
The growth rate and, as a consequence, the increase in the cost of equipment significantly reduces the number of IP phone models that can be used in projects. On the other hand, companies are not ready to abandon the usual quality level of corporate phones.
In response to numerous requests from partners, Escene launches sales of new low-cost IP phones, with the most attractive price in the ES corporate lineup.The compact office phone ES205, retains the quality features of the corporate line (cast tube, separate plastic round buttons, stand) for less money than the 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 power over PoE (optional), and also supports voice transmission in HD quality.
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The device is presented in two models:
Escene ES205-PN with PoE support (power supply via Ethernet) and
Escene ES205-N (without PoE support — it is equipped with the Escence 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- Compact design, body size only 21.3x15.7x3.9 centimeters.
- High quality body materials.
- Large and clear graphic screen.
- High ergonomics.
- Two-position stand.
- 4 context buttons
- Simplicity of setup due to the clear interface.
- Russified web-interface and on-screen menu.
- The ability to fully customize the phone using the screen and buttons, including SIP accounts.
- Ability to adapt the phone to work with SIP-compatible equipment.
- The functionality is more than most of the IP PBX 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 ports (PC / LAN) with VLAN support and the ability to work in the switching or routing mode.
- Easy installation and operation, advanced configuration options (including SIP and DVO functions) via the on-screen menu or via the web interface.
- Supports two simultaneous calls on two independent SIP accounts.
- Full duplex speakerphone, caller ID, call hold, call transfer and call forwarding, as well as other advanced features.
- High definition audio support Voice HD (G.722 codec).
- Built-in VPN client.
- Encryption of SIPS signaling and SRTP traffic media.
- Support for a corporate notebook using the LDAP or XML protocol or a personal notebook.
- Russified on-screen menu and web-interface phone.
- HTTP / TFTP / FTP auto setup, 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).
- DNS SRV support.
- Two SIP accounts with the possibility of registration on two independent SIP servers and the possibility of automatic switching in case of loss of registration.
- Two simultaneous calls to the phone from any of the 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 destination.
- 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 microphone hands-free (Full-duplex).
- Two buttons for selecting line 1 and line 2 with light indication of line status.
- Buttons to adjust the volume of the phone / ring signal.
- 4 multifunction buttons below the screen.
- 5 navigation multifunction buttons (4 navigation buttons and a button for deleting the symbol “C”).
- Redial button.
- Speakerphone button with light indication.
- "Mute microphone" button.
- Connector for RJ9 tube connection.
Additional services (additional features)- Waiting for a second call, a queue (if it supports an IP PBX), call transfer, call forwarding, call hold, call pickup, callback, call recall, auto answer.
- Speed ​​dialing, start recording button for the old-code conversation (if it supports IP PBX).
- Multilateral conference (if supports IP PBX), 3-way conference on the phone.
- Do not disturb (DND).
- Voice mail (if the function is supported by IP PBX).
- Personal note book, corporate note book (LDAP or XML).
Control- Protocols update: FTP / HTTPS / HTTP / TFTP / PnP auto-tuning (PnP auto-provision).
- Configuration: via phone's on-screen menu / web-interface / auto-provision
- SNMP V1 / 2, TR069
- Debugging: telnet / phone screen / web-interface.
Nutrition- Adapter model AD200 (AV 220/110 Volt, output DC 5 Volt / 1 A).
- LAN port Power over Ethernet (802.3af, class 0) for ES205-PN
- Power consumption 1.5 W
Package, appearance and packaging
PackagingThe 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 equipment reduces the cost of the phone.
PackagingPhone kit- Telephone set.
- Handset.
- Handset cord.
- RJ45 patch cord to connect to the network.
- Instruction and warranty card.
- Power supply Escence AD200 (for model ES205-N) .
The delivery package for the
Escene ES205-PN model is
missing the Escence AD200 power
supply unit (5 volts), it must be ordered separately.
PhoneFront panel and hardware buttons
The buttons on the phone are less than on older models, but this does not nearly interfere with its configuration and use. It seems that more buttons are not needed.
Conventionally, there are three blocks of buttons:
1. The multifunction buttons are four soft buttons under the phone screen, each of the multifunction buttons shows the function currently active, for example, “New call”, end a 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.
2. 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 press 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. The “Line 1” and “Line 2” buttons are the 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 is lit in red; if it is flashing, an incoming call has arrived. If the line is green, an active call is on the line; if it is flashing, the call is held on the line.
3. Service buttons - Redial redial button, two buttons for volume control and a speakerphone on button.
Back of phone
On the back of the phone is a standard sticker with the model number, serial number and MAC address. If, for convenience, you need to bring the handset wires or power cables to the top of the phone, you can put them on the phone body, for this, there are two grooves on the panel.
Installation standThe phone stand is very easy to install - you need to align the lower guides of the stand with the slots in the phone case, then leaning on the lower guides to 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 a block of interfaces. For AC power using the power adapter, there is a 5 volt socket on the panel, 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 network and PoE power.
Below in the image is a view of the back panel with the wires connected, I did not begin to remove them. Wires do not interfere with the phone is exactly on the table and does not move during dialing, conversation and other actions.
Rear panel with connected wiresView of the phone on the tableSo the phone looks assembled, high-quality plastic, the screen backlight is missing, 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 screenIt is worth noting a nice screen with good resolution. The phone has a monochrome LCD screen without backlight size 128 * 64, small, but its size is enough to easily read the information from the screen.

This is how the phone screen with the registered line in Russian looks. “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.
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 buttonsSo, 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 3), then “Language” (number 1), using the Up or Down buttons to select “Russian” and Click "OK"
Then press the "C" button until you exit the menu.
Configure 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 192.168.0.149, 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: 192.168.1.14
Setting up additional phone featuresAll these settings are made in the “Menu” -> “Functions” (number 2)
- “Auto Answer” allows you to set up an automatic answer to a call without lifting the handset.
- DND allows you to reject all calls when the subscriber is busy.
- The “hot line” allows you to set automatic dialing of a given number immediately or with a set timeout.
- Call Forwarding allows you to set conditional and unconditional forwarding to specified numbers.
- Call waiting allows you to enable or disable the ability to take a second call during a call.
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:
- The "Confr" soft button allows you to transfer a call, the call transfer is implemented using the contact information in the SIP 302 Moved Temporarily message. This message is currently supported by almost all IP PBXs on the market.
- The “Transfer” soft button - call transfer with consultation and blindly, also uses SIP 302 Moved Temporarily.
- The Hold button (also Pickup) allows you to either put a call on hold during a call or pick up a call. By default, pressing this button triggers the standard 123 combination, it can be reassigned via the web interface in the menu “Advanced settings” -> “Phone settings” -> “Basic” -> “Calls”, “Call pickup code” parameter.
- The “Redial” button allows you to redial the last number.
- The “Speakerphone” button allows you to turn on or off the speakerphone, answer the call with the speakerphone on, or end the call if the conversation is held over the speakerphone.
- “Mute” button - allows you to turn off the microphone during a call. An indication appears on the phone screen.
To access call logs:- Method 1: Press 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 "Call Log" menu item).
- Method 3: pressing the navigation button "Up" - opens all calls, "Down" - see missed calls, button "Redial" - see made 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 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:
- Network settings (interfaces, VLAN, VPN, etc.).
- VoIP settings (SIP records and additional functions for signaling and media traffic).
- Settings for additional phone functions (phonebook settings, 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 menu.The state of the phone . For him there is a separate menu of the same name:

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 TabYou can specify one of three connection methods: DHCP, static IP, or PPPoE.
Advanced Settings TabAlso 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 you can configure 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.
PC port setup.
PC port in routing mode.Menu "Network" -> "Advanced Settings"VPN Settings tab
VPN configurationIf 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 just 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
VLAN configurationIn 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 SettingsThe 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 - 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.
SIP account setupIn 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 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 SIP account settingsHere you can specify additional settings to overcome NAT, enable DNS SRV.
Security tab
Encryption SettingThe phone supports encryption of RTP and SIP signaling traffic using TLS protocol.
Menu "Phone Settings" -> "Basic"Tab "Basic"Here you can configure various functions of the phone, such as the “Hotline” - when you pick up the handset, the preset number is automatically dialed, you can turn on auto-search in the address book during dialing and auto answer the call.
The important setting is DTMF type - by default it is set to RFC2833.
Basic phone settingsCalls tab
Basic phone settings - CallsIn this menu, global functions for the phone are configured.
If you set up the SIP settings here, they will be applied to both lines automatically, except for the settings “Local SIP port” and “RTP port range”, which can be useful for properly configuring the firewall.
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.
Here you can also set codes that will be transmitted when you press the “Pickup” buttons (the value in the “Call Pickup Code” field) and “Voice Mail” (the value in the “Voice Mail Number” field).
Intercept the call in three ways:
- Assigning a function to one of the buttons on the phone.
- Using an explicit dialing combination on the phone keypad.
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.
Tab "Forwarding VOIP calls"
Call ForwardingThe tab is set forwarding: unconditional, if the subscriber did not answer or the line is busy.
Menu "Phone Settings" -> "Advanced Settings"“Sound - Basic” and “Sound - Extended” tabsHere you can set the volume of the telephone, speaker and ringing tone. Also, the volume settings of the microphone tube and speakerphone. 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, you can download your own ringtone.
Media setupLine tab
Customize line button functionsThe 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.
Function Keys TabIn 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.
Setting the function buttonsSoft keysAllows 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.
Customize soft buttonsMenu "Phonebook"
Internal phone bookThe 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.
LDAP Corporate Book SetupIf 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.
Service Settings
Menu "Phone Settings" -> "Basic"
Update and backup your phoneYou 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"
DebuggingTo 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 file c logs.
Also, the phone has the ability to collect network dumps of 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 menuHere you can set a username and password for the administrator and user of the phone, as well as download an SSL certificate.
Setting up connection to IP PBX Asterisk using web-interfaceSuppose 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
Equivalently, when configured in the Free-PBX web interface, using the first line as an example:
FreePBX Setup
Setting up a SIP account for FreePBXTo 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.
Exactly the same thing needs to be done with the second line, for example, city number 78126470011 on the SIP server West Coll. 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
SIP account setup for virtual PBXTo 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.
findings
IP phone Escene ES205 is a worthy representative of the line of phones Escene. This is one of the most compact desktop phones on the market today. Despite the size of only 21.3x15.7x3.9 centimeters, the device has a large number of advantages, such as a pleasant appearance, decent workmanship and ease of use. The ease of setup, stability and high quality sound makes this model worthy of attention.
Key features of the phone include:- The compact size of the device with high ergonomics.
- Support two independent SIP accounts per 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 functions of the software and hardware buttons on the phone.
- PoE support (model ES205-PN ).
- Large and contrast LCD screen which is especially attractive for a phone with such dimensions.
- The ability to configure in addition to network settings, SIP accounts, speed dial buttons and redirects directly from the phone screen.
- Simplicity of tinctures and intuitively understand Russified web interface.
- The phone can be configured using only the on-screen menu.
- Advanced Debugging Features