Escene ES205 Compact IP Phone Review / Digital Angel Blog

The increase in the rate and, as a consequence, the increase in the cost of equipment significantly reduces the number of IP phones that can be used in projects. On the other hand, companies are not ready to abandon the usual level of quality of corporate phones.

In response to numerous requests from partners, Escene launches sales of new budget IP phones, with the most attractive price in the corporate ES line.

The compact office phone ES205 has retained the high-quality features of the corporate line (molded handset, separate plastic round buttons, stand) for less money than with older models.
The phone is made in a strict and elegant style, made of high quality materials, has a large clear screen, two independent lines, an intuitive interface in Russian, a large set of functions, two Ethernet ports and PoE power (optional), and also supports voice transmission in HD quality.

The device is presented by two models:  Escene ES205-PN  with PoE support (power over Ethernet) and  Escene ES205-N  (without PoE support - equipped with an Escence 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.
• High ergonomics.
• On-off stand.
• 4 context 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 (for example, Broadworks, MFI RTU, Metaswitch, Alcatel-Lucent) and to office IP PBXs (for example, 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, RTU MFI, 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 without backlight 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.
• 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.
• 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, the button to start recording the conversation by 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.
• 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 ES205-PN
• Power Consumption 1.5 W

Scope of delivery, appearance and packaging

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 the phone is neatly packed, there is nothing superfluous in the box. Obviously, this configuration reduces the cost of the phone.


Packing

Scope of delivery

• Telephone set.
• Handset.
• Handset cord.
• RJ45 patch cord for connecting to a network.
• Instruction and warranty card.
• Power supply Escence AD200 (for model ES205-N) .

In the delivery package for the Escene ES205-PN model  there is  no Escence AD200 power supply (5 volts), it needs to be ordered separately.


Complete set of the phone The

front panel and hardware buttons There are



fewer buttons on the phone than on older models, but it almost does not interfere with its configuration and use. It feels like more buttons are not needed.

Conventionally, there are three blocks of buttons:

1. Multifunction buttons - these are four soft buttons under the phone screen, each of the multifunction buttons displays the function currently active, for example, “New call”, end the call, “Do not disturb”, “Transfer call” other.

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.

2. Line control buttons- The phone has two independent SIP accounts (two SIP lines). By default, outgoing calls are established from line 1, unless 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. The buttons “Line 1” and “Line 2” have a light indication, when a call arrives, the diode of the line to which the call is being received flashes. If the line is busy, the line button lights up in red; if it is blinking, an incoming call has arrived. If the line lights up green - an active call is on the line; if it blinks - the call is held on the line.

3. Service buttons - a redial redial button, two buttons for adjusting the volume and a button for turning on the speakerphone.

Phone



back panel A standard sticker with the model number, serial number and MAC address is on the back of the phone. If, for convenience, you need to bring the handset wires or power wires to the top of the phone, they can be laid on the phone body, for this there are two grooves on the panel.


Installing the stand

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, raise the stand counterclockwise until it clicks. As can be seen from the image, the stand can be fixed in two positions.

Phone Interfaces & Connectors



The first photo shows the interface block. For power from the AC mains using the power adapter, the panel has a 5 Volt socket, a connector for connecting a tube with an RJ9 socket. Two Ethernet interfaces - PC for connecting the phone to a computer and LAN for connecting to a local area network and PoE power.
Below in the image is 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 during dialing, talking and other actions.


Rear panel with connected wires.

View of the phone on the table.

So the phone looks complete, high-quality plastic, there is no screen backlight, but the contrast of the text and the screen is quite bright. When the phone first appears on the table, it feels how compact it is.



Phone screen

It is worth mentioning a nice screen with good resolution. The phone has a monochrome LCD screen without backlight size of 128 * 64, small, but its size is enough to easily read information from the screen.



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



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.



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

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 loading the phone, 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 192.168.0.149 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. Please note 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, 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: 192.168.1.14

Configuring additional functions of the phone

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 through 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 the missed calls, button "Redial" - to see the completed calls.

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 192.168.1.36.

Default username and password: root / root



We get to the main menu of the phone’s web configurator. 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.)

The choice of the Russian language and the type of menu.

The state of the phone . There is a separate menu of the same name for it: The



menu refers to the service settings, it allows you to get detailed information about the status of the settings and the phone’s statistics, 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”:



“Basic settings” tab.

You can specify 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).
Also here you can configure Paging - group alert.

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.


PC port setup.


PC port in routing mode.

Menu "Network" -> "Advanced Settings"

tab "VPN Settings" VPN


Settings

If you need to connect your phone through a secure VPN channel, you can do this directly from your 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 VLAN


settings

In the corporate network, it is recommended to isolate the computer network traffic from the 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”

tab “Basic”
In addition to the standard settings for the SIP account - Username (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.


SIP account setup

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 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. 

Calls tab


Additional settings for the SIP account

Here you can set additional settings to overcome NAT, enable DNS SRV.

Security Tab


Encryption Setup

The phone supports RTP and SIP signaling traffic over TLS.

Menu “Phone Settings” -> “Basic”

Tab “Basic”

Here you can configure various functions of the phone, such as “Hot Line” - when you pick up the handset, the specified number is automatically dialed, you can enable auto-search by address book while dialing and auto answer the call.

An important DTMF type setting is the default setting in RFC2833.


Basic phone settings

“Calls” tab


Basic phone settings - Calls

In this menu global functions for the phone are configured.

If you register the SIP settings here, they will be applied for both lines automatically, except for the settings “Local SIP Port” and “Range of RTP Ports”, which can be useful for the correct configuration of the firewall.

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. 

Here you can configure the codes that will be transmitted when the “Pickup” buttons (value in the “Call Pickup Code” field) and “Voicemail” (value in the “Voicemail Number” field) are pressed. 

There are three ways to intercept a call:

• Assigning a function to one of the phone buttons.
• By explicitly dialing a combination on the telephone keypad.

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.

Tab “VOIP Call


Forwarding ” Call Forwarding

The tab sets call forwarding: unconditional if the subscriber did not answer or the line is busy.

Menu "Phone Settings" -> "Advanced Settings"

Tabs "Sound - Basic" and "Sound - Advanced"

Here you can set the volume of the phone, speaker and ringtone. Also, the volume settings of the microphone of the handset and speakerphone. 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, you can download your ringtone.


Media settings

The Line tab.


Setting the functions of the line buttons

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

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


Setting the function buttons

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 (off-hook, off-hook, connecting, talking, etc.) This is a very useful function that allows you to control the functions available to the user.


Configuring Softkeys

Phonebook Menu


Internal 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.


Configure LDAP Corporate Workbook

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.

Service Settings

Menu “Phone Settings” -> “General”


Updating and backing up your phone 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"


Debugging

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, and also download the SSL certificate.

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

Equivalent, when setting up in the Free-PBX web interface, using the first line as an example:


Setting up FreePBX


Setting up a SIP account for FreePBX

To work with Asterisk, just 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 time of re-registration from the 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  .

Exactly the same thing must be done with the second line, for example, city number 78126470011 on the West Call SIP server. 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

Configuring a SIP account for virtual PBX

To specify a non-standard SIP port (other than 5060), you must 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.

conclusions

IP-phone Escene ES205 is a worthy representative of the line of Escene phones. This is one of the most compact desk phones currently on the market. 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 nice appearance, decent workmanship and ease of use. Ease of setup, stability and high sound quality makes this model worthy of attention.

The key features of the phone include:

• Compact dimensions with high ergonomics.
• Support for two independent SIP accounts on the phone.
• 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.
• PoE support ( ES205-PN model ).
• Large and contrasting LCD screen, which is especially attractive for a phone with such dimensions.
• The ability to configure in addition to network parameters, SIP accounts, speed dial buttons and call forwarding directly from the phone screen.
• Simplicity of tinctures and an intuitive Russified web interface.
• The phone can be configured using only the on-screen menu.
• Advanced Debugging Features