Escene ES410-PE IP Phone Review with Escene ESM 32 Expansion Module
We continue to move along the line of Escene corporate phones from simpler models to more advanced ones, today we are going to look at the Escene ES410-PE model and the Escene ESM 32 extension console.
Compared with the younger model of the ES320, this device has a screen with twice the resolution, four line buttons to which you can assign four independent SIP accounts and the ability to connect up to 6 expansion modules to 32 buttons.
The Escene ES410-PE phone supports PoE (Power over Ethernet), in the case of operation from a network without PoE, the Escene AD300 power supply unit will be suitable for the device . For a model with a PoE power supply unit is not included, but if necessary it can be purchased separately. The retail cost of the model at the moment is 5470 rubles .
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Positive features:
VoIP
Data transfer
Physical parameters
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.
Adapter model AD300 (AV 220/110 Volt, output DC 12 Volt / 1A).
Power over Ethernet (IEEE 802.af) LAN port.
USB powered with USB-AMAM cable (“dad” - “dad”)
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 delivery package for the Escene ES410-PE model is missing the Escence AD300 power supply unit (12 volts), it must be ordered separately.
Conventionally, the buttons on the phone can be divided into 5 blocks.
The first block is the management of the service functions of the phone.
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 is the management of lines and multifunctional buttons on the screen.
The phone has four independent SIP accounts (four 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 four simultaneous calls. The Line 1-4 buttons have light indication, when a call arrives, the diode of the line on 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.
Each of the multifunctional buttons displays the function currently active, for example: New call, end call, Do Not Disturb, Call Transfer and others.
The fourth block - 8 programmable buttons.
The buttons can work in BLF mode, speed dialing (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 increase to expand to 200 by connecting the required number of expansion modules.
The fifth block is 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.
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 in two ways:
Here is the panel with the connected wires. The picture shows that all the wires are ergonomically connected and do not interfere.
This is how the phone looks assembled, high-quality plastic, the screen backlight is not very bright, but bright enough to read messages on the screen without difficulty.
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.
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 the button from “1” to "6", which corresponds to the number of the expander that needs to be detected. For example, pressing the "1" button initializes the first expansion module.
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 screen with good resolution. The phone has a large monochrome LCD screen with a backlight resolution of 240x160 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. “L1 # 10” and “L1 # 11” is an arbitrary label that is configured in the “SIP Accounts” menu and is called “Label”.
When dialing a number, the button of the line through which the number is dialed lights up in red.
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.
In the photo below, 4 lines work in the BLF mode, 1, the line is busy - the diode is lit in red. On line 3 incoming call - the diode flashes red. Lines 2 and 4 are free - the diodes are green. The remaining lines are registered as speed dial, DTMF, dial prefix and SIP URI - in this mode the line diodes are canceled.
The same indication of the state of the lines and on the Escene ESM 32 expansion module connected to the phone.
The phone can be configured either using the phone menu or using the web 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 accessible from the phone menu in addition to the standard settings, also settings related to SIP accounts.
Such a move is fully justified, in some cases, you can configure the phone faster. In addition, sometimes there may be problems with access to the phone through a web-interface or it may be necessary to explain to an employee remotely how to reconfigure his phone. It will be easier for an unprepared person to use the phone menu, rather than a web interface.
Initial setup using phone buttons
So, we turned on the phone, connected the LAN port to the local network that has access to the IP PBX. Employee's computer through the cable connected to the PC port.
Now we need to include the Russian language in the menu:
All these settings are made in the “Menu” -> “Functions” (number 2)
The phone supports four independent SIP accounts, that is, registration on four different IP PBXs. If all 4 lines are registered simultaneously, by default, the first line will be used. To switch to the other lines (they must be configured) and return to the first one, use the “Line 1-4” buttons.
I note that the phone supports four simultaneous calls, but maximum two calls to a SIP account, so to use simultaneous SIP registration on 2 lines in the SIP account settings for each line, you must set the parameter “Number of lines used by the account” to 1 (default value is 2). That is, the device supports only four lines, you can distribute them at your discretion: either assign two lines to two SIP accounts, or distribute one line to each SIP account and register all four SIP accounts at the same time.
As for the DVO, they all work correctly:
:
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 192.168.1.200
192.168.1.200
Default login and password:
root
root
There are two access levels on the phone: the administrator level, which can change any settings and the 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 setup, immediately select the Russian language in the lower left menu:
The menu is divided into several groups:
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: 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"
L2 switching is switched on by default between the LAN and PC ports of the telephone - the “Bridge” mode. The phone can switch to L3 routing mode — a NAT address translation will start on the LAN port, an IP address will need to be configured on the PC port, and if necessary, a DHCP server must be enabled in which to set the pool of IP addresses for clients.
Menu “Network” -> “VLAN Settings”
In a corporate network it is recommended to isolate the traffic of a computer network from the traffic of a voice network, this is most often implemented using two VLANs. The phone supports VLAN on both ports.
Menu "Network" -> "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 only need to configure the VPN client built into the phone.Further, through the tunnel to register his phone on the IP PBX in the central office.
Secondly, VPN improves security, more and more administrators are thinking about how to protect terminals that are on the Internet, two problems are becoming more acute: the danger of hacking the terminal and the difficulty of accessing telecoms operators to configure it, because often the terminal is behind NAT. Using a VPN client solves both of these problems, so this useful feature will become increasingly popular.
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 “Label” 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 is 32 seconds by default, 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 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.
In this menu, you can configure the mode of operation of each of the 12 programmable buttons on the extension panel.
The following modes are available:
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.
<img src = " habrastorage.org/storage3/0c0/291/222/0c0291222d998fff555d93fe563edd1f.jpg " title = "Sound menu" alt = "Sound menu" />
If you set up the SIP settings here, they will be applied to all 4 lines automatically, except for the settings “Local SIP port” and “RTP port range”, which can be useful for correctly setting the firewall.
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).
Intercept the call in three ways:
The phone has a built-in phone book, which is quite advanced. It allows you to store up to 300 contact entries, each of which can store up to 3 phone numbers. Entries can be made via the phone's on-screen menu using the web-based interface. To download or save an already-created phone book in XML format, use the Phone Service menu -> Update over HTTP -> XML Phone Book; here you can save or load the phone book in XML format.
If your company uses an LDAP server, you can connect a phone to it and synchronize corporate contacts.2 and 3 protocol versions are supported, as well as using the “LDAP Lookup For Incoming Call” and “LDAP Lookup For PreDial / Dial” settings, you can search for the contact name for an incoming and outgoing call. If the contact is in the LDAP directory, then its name will be automatically added to the number.
The phone also supports blacklists or ban lists: an unwanted phone number is added to such a list and can no longer reach you.
Debug menu
To debug the phone, you can turn on logging by specifying the necessary logs (Menu “Phone maintenance” -> Log). You can view them in two ways:
1. In the same menu, enable sending logs to the syslog server.
2. In the “Phone maintenance” menu -> “Update over HTTP” 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.
For BLF to work, you need to enable this feature on Asterisk in the Free-PBX configuration files:
In the /etc/asterisk/sip_general_custom.conf file, you need to add lines that allow subscribers to monitor the status of the lines:
notifyringing = yes
notifyhold = yes
For more information on setting up BLF for Escene phones, click here.
Setting up BLF on the phone is very simple, you just need to specify the numbers for which you need to activate the BLF function, in our case, these are lines from 12 to 15:
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.
As for the setting of the buttons for the Escene ESM 32 expansion module, it is no different from the settings described above. In the menu "Expansion modules" -> "Expansion module 1", you must program the buttons of the first panel, all changes are applied without rebooting.
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 BLF mode, when you press the line button, the phone will automatically make a call to the line corresponding to this button.
Support for 4 simultaneous calls, a large and clear screen, the presence of a built-in programmable eight-button panel and the ability to connect up to 6 additional modules of 32 buttons without an additional power source, as well as three ways to connect the headset at once makes the phone a very attractive option for handling large call numbers, for example, to the secretary. Summarizing what has been written, we can conclude that the phone is an economical and efficient product in its class.
Key features of the phone include:
Compared with the younger model of the ES320, this device has a screen with twice the resolution, four line buttons to which you can assign four independent SIP accounts and the ability to connect up to 6 expansion modules to 32 buttons.
The company is positioning this model as an advanced IP phone, and indeed the size and resolution of the screen, the number of lines, the number of additional functions and buttons allows the phone to be called one of the most advanced devices in the corporate line.
The Escene ES410-PE phone supports PoE (Power over Ethernet), in the case of operation from a network without PoE, the Escene AD300 power supply unit will be suitable for the device . For a model with a PoE power supply unit is not included, but if necessary it can be purchased separately. The retail cost of the model at the moment is 5470 rubles .
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Positive features:
- Part of a single corporate level lineup.
- High quality body materials.
- Large and clear graphic screen.
- Ability to connect expansion modules.
- High ergonomics.
- Adjustable stand.
- Suitable for equipping the workplace of the secretary.
- Suitable for contact center operations.
- 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.
- Supports four simultaneous calls on four independent SIP accounts.
- Support up to 6 extension consoles with 32 buttons each (total of 192 buttons).
- Adaptation for the work of the operator in the contact center (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 additional functions.
- 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.
- 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.
- Up to two simultaneous calls to a SIP-account and 4 calls to the phone.
- 4 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
- Monochrome graphic LCD screen with backlight and size of 240x160 pixels.
- 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.
- Four buttons of the choice of lines 1-4 with light indication of a condition of the line.
- Buttons to adjust the volume of the phone / ring signal.
- 4 multifunction buttons below the screen.
- 8 programmable buttons that 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).
Power over Ethernet (IEEE 802.af) LAN port.
USB powered with USB-AMAM cable (“dad” - “dad”)
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 delivery package for the Escene ES410-PE model is missing the Escence AD300 power supply unit (12 volts), it must be ordered separately.
Front panel and hardware buttons
Conventionally, the buttons on the phone can be divided into 5 blocks.
The first block is the management of the service functions of the phone.
- "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 is the management of lines and multifunctional buttons on the screen.
The phone has four independent SIP accounts (four 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 four simultaneous calls. The Line 1-4 buttons have light indication, when a call arrives, the diode of the line on 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.
Each of the multifunctional buttons displays the function currently active, for example: New call, end call, Do Not Disturb, Call Transfer and others.
The fourth block - 8 programmable buttons.
The buttons can work in BLF mode, speed dialing (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 increase to expand to 200 by connecting the required number of expansion modules.
The fifth block is 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 in two ways:
- 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 backlight is not very bright, but bright enough to read messages on the screen without difficulty.
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 is 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.
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 the button from “1” to "6", which corresponds to the number of the expander that needs to be detected. For example, pressing the "1" button initializes the first expansion module.
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 screen with good resolution. The phone has a large monochrome LCD screen with a backlight resolution of 240x160 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. “L1 # 10” and “L1 # 11” is an arbitrary label that is configured in the “SIP Accounts” menu and is called “Label”.
Dialing a number.
When dialing a number, the button of the line through which the number is dialed lights up in red.
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.
Built-in expansion panel in BLF mode.
In the photo below, 4 lines work in the BLF mode, 1, the line is busy - the diode is lit in red. On line 3 incoming call - the diode flashes red. Lines 2 and 4 are free - the diodes are green. The remaining lines are registered as speed dial, DTMF, dial prefix and SIP URI - in this mode the line diodes are canceled.
The same indication of the state of the lines and on the Escene ESM 32 expansion module connected to the phone.
Phone setup
The phone can be configured either using the phone menu or using the web 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 accessible from the phone menu in addition to the standard settings, also settings related to SIP accounts.
Such a move is fully justified, in some cases, you can configure the phone faster. In addition, sometimes there may be problems with access to the phone through a web-interface or it may be necessary to explain to an employee remotely how to reconfigure his phone. It will be easier for an unprepared person to use the phone menu, rather than a web interface.
Initial setup using phone buttons
So, we turned on the phone, connected the LAN port to the local network that has access to the IP PBX. Employee's computer through the cable connected to the PC port.
Now we need to include the Russian language in the menu:
- Press the "Menu" button or the "OK" button, it is located in the middle of the navigation button block, a menu will open. To navigate through the menu, use the Up or Down navigation buttons; to return to the previous item, use the Back button. 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” button. Then press the "C" button until you exit the menu.
- Now you need to configure the network settings:
- Press "Menu", then select the "Settings" menu (or press the number 6), number 2 - "Advanced settings", the default password is empty, just click "OK". If you need to configure a VLAN (menu item 2), 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 press “Menu”. 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, as well as the port for accessing 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, press the “Menu” button, press the number 7 - “Status”, then the number 1 - “Network”, in my case the IP address assigned by DHCP: 192.168.1.200
- 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 if 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 set timeout.
- Forwarding allows you to set conditional and unconditional forwarding to the specified numbers.
- Button - program the buttons of the extension panel.
Support for additional services (DVO) and programmable buttons
The phone supports four independent SIP accounts, that is, registration on four different IP PBXs. If all 4 lines are registered simultaneously, by default, the first line will be used. To switch to the other lines (they must be configured) and return to the first one, use the “Line 1-4” buttons.
I note that the phone supports four simultaneous calls, but maximum two calls to a SIP account, so to use simultaneous SIP registration on 2 lines in the SIP account settings for each line, you must set the parameter “Number of lines used by the account” to 1 (default value is 2). That is, the device supports only four lines, you can distribute them at your discretion: either assign two lines to two SIP accounts, or distribute one line to each SIP account and register all four SIP accounts at the same time.
As for the DVO, they all work correctly:
- Conference , SIP 302 Moved Temporarily. IP .
- Transfer — , SIP 302 Moved Temporarily.
- Hold ( Pickup) . 123, web- « » -> « » « ».
- «Redial» .
- « » , , .
:
- 1: «Directories» , 2. , .
- 2: «Menu» «OK», 3 ( « »).
- 3: : «» — , «» — , «» — .
web-
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 192.168.1.200
192.168.1.200
Default login and password:
root
root
There are two access levels on the phone: the administrator level, which can change any settings and the 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 setup, 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: 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"
L2 switching is switched on by default between the LAN and PC ports of the telephone - the “Bridge” mode. The phone can switch to L3 routing mode — a NAT address translation will start on the LAN port, an IP address will need to be configured on the PC port, and if necessary, a DHCP server must be enabled in which to set the pool of IP addresses for clients.
Menu “Network” -> “VLAN Settings”
In a corporate network it is recommended to isolate the traffic of a computer network from the traffic of a voice network, this is most often implemented using two VLANs. The phone supports VLAN on both ports.
Menu "Network" -> "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 only need to configure the VPN client built into the phone.Further, through the tunnel to register his phone on the IP PBX in the central office.
Secondly, VPN improves security, more and more administrators are thinking about how to protect terminals that are on the Internet, two problems are becoming more acute: the danger of hacking the terminal and the difficulty of accessing telecoms operators to configure it, because often the terminal is behind NAT. Using a VPN client solves both of these problems, so this useful feature will become increasingly popular.
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 “Label” 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 is 32 seconds by default, 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 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 configure the mode of operation of each of the 12 programmable buttons on the extension panel.
The following modes are available:
- Asterisk BLF - Busy Lamp Field allows you to track the current state of the lines of other subscribers in real time.
- Broadsoft BLF - the same as the first point but with features for working with the Broadsoft platform.
- Speed ​​Dial - Allows you to dial a saved number with one touch.
- Speed ​​dial prefix - Allows you to dial a combination of numbers and then wait for the end of dialing from the subscriber.
- DTMF - allows you to send a saved DTMF combination
- SIP URI - allows you to dial a previously saved address, for example 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.
<img src = " habrastorage.org/storage3/0c0/291/222/0c0291222d998fff555d93fe563edd1f.jpg " title = "Sound menu" alt = "Sound menu" />
Menu "Advanced Settings" -> "Global SIP Settings"
If you set up the SIP settings here, they will be applied to all 4 lines automatically, except for the settings “Local SIP port” and “RTP port range”, which can be useful for correctly setting 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).
Intercept the call in three ways:
- 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 300 contact entries, each of which can store up to 3 phone numbers. Entries can be made via the phone's on-screen menu using the web-based interface. To download or save an already-created phone book in XML format, use the Phone Service menu -> Update over HTTP -> XML Phone Book; here you can save or load the phone book in XML format.
If your company uses an LDAP server, you can connect a phone to it and synchronize corporate contacts.2 and 3 protocol versions are supported, as well as using the “LDAP Lookup For Incoming Call” and “LDAP Lookup For PreDial / Dial” settings, you can search for the contact name for an incoming and outgoing call. If the contact is in the LDAP directory, then its name will be automatically added to the number.
The phone also supports blacklists or ban lists: an unwanted phone number is added to such a list and can no longer reach you.
Service Settings
Debug menu
To debug the phone, you can turn on logging by specifying the necessary logs (Menu “Phone maintenance” -> Log). You can view them in two ways:
1. In the same menu, enable sending logs to the syslog server.
2. In the “Phone maintenance” menu -> “Update over HTTP” 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.
Setting up BLF on the phone is very simple, you just need to specify the numbers for which you need to activate the BLF function, in our case, these are lines from 12 to 15:
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.
As for the setting of the buttons for the Escene ESM 32 expansion module, it is no different from the settings described above. In the menu "Expansion modules" -> "Expansion module 1", you must program the buttons of the first panel, all changes are applied without rebooting.
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 BLF mode, when you press the line button, the phone will automatically make a call to the line corresponding to this button.
findings
The enhanced Escene ES410-PE corporate phone is very attractive considering its cost and functionality. 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 4 simultaneous calls, a large and clear screen, the presence of a built-in programmable eight-button panel and the ability to connect up to 6 additional modules of 32 buttons without an additional power source, as well as three ways to connect the headset at once makes the phone a very attractive option for handling large call numbers, for example, to the secretary. Summarizing what has been written, we can conclude that the phone is an economical and efficient product in its class.
Key features of the phone include:
- Support four independent SIP accounts per phone.
- 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.
- Built-in programmable panel with 8 buttons.
- Expansion module support.
- Can be mounted on the wall and height-adjustable stand.
- PoE support.
- The possibility of power through the USB port.
- Clear backlit LCD screen.
- The ability to configure in addition to network settings, SIP accounts, speed dial buttons and redirects directly from the phone screen.
Source: https://habr.com/ru/post/196662/
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