Escene ES620 IP Phone Review with Escene ESM 32 Expansion Module
Today we’ll look at the Escene ES620 IP Phone and ESM 32 Extension Console. By its characteristics and functional features, this IP-phone is the oldest model in the line of corporate phones.
The main purpose of the phone is when the manager needs to put a large, solid phone with a huge color screen on the table. If you need: broadband voice communication (HD Voice), bright color display, 8 independent lines, an expanded set of functions for corporate communications and an interface in Russian - the 620 series will meet these requirements.
A large number of lines allows you to effectively control all communications, and the display transforms the workplace (for example, the flag of your favorite football team).
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Escene is positioning this model as a representative-level IP phone, and indeed the size and resolution of the screen, the number of lines, the number of additional functions and buttons make it possible to call the phone the most advanced device in the corporate line
Instead of installing additional phones on the table, you can easily extend the capabilities of your existing Escene 620 Series IP Phone by equipping it with the Escene ESM 32 expansion module. Each module is equipped with 32 physical programmable buttons with color indication of the status, up to 6 expansion modules can be connected to one telephone.
The retail cost of the Escene ES620-PE model today is 9,000 rubles, considering the capabilities of the phone, this is one of the most profitable offers on the market.
Key features of the phone
Functionality
Specifications
VoIP
Data transfer
Physical parameters
Additional services (additional features)
Control
Nutrition
Packaging
The phone is delivered in a cardboard box that shows the company logo, on the side of the package there is a sticker with the model number and barcode of the device.
Phone kit
Inside the phone is neatly packed, there is nothing superfluous in the box. Obviously, this equipment reduces the cost of the phone.
The box contains standard equipment, which includes:
The Escene ES620-PE model does not include the Es300 AD300 power supply unit (12 volts), it must be ordered separately. Also, the phone can be powered via a USB cable.
Front panel and hardware buttons
Conventionally, the buttons on the phone can be divided into 4 blocks.
The first block is the management of phone service functions.
There is a separate voice mail call button with an envelope image, if there are unread messages in the voice mail box, the button lights up in red. A very useful button (the headset icon is shown above it) to switch to the headset and back, the button also has a light indicator, which allows the operator to control whether the headset is on or off. Also on the panel is a separate microphone mute button, if the microphone is turned off, the button is lit in red. The phone has a setting that allows you to automatically turn on the headset for any incoming call.
The second block is the management of additional functions.
Here are all the necessary buttons, just the ones that are used most often:
There are also two buttons for adjusting the phone / ring volume.
The third block - line management and multifunctional screen buttons
The phone has eight independent SIP accounts (eight SIP lines). By default, outgoing calls are established from line 1, if 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 line.
The phone can accept up to eight simultaneous calls. The Line 1-8 buttons have a light indication, when a call arrives, the diode of the line on which the call is 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.
Also, any of the eight line buttons can be reprogrammed and work in BLF mode, speed dial (number, prefix or SIP URI), or DTMF tone transmissions. If we are talking about dialing a prefix, this function differs from dialing that the saved combination is automatically dialed, but the block of digits is not sent from the phone to the IP PBX - the phone is waiting for the dialing of the remaining digits of the number.
The BLF (Busy Lamp Field) function allows you to monitor the current status of other subscriber lines in real time. When the button is in BLF mode, if the button is red, the line is busy, if it is green, the line is free.
The number of programmable buttons on the phone to expand to expand to 192, connecting the required number of expansion modules.
Each of the multi-function buttons below the screen displays the function currently active, for example: New call, end call, Do Not Disturb, Call Transfer and others.
The fourth block - multifunctional navigation keys
The block is used primarily for easy navigation through the menu, the “C” button is used to delete a character. Using the Up and Down buttons, you can adjust the ring volume or the volume of the phone during a call.
The panel has a separate large red button - “Hands-free”, which allows you to turn on or off the speakerphone (speakerphone), it is full-duplex in the phone. When the speakerphone is working, a red indicator lights up on the button.
The back of the phone. Wall mounting.
On the back of the phone is a standard sticker with the model number, serial number and MAC address. The model is equipped with a swivel stand, with which you can choose a convenient angle of the phone, you just need to press the buttons on the sides of the stand and set the desired angle.
For mounting on the wall, it is enough to bend the stand up to the stop, for installation in the stand there are two holes with which you can mount the phone on the wall.
The first photo shows a block of interfaces. For AC power using a power adapter, there is a 12-volt socket on the panel, two headset connectors and a tube with an RJ11 socket.
Two Ethernet interfaces - PC for connecting the phone to a computer and LAN for connecting to a local network and PoE power.
Two jack 3.5 mm are used to connect the headset. This is extremely convenient, since many headsets have such a connector.
USB connector can be used for:
Here is the panel with the connected wires. The picture shows that all the wires are ergonomically connected and do not interfere.
View of the phone on the table
This is how the phone looks assembled, high-quality plastic, the screen of the phone is bright, sharp and clear, it looks very solid.
Expansion module - a console with 32 programmable buttons connected to the phone. Each button can be programmed to monitor the status of any SIP IP PBX line in BLF mode (busy / free / incoming call) or used for speed dialing, DTMF combination, dialing prefix or calling by SIP URI address.
The console is an indispensable tool for improving the performance of employees who work with a large number of calls. The important advantages can also be attributed to the lack of need for an additional power source for the console - all 6 expansion modules receive power from the phone.
As described above, up to 6 external expansion modules can be connected to the phone, we will take a closer look at the console.
The device comes in a compact box.
The delivery package includes an expansion module, a metal bracket with four screws for securely fastening the phone with an expansion module, a patch cord with RJ-45 connectors for connecting the console and the phone, and a warranty card.
On the rear panel are two connectors:
This is the look of the phone and the panel with the connecting bracket installed. To fix it you need to tighten the four screws, it can be seen in the photo.
The user manual says that after connecting the cable to the expansion module, to activate it, you need to restart the phone or open the menu — the Menu button on the telephone panel, then button 2 - “Functions”, then button 9 - “Expansion Module”, then press button from "1" to "6", which corresponds to the number of the expansion module that must be detected.
For example, pressing the "1" button initializes the first expansion module. In the latest version of the phone software, after connecting the console and rebooting the phone, it was determined by itself and no additional actions were required.
Expansion module and telephone on the table. The back stand of the module is fixed in the same way as the phone and therefore they stand exactly and are equally adjustable in height, the wires do not interfere. The design itself turned out to be reliable, there are no backlashes between the phone and the module.
The panel is successfully combined with the design of the phone of the same size and shape, has compact dimensions and conveniently located buttons, the size and location of the buttons makes it easy to accurately "get" into the right one.
It should be noted separately high-quality color screen with good resolution. The phone has a large color LCD screen with a diagonal of 4.3 inches and a resolution of 480x272 pixels, which makes it easy to read information from the screen and display all the necessary information on the screen.
This is how the phone screen with the registered line in Russian looks. “Number 10”, “number 11” and “ext 11” is an arbitrary label that is configured in the “SIP Accounts” menu and is called “Label”. The button with the name “Ignat” is programmed as a speed dial of the internal number, the “add” button is programmed as a DTMF code. international communication.
The example shows the wide possibilities for arbitrary programming of the buttons of the telephone lines.
Dialing a number
When dialing the number, the button of the line through which the number is dialed lights up in red, before sending the entered combination of numbers to the IP PBX, all lines are green, because the line has not been selected yet. By default, when you press the call button, it is sent via the first line, unless otherwise configured. To call through an arbitrary line, you must first click on the button of this line, it lights up in red, then dial the number and wait until the end of the waiting time for a set or immediately press the call button.
Incoming call. The button of the line to which the call came flashes in red.
State of conversation. During a call, the line button is green.
Call Logs
Due to the large screen, the phone shows 8 entries without scrolling.
In the photo below, 16 lines work in the BLF mode, the 6th line is busy - the diode is lit in red. On line 7 incoming call - the diode flashes red. The remaining lines are free - the diodes are lit in green. Starting from the 17th line, the buttons are registered as speed dial, DTMF, dial prefix and SIP URI - in this mode the line diodes are extinguished.
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 "Menu" button or the "OK" button, it is located in the middle of the navigation button block, 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 softkey (for example, “Enter” or “Select”) to select a menu item, use the “C” button to return to the previous item .
Next, press the number 1 (or the “OK” button), which corresponds to the choice of the “Language” menu, using the Up or Down navigation buttons select “Russian” and press the “OK”
Then press the "C" button until you exit the menu.
Now you need to configure network settings
Click "Menu" (or "OK"), then select the "Settings" menu (or press number 6), number 2 - "Advanced settings", by default the 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 activated that tries to get 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, press the number 1 - “Type”, 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 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.
Special attention is given to the setting of the PC port (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, an IP and a mask are assigned to the PC port, 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, press "Menu" (or "OK"), press number 7 - "Status", then select "Network" (or press number 1) , the screen displays the IP assigned to the phone, in my case the IP address assigned by DHCP: 192.168.1.34
All these settings are made in the “Menu” -> “Functions” (number 2)
Support for additional services (DVO) and programmable buttons The
phone supports eight independent SIP accounts, that is, registration with eight different IP PBXs. At simultaneous registration of all 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 first through eighth line buttons.
I note that the phone supports simultaneous calls, so to use simultaneous SIP registration on eight 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 4 dual-channel SIP accounts or 8 single-channel accounts; 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 eight at a time.
As for the DVO - they all work correctly
To access the web interface from a computer with 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 192.168.1.34
192.168.1.34 The
default login and password are:
root
root
There are two on the phone access level: an administrator level that can change any settings and a user who can perform a limited number of settings.
We get to the main menu of the web configurator of the phone. For convenience of configuration, we immediately select the Russian language in the lower left menu: The
menu is divided into several groups:
Network settings (interfaces, VLAN, VPN, etc.).
Consider the most important menu items phone. The “Setup Wizard” menu is used to quickly set up the phone, allows you to configure two tabs in sequence: the “Network” menu -> “LAN port” and the basic SIP account settings in the “SIP accounts” -> “Account1” menu. These tabs will be discussed in more detail below.
Menu “Network” -> “LAN port”
You can set one of three connection methods: via DHCP, static IP or PPPoE. The 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).
Menu “Network” -> “PC port”
Between the LAN ports and the telephone PC, by default L2 switching is switched on - “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 “Advanced Settings” -> “VLAN Settings”
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.
Menu “Advanced Settings” -> “VPN Settings”
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 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.
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”
In addition to the standard SIP account settings - User Name (UserID), Name (AuthID) and password, there is a “Tag” field, it allows you to insert an arbitrary description of the line 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 two lines, because the second line must be assigned to the second 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.
The phone supports encryption of RTP and SIP signaling traffic using TLS protocol.
Menu "Programmable buttons"
In this menu you can customize the mode of operation of each of the 8 buttons of the phone lines.
The following modes are available:
“Sound” menu
By default, during a call, the phone claims all possible codecs. If necessary, unused codecs can be disabled.
In the menu, you can adjust various volume parameters: handset, ringer, microphone, speakerphone. You can enable echo cancellation and VAD. Moreover, you can download your own ringtone.
Menu “Advanced Settings” -> “Global SIP Settings”
If you set the SIP settings here, they will be applied to all 8 lines automatically, except for the “Local SIP Port” and “RTP Port Range” settings, which can be useful for correct settings of the firewall.
Menu "Advanced Settings" -> "Phone Settings"
In this menu, additional functions are configured. Such as the “Hot Line” 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.
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 menu, you can configure 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).
You can intercept a call in three ways:
By pressing the “Hold” button — a combination for call pickup will be sent to the IP PBX, assigned in the Call Pickup Code field.
By assigning a speed dial combination to one of the extension panel buttons for call pickup. Using an explicit dialing combination on the phone keypad.
Menu "Phonebook"
The phone has a built-in phone book, which is quite advanced. It allows you to store up to 800 contact records, 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.
Debug menu
To debug your phone, you can enable logging by selecting the necessary logs (Phone Maintenance menu -> Log). You can view them in two ways:
Menu "Phone maintenance" -> "Auto Provision" (Automatic update)
Using this menu, you can configure automatic downloading of configuration, firmware and address book to your phone. You can download one of several protocols: http / https / ftp / tftp.
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”.
Backup and Software Update
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.
The status of the phone and system software can be viewed in the menu items “Status” and “System Information”. Here you can also find information about registering SIP phone accounts, for example, in our case two SIP accounts are registered - 10 and 11, with a capacity of two simultaneous calls each.
On the phone screen, the first two buttons will be assigned line 10, and the third and fourth line 11. In the lines “Account 1 and 2” the status will be “Registered”.
Since an extension module is connected to the phone, the corresponding status is displayed in the “Expansion Module 1” line: Connected.
Suppose we need to configure two extensions (two SIP accounts). For example, the first entry on the IP PBX Asterisk (with FreePBX) + configure the BLF buttons, the second one on the virtual IP PBX:
IP address of the server with Asterisk = 10.10.10.1
UserID = 10
password = Tc6SAzsD
SIP server (Asterisk) = 10.10.10.1
In the Asterisk sip.conf configuration, this will be equivalent to:
[ten]
deny = 0.0.0.0 / 0.0.0.0
secret = Tc6SAzsD
dtmfmode = rfc2833
canreinvite = no
context = from-internal
host = dynamic
type = friend
nat = yes
port = 5060
qualify = yes
callgroup = 01
pickupgroup = 01
allow = ulaw
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 example of the first line (line 10), numbers from 12 to 15 will be used for BLF:
To work with Asterisk, it is enough to configure the Username (Username) = 10, password (Password) = Tc6SAzsD 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 “Line 1”.
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 "Submit".
Exactly the same must be done with the second line, for example, city number 78126470011 on the SIP server West Call. Let's register it on a virtual PBX with a non-standard SIP port 9966:
userid = 78126470011
authid = 6470011
password = eIoMzKsf
sip proxy = 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.
Further, in the menu “Expansion modules” -> “Expansion module 1” it is necessary to program the buttons of the first panel, all changes are applied without rebooting.
Setting up the BLF buttons for the Escene ESM 32 expansion module is very simple, you just need to specify the numbers for which you need to activate the BLF function, in our example in the picture these are lines from 12 to 17:
If everything is correct, then the first four buttons on the phone’s panel will become active - their status will be displayed, in my case in green, this means that the lines are free.
After successfully setting up the buttons in BLF mode, the buttons of the lines that are currently free will turn green, the buttons programmed in other modes (speed dial, DTMF, etc.) do not have a light indication (in the example button 17-20). In BLF mode, when you press the line button, the phone will automatically make a call to the line corresponding to this button.
Key features of the phone include:
Support for 8 independent lines, large and bright color screen, good quality of parts and housing, connectivity up to 6 modules with 32 buttons without additional power, PoE and power supply via USB cable, two ways to connect the headset makes the phone a very attractive option for representative functions, as well as for handling a large number of challenges, for example, for managers and their secretaries. Summarizing what has been written, we can conclude that the phone is the best product in its class.
The main purpose of the phone is when the manager needs to put a large, solid phone with a huge color screen on the table. If you need: broadband voice communication (HD Voice), bright color display, 8 independent lines, an expanded set of functions for corporate communications and an interface in Russian - the 620 series will meet these requirements.
A large number of lines allows you to effectively control all communications, and the display transforms the workplace (for example, the flag of your favorite football team).
')
Escene is positioning this model as a representative-level IP phone, and indeed the size and resolution of the screen, the number of lines, the number of additional functions and buttons make it possible to call the phone the most advanced device in the corporate line
Instead of installing additional phones on the table, you can easily extend the capabilities of your existing Escene 620 Series IP Phone by equipping it with the Escene ESM 32 expansion module. Each module is equipped with 32 physical programmable buttons with color indication of the status, up to 6 expansion modules can be connected to one telephone.
A series of devices 620, at the moment, is represented by three models: Escene ES620-PE to which this review is devoted, has two Fast Ethernet network interfaces with PoE support (power supply via Ethernet), Escene GS620-PE is equipped with a Gigabit Etherhet network controller, model Escene WS620- E has a wireless Wi-Fi interface for connecting to a network and a Fast Ethernet interface for connecting a computer.
The retail cost of the Escene ES620-PE model today is 9,000 rubles, considering the capabilities of the phone, this is one of the most profitable offers on the market.
Key features of the phone
- Part of a single corporate level lineup.
- High quality body materials.
- Large and clear color screen.
- High ergonomics.
- Two-position stand and the possibility of wall mounting.
- Suitable for secretarial work.
- 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) and to office IP PBXs (for example, Asterisk, 3CX IP PBX, Avaya IP Office).
- 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.
- Support for four simultaneous calls on eight independent SIP accounts.
- Support up to 6 extension consoles with 32 buttons each (total of 192 buttons).
- Adaptation for the work of the secretary (ergonomics, an additional RJ11 connector for the headset of the operator of the contact center).
- 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.
Specifications
VoIP
- RFC 3261 standard SIP server, Asterisk, Avaya, Cisco, Broadsoft, RTU MFI, 3CX IP PBX, Panasonic SIP-PBX and others.
- Encryption of SIPS signaling traffic and SRTP media traffic.
- Audio codecs: G.711 u / a, G.722 (HD Voice), G.729 a / b, G.723.1.
- QoS: TOS, Jiffer Buffer, VAD, CNG, G.168 (32ms).
- DNS SRV support.
- Eight SIP accounts with the ability to register on eight independent SIP servers and the ability to automatically switch in case of loss of registration.
- Up to two simultaneous calls to a SIP account and 8 calls to the phone.
- 8 independent SIP accounts.
Data transfer
- 2 * RJ45 10 / 100M Ethernet interfaces (LAN / PC).
- VLAN / QoS support.
- IP addressing: DHCP client or static IP destination.
- Built-in VPN client L2TP or SSL VPN.
- Network protocols HTTP, BOOTP, FTP, TFTP, IEEE 802.1Q (VLAN), IEEE 802.1X.
Physical parameters
- Color graphic LCD screen measuring 480x272 pixels (4.3 inches).
- 3 additional headset jacks — one of three ways to support the connection: with a USB connector, using jack jacks of 3.5 mm or RJ11.
- Line status indicator (two-color LED).
- Full duplex speaker and handsfree microphone.
- Eight line selection buttons 1-8 with light indication of line status.
- Buttons to adjust the volume of the phone / ring signal.
- 4 multifunction buttons below the screen.
- Any of the 8 line selection buttons can operate in BLF (Busy Lamp Field) mode, speed dial (number, prefix or SIP URI), or DTMF tone transmissions.
- 6 navigation multifunction buttons (4 navigation buttons, the “OK” button and the button for deleting the “C” symbol).
- Buttons of additional services: conference, transfer, hold and redial.
- Speakerphone button with light indication.
- Button "Mute microphone" with light indication.
- Button "Voicemail" with light indication.
- Button to switch to a headset with light indication.
- Button "Menu".
- Button "Service".
- Button "Directories" to access the directory.
- Connector for RJ11 tube connection.
- The possibility of mounting the phone on the wall.
- Adjustable stand.
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, button to start recording a conversation using the old code (if it supports IP PBX).
- BLF (Busy Lamp Field).
- 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
- Updating via protocols: HTTP / TFTP / (PnP auto-tuning) PnP auto-provision / TR069.
- Configuration: via the phone's on-screen menu / web-interface / auto-provision / TR069.
- Debugging: telnet / phone screen / web-interface.
Nutrition
- Adapter model AD300 (AV 220/110 Volt, output DC 12 Volt / 1A).
- USB powered via USB-AMAM cable (“dad” - “dad”).
- Power over Ethernet (IEEE 802.af) LAN port.
Package, appearance and packaging
Packaging
The phone is delivered in a cardboard box that shows the company logo, on the side of the package there is a sticker with the model number and barcode of the device.
Phone kit
Inside the phone is neatly packed, there is nothing superfluous in the box. Obviously, this equipment reduces the cost of the phone.
The box contains standard equipment, which includes:
- Telephone set.
- Handset.
- Handset cord.
- RJ45 patch cord to connect to the network.
- Instruction and warranty card.
The Escene ES620-PE model does not include the Es300 AD300 power supply unit (12 volts), it must be ordered separately. Also, the phone can be powered via a USB cable.
Front panel and hardware buttons
Conventionally, the buttons on the phone can be divided into 4 blocks.
The first block is the management of phone service functions.
- "Menu" button to access the on-screen menu.
- The “Service” button opens and closes the service menu.
- The "Directories" button allows you to access your phone's address book or one-touch call logs.
There is a separate voice mail call button with an envelope image, if there are unread messages in the voice mail box, the button lights up in red. A very useful button (the headset icon is shown above it) to switch to the headset and back, the button also has a light indicator, which allows the operator to control whether the headset is on or off. Also on the panel is a separate microphone mute button, if the microphone is turned off, the button is lit in red. The phone has a setting that allows you to automatically turn on the headset for any incoming call.
The second block is the management of additional functions.
Here are all the necessary buttons, just the ones that are used most often:
- Conference - creation of a 3-way conference (initiator, and two participants). Creating a conference with a large number of participants requires the support of such a function on an IP PBX.
- Transfer - call transfer during a call.
- Hold (Pickup) - during a conversation when you press the button, the call will be put on hold
- Redial - to redial the last number.
There are also two buttons for adjusting the phone / ring volume.
The third block - line management and multifunctional screen buttons
The phone has eight independent SIP accounts (eight SIP lines). By default, outgoing calls are established from line 1, if 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 line.
The phone can accept up to eight simultaneous calls. The Line 1-8 buttons have a light indication, when a call arrives, the diode of the line on which the call is 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.
Also, any of the eight line buttons can be reprogrammed and work in BLF mode, speed dial (number, prefix or SIP URI), or DTMF tone transmissions. If we are talking about dialing a prefix, this function differs from dialing that the saved combination is automatically dialed, but the block of digits is not sent from the phone to the IP PBX - the phone is waiting for the dialing of the remaining digits of the number.
The BLF (Busy Lamp Field) function allows you to monitor the current status of other subscriber lines in real time. When the button is in BLF mode, if the button is red, the line is busy, if it is green, the line is free.
The number of programmable buttons on the phone to expand to expand to 192, connecting the required number of expansion modules.
Each of the multi-function buttons below the screen displays the function currently active, for example: New call, end call, Do Not Disturb, Call Transfer and others.
The fourth block - multifunctional navigation keys
The block is used primarily for easy navigation through the menu, the “C” button is used to delete a character. Using the Up and Down buttons, you can adjust the ring volume or the volume of the phone during a call.
The panel has a separate large red button - “Hands-free”, which allows you to turn on or off the speakerphone (speakerphone), it is full-duplex in the phone. When the speakerphone is working, a red indicator lights up on the button.
The back of the phone. Wall mounting.
On the back of the phone is a standard sticker with the model number, serial number and MAC address. The model is equipped with a swivel stand, with which you can choose a convenient angle of the phone, you just need to press the buttons on the sides of the stand and set the desired angle.
For mounting on the wall, it is enough to bend the stand up to the stop, for installation in the stand there are two holes with which you can mount the phone on the wall.
Phone Interfaces & Connectors
The first photo shows a block of interfaces. For AC power using a power adapter, there is a 12-volt socket on the panel, two headset connectors and a tube with an RJ11 socket.
Two Ethernet interfaces - PC for connecting the phone to a computer and LAN for connecting to a local network and PoE power.
Two jack 3.5 mm are used to connect the headset. This is extremely convenient, since many headsets have such a connector.
USB connector can be used for:
- USB headset connection. The manufacturer does not guarantee the operation of any USB headset, a list of supported headsets will be prepared in the future.
- To power the phone via USB cable, instead of PoE or power supply. To do this, you need a USB cable type AMAM, that is, "dad-dad"
- His image is shown in the figure below. When working with expansion modules it is recommended to use the power from the adapter.
Here is the panel with the connected wires. The picture shows that all the wires are ergonomically connected and do not interfere.
View of the phone on the table
This is how the phone looks assembled, high-quality plastic, the screen of the phone is bright, sharp and clear, it looks very solid.
Escene ESM 32 Expansion Module
Expansion module - a console with 32 programmable buttons connected to the phone. Each button can be programmed to monitor the status of any SIP IP PBX line in BLF mode (busy / free / incoming call) or used for speed dialing, DTMF combination, dialing prefix or calling by SIP URI address.
The console is an indispensable tool for improving the performance of employees who work with a large number of calls. The important advantages can also be attributed to the lack of need for an additional power source for the console - all 6 expansion modules receive power from the phone.
As described above, up to 6 external expansion modules can be connected to the phone, we will take a closer look at the console.
The device comes in a compact box.
The delivery package includes an expansion module, a metal bracket with four screws for securely fastening the phone with an expansion module, a patch cord with RJ-45 connectors for connecting the console and the phone, and a warranty card.
On the rear panel are two connectors:
- IN - the input interface for connecting the phone and the extension console. In the case of connecting the first expansion module, you must connect this connector to the EXT connector on the back of the IP phone using a patch cord.
- OUT - output interface for connecting the next expansion module.
This is the look of the phone and the panel with the connecting bracket installed. To fix it you need to tighten the four screws, it can be seen in the photo.
The user manual says that after connecting the cable to the expansion module, to activate it, you need to restart the phone or open the menu — the Menu button on the telephone panel, then button 2 - “Functions”, then button 9 - “Expansion Module”, then press button from "1" to "6", which corresponds to the number of the expansion module that must be detected.
For example, pressing the "1" button initializes the first expansion module. In the latest version of the phone software, after connecting the console and rebooting the phone, it was determined by itself and no additional actions were required.
Expansion module and telephone on the table. The back stand of the module is fixed in the same way as the phone and therefore they stand exactly and are equally adjustable in height, the wires do not interfere. The design itself turned out to be reliable, there are no backlashes between the phone and the module.
The panel is successfully combined with the design of the phone of the same size and shape, has compact dimensions and conveniently located buttons, the size and location of the buttons makes it easy to accurately "get" into the right one.
Phone screen
It should be noted separately high-quality color screen with good resolution. The phone has a large color LCD screen with a diagonal of 4.3 inches and a resolution of 480x272 pixels, which makes it easy to read information from the screen and display all the necessary information on the screen.
This is how the phone screen with the registered line in Russian looks. “Number 10”, “number 11” and “ext 11” is an arbitrary label that is configured in the “SIP Accounts” menu and is called “Label”. The button with the name “Ignat” is programmed as a speed dial of the internal number, the “add” button is programmed as a DTMF code. international communication.
The example shows the wide possibilities for arbitrary programming of the buttons of the telephone lines.
View of the menu on the phone screen
Dialing a number
When dialing the number, the button of the line through which the number is dialed lights up in red, before sending the entered combination of numbers to the IP PBX, all lines are green, because the line has not been selected yet. By default, when you press the call button, it is sent via the first line, unless otherwise configured. To call through an arbitrary line, you must first click on the button of this line, it lights up in red, then dial the number and wait until the end of the waiting time for a set or immediately press the call button.
Incoming call. The button of the line to which the call came flashes in red.
State of conversation. During a call, the line button is green.
Call Logs
Due to the large screen, the phone shows 8 entries without scrolling.
Expansion panel in BLF mode
In the photo below, 16 lines work in the BLF mode, the 6th line is busy - the diode is lit in red. On line 7 incoming call - the diode flashes red. The remaining lines are free - the diodes are lit in green. Starting from the 17th line, the buttons are registered as speed dial, DTMF, dial prefix and SIP URI - in this mode the line diodes are extinguished.
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 "Menu" button or the "OK" button, it is located in the middle of the navigation button block, 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 softkey (for example, “Enter” or “Select”) to select a menu item, use the “C” button to return to the previous item .
Next, press the number 1 (or the “OK” button), which corresponds to the choice of the “Language” menu, using the Up or Down navigation buttons select “Russian” and press the “OK”
Then press the "C" button until you exit the menu.
Now you need to configure network settings
Click "Menu" (or "OK"), then select the "Settings" menu (or press number 6), number 2 - "Advanced settings", by default the 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 activated that tries to get 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, press the number 1 - “Type”, 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 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.
Special attention is given to the setting of the PC port (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, an IP and a mask are assigned to the PC port, 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, press "Menu" (or "OK"), press number 7 - "Status", then select "Network" (or press number 1) , the screen displays the IP assigned to the phone, in my case the IP address assigned by DHCP: 192.168.1.34
Setting up additional phone features
All 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
- VM number - set the number for access to voice mail (by default it is number * 97 - the standard number for accessing voice mail from the Asterisk distribution with FreePBX)
- The hotline allows you to set automatic dialing of a given number immediately or with a timeout set.
- Forwarding allows you to set conditional and unconditional forwarding to specified numbers
- Button - program extension panel buttons
- 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 eight independent SIP accounts, that is, registration with eight different IP PBXs. At simultaneous registration of all 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 first through eighth line buttons.
I note that the phone supports simultaneous calls, so to use simultaneous SIP registration on eight 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 4 dual-channel SIP accounts or 8 single-channel accounts; 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 eight at a time.
As for the DVO - they all work correctly
- The Conference button allows you to transfer a call; call transfer is implemented using the SIP 302 Moved Temporarily message. This message today almost all IP PBX on the market
- Transfer — , SIP 302 Moved Temporarily
- Hold ( Pickup) . 123, web- « » -> « » « »
- «Redial»
- « » , ,
- 1: «Directories», « ». , .
- 2: «» «OK», 3 ( « »)
- 3: «» — , «» — , «» — , «» —
web-
To access the web interface from a computer with 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 192.168.1.34
192.168.1.34 The
default login and password are:
root
root
There are two on the phone access level: an administrator level that can change any settings and a user who can perform a limited number of settings.
We get to the main menu of the web configurator of the phone. For convenience of configuration, 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 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.).
Consider the most important menu items phone. The “Setup Wizard” menu is used to quickly set up the phone, allows you to configure two tabs in sequence: the “Network” menu -> “LAN port” and the basic SIP account settings in the “SIP accounts” -> “Account1” menu. These tabs will be discussed in more detail below.
Network settings
Menu “Network” -> “LAN port”
You can set one of three connection methods: via DHCP, static IP or PPPoE. The 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).
Menu “Network” -> “PC port”
Between the LAN ports and the telephone PC, by default L2 switching is switched on - “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 “Advanced Settings” -> “VLAN Settings”
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.
Menu “Advanced Settings” -> “VPN Settings”
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 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.
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”
In addition to the standard SIP account settings - User Name (UserID), Name (AuthID) and password, there is a “Tag” field, it allows you to insert an arbitrary description of the line 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 two lines, because the second line must be assigned to the second 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.
The phone supports encryption of RTP and SIP signaling traffic using TLS protocol.
Menu "Programmable buttons"
In this menu you can customize the mode of operation of each of the 8 buttons of the phone lines.
The following modes are available:
- Asterisk BLF — Busy Lamp Field .
- Broadsoft BLF – , Broadsoft.
- – .
- – .
- DTMF – DTMF
- SIP URI – , , sip:ignat@ucexpert.ru
“Sound” menu
By default, during a call, the phone claims all possible codecs. If necessary, unused codecs can be disabled.
In the menu, you can adjust various volume parameters: handset, ringer, microphone, speakerphone. You can enable echo cancellation and VAD. Moreover, you can download your own ringtone.
Menu “Advanced Settings” -> “Global SIP Settings”
If you set the SIP settings here, they will be applied to all 8 lines automatically, except for the “Local SIP Port” and “RTP Port Range” settings, which can be useful for correct settings of the firewall.
Menu "Advanced Settings" -> "Phone Settings"
In this menu, additional functions are configured. Such as the “Hot Line” 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.
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 menu, you can configure 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).
You can intercept a call in three ways:
By pressing the “Hold” button — a combination for call pickup will be sent to the IP PBX, assigned in the Call Pickup Code field.
By assigning a speed dial combination to one of the extension panel buttons for call pickup. Using an explicit dialing combination on the phone keypad.
Menu "Phonebook"
The phone has a built-in phone book, which is quite advanced. It allows you to store up to 800 contact records, 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.
Service Settings
Debug menu
To debug your phone, you can enable logging by selecting the necessary logs (Phone Maintenance menu -> Log). You can view them in two ways:
- In the same menu, enable sending logs to the syslog server.
- In the “Phone maintenance” menu -> “HTTP update” download the file with logs.
Menu "Phone maintenance" -> "Auto Provision" (Automatic update)
Using this menu, you can configure automatic downloading of configuration, firmware and address book to your phone. You can download one of several protocols: http / https / ftp / tftp.
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”.
Backup and Software Update
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.
The status of the phone and system software can be viewed in the menu items “Status” and “System Information”. Here you can also find information about registering SIP phone accounts, for example, in our case two SIP accounts are registered - 10 and 11, with a capacity of two simultaneous calls each.
On the phone screen, the first two buttons will be assigned line 10, and the third and fourth line 11. In the lines “Account 1 and 2” the status will be “Registered”.
Since an extension module is connected to the phone, the corresponding status is displayed in the “Expansion Module 1” line: Connected.
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 (with FreePBX) + configure the BLF buttons, the second one on the virtual IP PBX:
IP address of the server with Asterisk = 10.10.10.1
UserID = 10
password = Tc6SAzsD
SIP server (Asterisk) = 10.10.10.1
In the Asterisk sip.conf configuration, this will be equivalent to:
[ten]
deny = 0.0.0.0 / 0.0.0.0
secret = Tc6SAzsD
dtmfmode = rfc2833
canreinvite = no
context = from-internal
host = dynamic
type = friend
nat = yes
port = 5060
qualify = yes
callgroup = 01
pickupgroup = 01
allow = ulaw
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 example of the first line (line 10), numbers from 12 to 15 will be used for BLF:
To work with Asterisk, it is enough to configure the Username (Username) = 10, password (Password) = Tc6SAzsD 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 “Line 1”.
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 "Submit".
Exactly the same must be done with the second line, for example, city number 78126470011 on the SIP server West Call. Let's register it on a virtual PBX with a non-standard SIP port 9966:
userid = 78126470011
authid = 6470011
password = eIoMzKsf
sip proxy = 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.
Further, in the menu “Expansion modules” -> “Expansion module 1” it is necessary to program the buttons of the first panel, all changes are applied without rebooting.
Setting up the BLF buttons for the Escene ESM 32 expansion module is very simple, you just need to specify the numbers for which you need to activate the BLF function, in our example in the picture these are lines from 12 to 17:
If everything is correct, then the first four buttons on the phone’s panel will become active - their status will be displayed, in my case in green, this means that the lines are free.
After successfully setting up the buttons in BLF mode, the buttons of the lines that are currently free will turn green, the buttons programmed in other modes (speed dial, DTMF, etc.) do not have a light indication (in the example button 17-20). In BLF mode, when you press the line button, the phone will automatically make a call to the line corresponding to this button.
findings
Key features of the phone include:
- Support eight independent SIP accounts per phone.
- A large, good-quality 4.3-inch color screen
- The presence of an additional Ethernet port for connecting to a computer and the ability to work in routing mode with NAT.
- The ability to connect the headset in one of three ways: via the RJ11 jack, 2x jack 3.5 mm or USB port.
- Most of the functions are displayed on the hardware buttons.
- The ability to reprogram the hardware buttons of the phone.
- Expansion module support.
- Can be mounted on the wall and height-adjustable stand.
- PoE support.
- The possibility of power through the USB port.
- The ability to configure in addition to network settings, SIP accounts, speed dial buttons and redirects directly from the phone screen.
The corporate Escene ES620-PE IP phone is a good choice if a beautiful and reliable phone should be on the table. The device is easily configured, work stably, do not lose registration, has good sound quality, additional functions (transfer, hold, call forward, BLF, etc.) also work stably. All this significantly increases the convenience and efficiency of working with the phone.
Support for 8 independent lines, large and bright color screen, good quality of parts and housing, connectivity up to 6 modules with 32 buttons without additional power, PoE and power supply via USB cable, two ways to connect the headset makes the phone a very attractive option for representative functions, as well as for handling a large number of challenges, for example, for managers and their secretaries. Summarizing what has been written, we can conclude that the phone is the best product in its class.
Source: https://habr.com/ru/post/210682/
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