

    • Method 1: click the "Journal" button. The call log contains records of recent outgoing, incoming, and missed calls.
    • Method 2: press the "Menu" button, then the number 5 (corresponds to the menu item "Call Log").
    • Method 3: pressing the navigation button "Up" - will open all calls, "Down" - to see missed calls, the "Redial" button - to see completed calls, and double-clicking to call the last number dialed.

    Phone web interface


    To access the web interface from a computer that has access to the network where the phone is located, enter the IP address of the phone in the address bar of the web browser, in my case it is 192.168.1.41.

    Default username and password:

    root
root



    We get to the main menu of the web-configurator of the phone. For convenience, we immediately select the Russian language in the lower left menu.

    The menu is divided into several groups:

    • Network settings (interfaces, VLAN, VPN, etc.).
    • VoIP settings (SIP entries and additional functions for signaling and media traffic).
    • Settings for additional phone functions (settings for the phone book, programmable buttons, dial plan, sound, etc.).
    • Service settings (logging, reset and reboot, configuration management and software updates, etc.)



    Phone status


    Menu “Phone Status” The



    menu refers to the service settings, it allows you to get detailed information about the status of the settings and statistics of the phone, such as the time in operation, the status and condition of the network connection, the status of registration of SIP lines, firmware version and others.

    Network settings


    Menu "Network" -> "LAN port"



    tab "Basic settings"

    You can set one of three connection methods: via DHCP, static IP or PPPoE.

    Advanced Settings Tab

    Also Important settings are HTTP and Telnet ports. They should be made non-standard if the phone is in an untrusted network (for example, with an external IP address on the Internet). Paging is also configured here - group notification.

    Menu "Network" -> "PC port"

    Between the LAN and PC ports of the phone, L2 switching is switched on by default - "Bridge" mode. The phone can switch to L3 routing mode - the NAT address translation will turn on on the LAN port, the IP address will need to be configured on the PC port, and if necessary, enable the DHCP server in which to register the pool of IP addresses for clients.





    Menu "Network" -> "Advanced Settings". “VPN Settings” tab



    If you need to connect the phone through a secure VPN channel, you can do this directly from the phone, without buying additional equipment (for example, a VPN router), the phone supports L2TP and OpenVPN (SSL VPN). This is a very useful feature for several reasons. 

    Firstly, if you have several phones that need to be delivered to a remote office, there is no need to buy a VPN concentrator at each remote point, just configure the VPN client built into the phone. Next, through the tunnel register his phone on the IP PBX at the central office. 

    Secondly, VPN increases security, more and more administrators are thinking about how to protect terminals that are on the Internet, two problems are becoming more acute: the danger of hacking a terminal and the difficulty of getting access to engineers for a telecom operator to configure it, because often the terminal is behind NAT. Using a VPN client solves both of these problems, so such a useful feature will become more and more popular. In the example, using the L2TP VPN type, a connection to the vpn.ucexpert.ru server is created.

    VLAN Settings tab



    In a corporate network, it is recommended to isolate computer network traffic from voice network traffic, this is most often done using two VLANs. The phone supports VLAN on both ports. 

    VoIP Settings

    The phone allows you to manage a large number of SIP signaling settings and settings for RTP media traffic.

    Menu “SIP Accounts” -> “Account 1”

    Basic Tab

    In addition to the standard settings for the SIP account (SIP account) - Username (UserID), Name (AuthID) and password, there is a “Label” field, it allows you to insert any Description of the line that will be displayed on the phone screen.



    In addition to the main IP address of the SIP server, you can add an additional IP SIP server. In case of registration failure during the timeout, which by default is 32 seconds, the address of the additional SIP server will be used for registration. The setting "Number of lines used by the account" must be equal to 1 if you want to use both lines, because the second line must be assigned to the second SIP account. If you leave the value equal to 2, then when applying the settings of the second line, the phone will display a message that there are not enough lines. 

    Advanced Settings Tab



    Here you can set additional settings to overcome NAT, enable DNS SRV. The phone supports RTP and SIP signaling traffic over TLS.

    Programmable Buttons Menu -> Speed ​​Dial Buttons

    In this menu, you can configure the operation mode of each of the 8 programmable multi-function buttons, then the functions of each of the buttons can be signed using the insert sheet. Its approximate dimensions are: length 7 cm, width 1.5 cm. The following modes are available:

    • Asterisk BLF - Busy Lamp Field allows you to monitor the current status of the lines of other subscribers in real time.
    • Broadsoft BLF is the same as the first item but with features for working with the Broadsoft platform.
    • Speed ​​dialing - Allows you to dial a saved number with one touch.
    • Speed ​​dial prefix - Allows you to dial a combination of numbers and then wait for the end of dialing from the subscriber.
    • DTMF - allows you to send a saved DTMF combination
    • SIP URI – позволяет набрать сохраненный ранее адрес, например sip:ignasio@ucexpert.ru



    An example of setting the panel buttons in BLF mode for Asterisk is given at the end of the review.

    The Programmable Buttons menu -> Line



    The Account selection menu can take the values ​​Account1 / Account2 / Any and becomes active if a button is assigned a dialing mode, for example, speed dialing, DTMF or a speed dial prefix.

    The "Programmable Buttons" menu -> "Function keys"

    In this menu, you can assign an action to the phone's function buttons if the actions they perform by default for some reason should be different. To do this, in the drop-down menu, select the action that will be performed when you click on the button. 



    Programmable Buttons Menu -> Programmable Keys

    Allows you to control the sets of soft buttons that appear on the phone’s screen depending on the state of the phone (the handset is off, the handset is off, the connection, conversation, etc.) This is a very useful feature that allows you to control the functions available to the user.



    Menu “Phone Settings” -> “Basic”

    Here, in the “Time Settings” tab , there are settings for time synchronization, you can select the synchronization source: SIP server / SNTP / Manual, time zone and date format. On other tabs, a timer for turning off the screen backlight, locking the keyboard, loading your ringtone, phone volume, microphone, handset, speakerphone is configured.

    Menu "Phone Settings" -> "Features"

    Tab "Call Forwarding VOIP"



    Forwarding is set on the tab: unconditional if the subscriber did not answer or the line is busy.

    Other Features Settings tab



    In this menu, you can configure various phone functions related to call processing and SIP signaling.

    If you register the SIP settings here, they will be applied automatically for all three SIP accounts.

    If you want to transfer a call using a special combination of buttons (old code), instead of the standard SIP message 302, this can be specified in the “Special code for transferring calls” setting. A useful setting that allows you to keep the connection in the conference if the initiator leaves it. You can set call forwarding by condition (busy and “no answer”) and unconditional. In this section, the code is configured, which is transmitted when the “Voicemail” button is pressed (the value in the “Voicemail Number” field).

    An important setting is “Failure Return Code” and “DND Return Code”; by default, the IP PBX returns a 603 Decline SIP message, these messages can be changed to others if necessary to correctly interpret the reason for the end.

    An important DTMF type setting is the default setting in RFC2833. You can enable auto search by address book while dialing, as well as search the phone book by the first characters or using the predictive input method T9.

    In the menu " Phone Settings" -> "Features" there are several more tabs.

    The “Hot line function” tab   allows you to immediately or automatically set the specified number with the specified delay, the “Auto answer” tab is used to configure and select the mode for automatically answering a call, the code will be configured on the “Call Pickup” tab , which will be transmitted when the “ Pickup ”(value in the“ Call pickup code ”field), there are three ways to intercept a call:

    • By pressing the “Hold” button, the combination to intercept the call assigned in the “Interception code” field will be sent to the IP PBX.
    • Assign a speed dial combination to one of the extension panel buttons to intercept a call.
    • By explicitly dialing a combination on the telephone keypad.

    Menu "Phone Settings" -> "Advanced Settings"



    Here, in the "Sound" tab, the settings of the RTP protocol are set. By default, when calling, the phone declares all possible codecs. If necessary, unused codecs can be disabled. Here you can enable echo cancellation and VAD. Moreover, in the “Call” tab you can download your ringtone. In the “Dial Plan” tab, you can create rules for modifying the dialed number before sending it to the SIP server.

    Menu "Phonebook"



    The device has a built-in phone book, and quite advanced. It allows you to store up to 300 records of contacts, in each of which you can save up to 3 phone numbers. Entries can be made through the on-screen menu of the phone, using the web interface. To download or save a ready-made phonebook in XML format, use the menu “Phone Service” -> “Update via HTTP” -> “XML Phonebook”, here you can save or download a phonebook in XML format.



    If the company uses an LDAP server, you can connect a phone to it and synchronize corporate contacts. 2 and 3 versions of the protocol are supported. Also, using the settings “LDAP Lookup For Incoming Call” and “LDAP Lookup For PreDial / Dial”, you can apply for a contact name for incoming and outgoing calls. If the contact is in the LDAP directory, then its name will be automatically added to the number.

    The phone also supports blacklists or ban lists: an unwanted phone number is added to this list and will no longer be able to reach you.

    Also, the phone has the opportunity to automatically dial the phone number from the "Phone Book" menu. Just enter it in the appropriate field, and then by clicking on the “Dial” button, the phone will call this number.

    Service Settings


    Menu “Phone Service” -> “General” You can



    copy configuration files using three different protocols. FTP, TFTP and HTTP - the choice of a protocol is a matter of taste and convenience. Software update is extremely simple, you need to select the firmware file, then click update. If the version of the phone’s downloadable firmware is lower than the installed one, a window will appear with the inscription “Filename is illegal”. In the menu, you can also restart the phone or reset it to the factory settings.

    Menu "Phone Settings" -> "Advanced Settings"



    To debug the phone, you can enable logging by specifying the necessary logs. You can view them in two ways:

    • In the same menu, enable sending logs to the syslog server.
    • Download the log file.

    Also, the phone has the ability to collect network dump packets into pcap files, which can then be analyzed using a sniffer, for example, Wireshark, this is an extremely effective debugging tool.

    To start capturing packets, click the "Start" button, after finishing, click the "Finish" button. To download the resulting dump, click the "Create backup" button.

    Also, on the “Automatic update” tab, you can configure the phone to update the firmware on a schedule using the TFTP / FTP / HTTP / HTTPS protocols.

    Security menu

    Here you can set the login and password for the administrator and user of the phone, as well as download the SSL certificate.

    Phone settings examples


    Configuring Asterisk IP PBX Connection Using the Web Interface

    Suppose we need to configure two extension numbers (two SIP accounts). For example, the first record on an Asterisk IP PBX, the second on a virtual IP PBX:

    IP адрес сервера с Asterisk= 10.10.10.1
UserID=10
password= QOXZuTcZ38qlBsr
SIP сервер(Asterisk)= 10.10.10.1

    In Asterisk sip.conf configuration, this will be equivalent:

    [10]
deny=0.0.0.0/0.0.0.0
secret= QOXZuTcZ38qlBsr
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=01
pickupgroup=01
allow=g722
dial=SIP/10
mailbox=10@device
permit=0.0.0.0/0.0.0.0
callerid=device <10>
callcounter=yes
faxdetect=no

    Similarly, when configuring in the Free-PBX web interface, using the first line as an example:





    To work with Asterisk, it’s enough to set Username = 10, password = QOXZuTcZ38qlBsr and SIP server (SIP Server) = 10.10.10.1. You can add a label (Label) that will be displayed on the phone screen, in this case, "L1 # 10".

    You can reduce the re-registration time from standard 3600 seconds to 600 seconds, this is especially true if the IP PBX is located outside the office, for example, Virtual PBX. If the phone is located on the local network and the IP PBX is on the Internet, no special settings are usually required to overcome NAT. Next, click the “Apply” button  .

    The same thing needs to be done with the second line, for example, city number 78126470011 on the SIP server of WestCall operator. Register it on a virtual PBX with non-standard SIP port 9966:

    userid=78126470011
authid=6470011
password= eIoMzKsf
sip прокси=uc.westcall.net
port=9966



    To specify a non-standard SIP port (other than 5060), you need to explicitly specify it in the SIP server line: uc.westcall.net:9966. Then click the “Apply” button  .

    In case of successful registration, the corresponding indication will appear on the phone screen, so information about the line registration status is available on the Status menu page  :

    Account 1: Registered 
    Account 2: Registered

    In order to use the DVO buttons (transfer, hold, conference) of additional settings not required.

    Configuring the operation of BLF

    For the operation of BLF, you must enable this feature on Asterisk in the Free-PBX configuration files:

    In /etc/asterisk/sip_general_custom.conf you need to add lines that allow subscribers to monitor the status of the lines: 

    notifyringing=yes
notifyhold=yes

    For more information about setting up BLF for Escene phones, read the link here .

    BLF setup on the phone is very simple, you just need to specify the numbers for which you want to activate the BLF function, in our case these are lines 12 and 13:



    If everything is correct, then the first two buttons on the panel will become active - their status will be displayed - the line is free, on a new call has come to the line or the line is busy - there is a conversation.

    In this example, the first line is busy, it lights up in red, the second line blinks in red - an incoming call has come to it, but it has not been answered yet.



    When you press the line button, the phone will automatically make a call to the number corresponding to this button.

    conclusions


    An increase in the exchange rate and, consequently, an increase in the cost of equipment, significantly reduced the number of available IP-phone models, because budgets cannot grow as fast as the rate. The ES206 phone is an improved version of the ES205 model, which adds the functions necessary for secretaries and call center operators, as well as for all those who use the advanced functions of the phone - a programmable button panel, BLF or a headset. Now for access to such functions there is no need to buy more expensive models, because the characteristics of the Escene ES206 will be enough. Using this model in some cases will help replace more expensive phones of other manufacturers with similar functionality.

    The Escene ES206 is one of the most compact, inexpensive desktop phones on the market right now. Despite the dimensions of only 21.3 x 15.7 x 3.9 centimeters, the device has a large number of advantages, such as a pleasant appearance, decent workmanship, ease of use, additional programmable buttons and a headset jack. The ease of setup, stability, Russian localization and high sound quality make this model worthy of attention and, perhaps, one of the most successful options for equipping workstations where it is necessary to maintain the maximum balance between quality, price and reliability.

    The key features of the phone include:

    • Compact dimensions with high ergonomics.
    • Support for two independent SIP accounts on the phone.
    • Built-in programmable panel for 8 LED buttons.
    • Separate headset jack.
    • The presence of an additional Ethernet port for connecting to a computer and the ability to work in IP routing mode.
    • 5 programmable buttons.
    • The ability to reconfigure the software and hardware buttons of the phone.
    • Support PoE (Model ES206-PN ).
    • Large and contrasting LCD screen with backlight, which is especially attractive for a phone with such dimensions.
    • The ability to configure network settings, SIP accounts, speed dial buttons and call forwarding directly from the phone screen.
    • Easy settings and intuitive Russified web interface.
    • Advanced debugging features.
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