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

The new model has what some users lacked in the previous model - additional multifunctional programmable buttons and a separate headset jack. This was one of the reasons why Escene added this model to the ES line of compact office IP phones. Almost in any company there are categories of employees (for example, secretaries) who, in addition to the basic functions of the phone, need some more special ones, for example, tracking 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, and the difference in price compared to the Escene ES205 will be minimal.

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

Positive features
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Functionality


Specifications


VoIP


Data transfer


Physical parameters


Additional services (additional features)


Control


Nutrition


Package, appearance and packaging


The phone is delivered 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 equipment reduces the cost of the phone. Brief user manual in Russian, it contains contact phone numbers of distributor technical support. This guide is quite enough to perform basic phone configuration.



Phone kit


The delivery package for the Escene ES206-PN model is missing the Escene AD200 power supply unit (5 volts), it must be ordered separately.





Front panel and hardware buttons



The device occupies a minimum of space in the user's work space 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 sacrificing functionality and ergonomics, except that on the case you can miss a couple of standard buttons, but those that are available can be reprogrammed for any functions. Moreover, in this model, compared with the Escene ES205, added a panel of eight programmable LED buttons that are located above the screen. The screen itself was left large enough to easily read everything that is written on it.

Conventionally, there are four sets of buttons:

The multi-function on-screen buttons are four soft buttons under the phone screen, each of which displays the function currently active, for example, New Call, End Call, Do Not Disturb, Call Transfer, and others.

When navigating the menu, these buttons are also used for navigation, for example, “Back”, “Enter” and others, besides there are two navigation buttons “Up” and “Down” in the block and the multifunctional 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 set up, if necessary to make a call from line 2, you need to click on the line button, then dial the number — the phone will send the call through the 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 arrives in red, the diode of the line to which the call arrives flashes. If the line is busy, the line button is lit in red. If the line is green - an active call is on the line; if it is flashing green, the call is held on the line.

The service buttons are the Redial redial button, two buttons for adjusting the volume, and the on / off button.

Programmable multifunctional buttons - a panel of 8 buttons, in the center of which there 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.

Back of phone



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



The phone stand is very easy to install - you need to align the lower guides of the stand with the slots in the phone, 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 & Connectors



The photo above shows a block of interfaces. 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 network and power supply via PoE.

In the bottom row there are two RJ9 connectors - for connecting the handset and for connecting the headset.
The following image shows the rear panel view with the wires connected, I didn’t remove them. The wires do not interfere, the phone is exactly on the table and does not move after lifting the handset, during dialing, talking and other actions.


So the phone looks complete, a good weighty tube, good quality plastic, bright screen backlighting allows you to read the screen in any light, and the contrast of the text and screen makes it easy to disassemble everything that is displayed on the screen even when the backlight is turned off. When the phone first appears on the table, it feels how compact it is.


Phone screen


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



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

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





Missed calls. Information about missed calls is immediately displayed on the screen. The button of the line to which the unanswered call has come is lit in yellow. The diode lines will be lit, and the display will remain on until the user sees the number of the missed call.



Incoming call. In addition to the beep and on-screen indications, when an incoming call is in progress, the button of the line to which the call has arrived flashes.



State of conversation. During a call, the line button is green. There is a talk time 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, which leave a minimum of settings in the phone menu and a larger number only through the web interface, the Escene developers decided to make available from the phone menu in addition to the standard settings, settings related to SIP accounts. That is, the phone can be fully customized using the on-screen menu.

Such a move is fully justified, in some cases, you can configure the phone faster. In addition, sometimes there may be problems with access to the phone through a web-interface or it may be necessary to explain to an employee remotely how to reconfigure his phone. It will be easier for an unprepared person to use the phone menu, rather than a web interface.

Initial setup using phone buttons

So, we turned on the phone, connected the LAN port to the local network that has access to the IP PBX. Employee's computer through the cable connected to the PC port.

Now we need to include the Russian language in the menu. Press the softkey "Menu", it is located on the left below the screen, a menu will open. Use the Up or Down navigation buttons to navigate the menu, press the corresponding button on the phone’s dialer or a soft button (for example, Enter) to select a menu item, use the Back button to return to the previous item.

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 up and down navigation buttons, select “Russian” and click “Save”
Then press the “C” button to exit the menu.

Now you need to configure the network settings:

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

If you need to use a static IP address, while in the “LAN port” menu, select “Static” and click “OK”. By default, the phone is configured with IP 192.168.0.200, to change the settings of the IP address, mask, gateway and DNS, use the menu buttons and navigation keys, after saving the settings, the phone will reboot. I pay attention 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 operation mode between the PC and LAN ports. In bridge mode, this 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, the PC port is assigned an IP and a mask, the NAT address translation is enabled between the LAN and the PC, you can also enable the DHCP server. Thus, the phone becomes also a router with NAT support.

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

Setting up additional phone features

All these settings are made in the “Menu” -> “Functions” (number 2)


Support for additional services (DVO) and programmable buttons


The phone supports two independent SIP accounts, that is, registration on two different IP PBXs. At simultaneous registration of both lines, by default, the first line will be used. To switch to the second line (it should be configured) and return to the first one, use the “Line 1” and “Line 2” buttons.

I note that the phone supports two simultaneous calls, so to use simultaneous SIP registration on both lines in the SIP account settings for each line, you must set the "Number of lines used by the account" parameter to 1 (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 DVO, they all work correctly:


To access call logs:


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 login and password:

root root



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

The menu is divided into several groups:




Phone status


Menu "Phone Status"



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

Network settings


Menu "Network" -> "LAN port"



Basic Settings Tab

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

Advanced Settings Tab

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

Menu "Network" -> "PC port"

L2 switching is switched on by default between the LAN and PC ports of the telephone - the “Bridge” mode. The phone can switch to L3 routing mode — a NAT address translation will start on the LAN port, an IP address will need to be configured on the PC port, and if necessary, a DHCP server must be enabled in which to set the pool of IP addresses for clients.





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



If you need to connect your phone via a secure VPN channel, this can be done 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 hub at each remote location, you only need to configure the VPN client built into the phone. Further, through the tunnel to register his phone on the IP PBX in the central office.

Secondly, VPN improves 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 the terminal and the difficulty of accessing telecoms operators to configure it, because often the terminal is behind NAT. Using a VPN client solves both of these problems, so this useful feature will become increasingly popular. In the example, using the VPN type L2TP, a connection to the vpn.ucexpert.ru server was created.

VLAN Settings Tab



In a corporate network, it is recommended to isolate the computer network traffic from the voice network traffic, this is most often implemented 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"

Tab "Basic"

In addition to the standard SIP account settings (SIP account) - User name (UserID), Name (AuthID) and password, there is a “Label” field, it allows you to insert an arbitrary line description that will be displayed on the phone screen.



In addition to the primary IP address of the SIP server, you can add an additional IP SIP server. In case of unsuccessful registration 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” should be equal to 1 if you need 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 you apply the settings of the second line, the phone will display a message that there are not enough lines.

Advanced Settings Tab



Here you can specify additional settings to overcome NAT, enable DNS SRV. The phone supports encryption of RTP and SIP signaling traffic using TLS protocol.

Menu "Programmable buttons" -> "Speed ​​dial buttons"

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




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

Menu "Programmable buttons" -> "Line"



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

Menu "Programmable buttons" -> "Function keys"

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



Menu "Programmable buttons" -> "Programmable keys"

Allows you to manage the sets of soft buttons that appear on the phone screen depending on the state of the phone (handset is on, handset is off, connected, talking, 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 the other tabs, you can adjust the timeout for turning off the backlight, locking the keyboard, loading your own ringing tone, microphone volume, handset, speakerphone.

Menu "Phone Settings" -> "Features"

Tab "Forwarding VOIP calls"



The tab is set forwarding: 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 to all three SIP accounts automatically.

In case the call transfer is required to be performed with a special key combination (old code), instead of the standard SIP message 302, this can be specified in the setting “Special code for call transfer”. A useful setting that allows you not to break the connection in the conference, if it left the initiator. You can set call forwarding by condition (busy and “no answer”) and unconditional. In this section, you configure the code that is transmitted when you click the Voice Mail button (the value in the Voice Mail Number field).

The important setting is “Return Code on Failure” and “Return Code at DND”, by default IP PBX returns SIP message 603 Decline, these messages can be changed to others, if necessary for correct interpretation of the reason for the release.

The important setting is DTMF type - by default it is set to RFC2833. You can turn on auto search by address book during dialing, as well as search in the phone book by the first characters or by using the predictive input method T9.

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

The tab “Hotline function” allows you to set up automatic dialing of the specified number immediately or with a specified delay, the “Auto Answer” tab serves to set up and select the mode of automatic answer to a call. Pickup "(the value in the" Call Pickup Code "field), you can intercept the call in three ways:


Menu "Phone Settings" -> "Advanced Settings"



Here, in the “Sound” tab, the RTP protocol settings are set. By default, when calling, the phone claims 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 own 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 phone has a built-in phone book, which is quite advanced. It allows you to store up to 300 contact entries, each of which can store up to 3 phone numbers. Entries can be made via the phone's on-screen menu using the web-based interface. To download or save an already-created phone book in XML format, use the Phone Service menu -> Update over HTTP -> XML Phone Book; here you can save or load the phone book in XML format.



If your company uses an LDAP server, you can connect a phone to it and synchronize corporate contacts. 2 and 3 protocol versions are supported, as well as using the “LDAP Lookup For Incoming Call” and “LDAP Lookup For PreDial / Dial” settings, you can search for the contact name for an incoming and outgoing call. 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 such a list and can no longer reach you.

Also, the phone has the opportunity to automatically dial a phone number from the "Phonebook" menu. Just enter it in the appropriate field, and then by pressing the "Dial" button, the phone will call this number.

Service Settings


Menu "Phone service" -> "Basic"



You can copy configuration files using three different protocols. FTP, TFTP and HTTP - the choice of a particular 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 downloaded firmware of the phone 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:


Also, the phone has the ability to collect network dump packages 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. When finished, click the “Finish” button. To download the received dump, click the "Create backup" button.

Also on the “Automatic Update” tab, you can configure the phone's firmware update according to the TFTP / FTP / HTTP / HTTPS protocols.

Security menu

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

Examples of phone settings


Setting up connection to IP PBX Asterisk using web-interface

Suppose we need to configure two extensions (two SIP accounts). For example, the first entry on the IP PBX Asterisk, the second on the virtual IP PBX:

 IP    Asterisk= 10.10.10.1 UserID=10 password= QOXZuTcZ38qlBsr SIP (Asterisk)= 10.10.10.1

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

 [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 configured in the Free-PBX web interface, using the first line as an example:





To work with Asterisk, just configure the Username (Username) = 10, password (Password) = QOXZuTcZ38qlBsr and SIP Server (SIP Server) = 10.10.10.1. You can add a label (Label) that will be displayed on the screen of the phone, 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 on a 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 must be done with the second line, for example, the city number 78126470011 on the SIP server of the operator WestCall. Let's register it on a virtual PBX with a non-standard SIP port 9966:

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



To specify a non-standard SIP port (other than 5060), you must explicitly specify it in the SIP server line: uc.westcall.net:9966. Next, click the "Apply" button.

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

Account 1: Registered
Account 2: Registered

In order to use the DVO buttons (transfer, hold, conference) no additional configuration is required.

Setting up the BLF

For BLF to work, you need to enable this feature on Asterisk in the Free-PBX configuration files:

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

 notifyringing=yes notifyhold=yes

For more information on setting up BLF for Escene phones, click here .

Setting up the BLF 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, 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 is lit in red, the second line is flashing red — an incoming call has arrived, but 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.

findings


The growth of the exchange rate and, as a result, the increase in the cost of equipment, significantly reduced the number of available models of IP phones, because budgets cannot grow as fast as the rate. The ES206 phone is an improved version of the ES205 model, in which the necessary functions for secretaries and call center operators, as well as all those who use the advanced functions of the phone, the programmable buttons panel, BLF or headset, have been added. Now, to access 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, similar in functionality, phones from other manufacturers.

Escene ES206 is one of the most compact low-cost desktop phones on the market today. Despite the dimensions of only 21.3x15.7x3.9 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 of work, Russian localization and high sound quality make this model worthy of attention and, perhaps, one of the most successful options for equipping workplaces where the balance between quality, price and reliability is to be kept to the maximum.

Key features of the phone include:

Source: https://habr.com/ru/post/271391/

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