Escene ES290 IP Phone Review
Today we look at the new IP phone Escene ES290 . In the line of corporate models of the company, the phone ranked between the younger model Escene ES220 and the older model Escene ES320, which was reviewed in a previous review. In terms of its characteristics and functional features, this innovative IP phone is closer to the older model.
When developing the devices, the engineers and designers of Escene used a unified approach to design, functional characteristics, as well as to control interfaces. The line includes models from basic to advanced phones, but they all share similar features - they meet the high requirements and have all the necessary functions of corporate IP-phones.
The device is presented in two models: Escene ES290-PN with PoE support (powered over Ethernet) and Escene ES290-N without PoE support (equipped with an Escene AD200 power supply unit). For a model with a PoE power supply unit is not included, but if necessary it can be purchased separately.
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At the same time, the retail price of the models today is 2650 rubles, considering the capabilities of the phone, this is one of the most profitable offers on the market.
Positive features
Functionality
VoIP
Data transfer
Physical parameters
Additional services (additional features)
Control
Nutrition
Packaging
The phone is delivered in the company's traditional cardboard box with the Escene logo, on the side of the package there is a sticker with the model number and barcode of the device.
Phone kit
Opening the box, we will see that the phone is neatly packed in a soft film, nothing more.
Inside the standard equipment, which includes:
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.
To the left of this block, there are two buttons for adjusting the phone / ring volume.
The second block is the management of additional functions.
Here are all the necessary buttons, just the ones that are used most often:
Below this block, there is a microphone mute button, if the microphone is turned off, the button is lit in red.
The third block - line management and multifunctional screen buttons
The phone has two independent SIP accounts (two SIP lines). By default, outgoing calls are established from line 1, unless of course it is set up, if necessary to make a call from line 2, you need to press the line button, then dial the number — the phone will send the call through the second SIP account.
The phone can accept two simultaneous calls. The “Line 1” and “Line 2” buttons are the light indication, when a call arrives, the diode of the line to which the call is being received flashes. If the line is busy, the line button is lit in red; if it is flashing, an incoming call has arrived. If the line is green, an active call is on the line; if it is flashing, the call is on hold
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 - 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.
Back of phone
On the back of the phone is a standard sticker with the model number, serial number and MAC address. If for convenience it is necessary to bring the wires of the handset or headset to the top of the phone, they can be laid on the phone body, for this, two grooves are located on the panel. The device can be hung up on a wall or be established on a table. To hang the phone on the wall, it is necessary to remove from it the holder “Part 1” which closes one of the mounting holes.
The design for installation on the table consists of two parts. To install the phone on the table, you need to attach the holder of the “Part 1” stand to the back panel, and then attach the “Part 2” stand itself to it, which are supplied in the kit. The stand can be installed at two angles, as shown in the figure below.
Phone Interfaces & Connectors
The photo shows a block of interfaces. For AC power using a power adapter, there is a 5 volt socket on the panel, two Ethernet interfaces — a PC for connecting the phone to a computer and a LAN for connecting to a local network and PoE power, and two connectors for a handset and a headset with an RJ9 jack .
On the back there are jack jack 3.5 mm and USB.
Jack 3.5 mm are used to connect the headset. This is extremely convenient, since many headsets have such a connector.
The USB connector is designed to power devices, such as charging a phone with a USB cable.
Here is the panel with the connected wires.
Wires do not interfere, the phone is exactly on the table.
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.
It is worth noting a nice screen with good resolution. The phone has a monochrome LCD screen with backlight size 132 * 64, not large, but its size is enough to easily read the information from the screen.
This is how the phone screen with the registered line in Russian looks. “Line1” and “Line 2” is an arbitrary label that is configured in the “SIP Accounts” menu and is called “Label”.
Enter and dial numbers
When dialing a number, the button of the line through which the number is dialed lights up in red
Incoming call
The call line button flashes red.
Conversation state
During a call, the line button is green.
Call Log
View of the menu on the phone screen
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 3 (or the “OK” button), which corresponds to the choice of the “System Settings” menu, then select “Phone Setting” (number 3), then “Language” (number 1), using the up or down buttons Down "select" Russian "and click" 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 menu "System Settings" (or press number 3), number 2 - "Advanced Settings", the default password is empty, just click "OK". If you need to configure a VLAN (menu item 2 - “Network”, 3 - VLAN), go to the appropriate menu and configure its ID and priority. Next, select “Network”, then “LAN port”, by default, after loading the phone, a DHCP client is 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, 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, the IP port and 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, click "Menu" (or "OK"), press the number 1 - "View status", using the up or down navigation buttons, find the IP assigned to the phone, in my case the IP address assigned by DHCP: 192.168.1.14
All these settings are made in the “Menu” -> “Functions” (number 2)
Support for additional services (DVO) and programmable buttons
The phone supports two independent SIP accounts, that is, registration on two different IP PBXs. At simultaneous registration of both lines, by default, the first line will be used. To switch to the second line (it should be configured) and return to the first one, use the “Line 1” and “Line 2” buttons.
I note that the phone supports two simultaneous calls, so to use simultaneous SIP registration on both lines in the SIP account settings for each line, you must set the "Number of lines used by the account" parameter to 1 (default value is 2). That is, the device supports only two lines, you can distribute them at your discretion: either assign both lines to the first SIP account, or distribute one line to each SIP account and register both at the same time.
As for the DVO - they all work correctly
To access call logs
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 192.168.1.14
Default login and password:
root
root
We get to the main menu of the web configurator of the phone. For convenience of setting, we immediately select the Russian language in the bottom left menu.
The menu is divided into several groups.
Menu "Phone Status"
The menu refers to the service settings, allows you to get detailed information about the status of settings and statistics of the phone, such as time in operation, the status and health of the network connection, the status of the registration of SIP-lines, the firmware version and others.
Menu "Network" -> "LAN port"
Basic Settings Tab
You can set one of three connection methods: DHCP, static IP, or PPPoE.
Advanced Settings Tab
It is also an important setting - HTTP and Telnet ports. They should be made non-standard if the phone is on an untrusted network (for example, with an external IP address on the Internet).
Also here you can configure Paging - group notification.
Menu "Network" -> "PC port"
L2 switching is switched on by default between the LAN and PC ports of the telephone - the “Bridge” mode.The phone can switch to L3 routing mode — a NAT address translation will start on the LAN port, an IP address will need to be configured on the PC port, and if necessary, a DHCP server must be enabled in which to set the pool of IP addresses for clients.
Network menu -> Advanced settings
VPN settings tab
If you need to connect your phone via a secure VPN channel, this can be done directly from the phone, without buying additional equipment (for example, a VPN router), the phone supports L2TP and SSL VPN. This is a very useful feature for several reasons.
Firstly, if you have several phones that need to be delivered to a remote office, there is no need to buy a VPN hub at each remote location, you just need to configure the VPN client built into the phone. Further, through the tunnel to register his phone on the IP PBX in the central office.
Secondly, VPN improves security, more and more administrators are thinking about how to protect terminals that are on the Internet, two problems are becoming more acute: the danger of hacking the terminal and the difficulty of accessing telecoms operators to configure it, because often the terminal is behind NAT. Using a VPN client solves both of these problems, so this useful feature will become increasingly popular. In the example, using the VPN type L2TP, a connection to the vpn.ucexpert.ru server was created.
VLAN Settings tab
In a corporate network, it is recommended to isolate the traffic of a computer network from the voice network traffic; this is most often implemented using two VLANs. The phone supports VLAN on both ports.
The phone allows you to manage a large number of SIP signaling settings and settings for RTP media traffic.
Menu “SIP accounts” -> “Account 1”
Tab “Basic”
Besides the standard settings of the SIP account - User Name (UserID), Name (AuthID) and password, there is a “Tag” field, it allows you to insert an arbitrary line description, which will be displayed on the phone screen.
In addition to the primary IP address of the SIP server, you can add an additional IP SIP server. In case of unsuccessful registration during the timeout, which by default is 32 seconds, the address of the additional SIP server will be used for registration. The setting “Number of lines used by the account” should be equal to 1 if you need to use both lines, because the second line must be assigned to the second 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.
Calls tab
Here you can specify additional settings for overcoming NAT, enable DNS SRV.
Security tab The
phone supports RTP encryption and SIP signaling traffic over TLS.
Menu “Phone Settings” -> “Basic”
Tab “Basic”
Here you can configure various phone functions, such as “Hotline” - when you pick up the handset, the preset number is automatically dialed, you can turn on autosearch in the address book during dialing and auto answer the call.
The important setting is DTMF type - by default it is set to RFC2833.
Calls tab
In this menu, global functions for the phone are configured.
If you set up the SIP settings here, they will be applied to both lines automatically, except for the settings “Local SIP port” and “RTP port range”, which can be useful for properly configuring the firewall.
In case the call transfer is required to be performed with a special key combination (old code), instead of the standard SIP message 302, this can be specified in the setting “Special code for call transfer”. A useful setting that allows you not to break the connection in the conference, if it left the initiator. You can set call forwarding by condition (busy and “no answer”) and unconditional.
Here you can also set codes that will be transmitted when you press the “Pickup” buttons (the value in the “Call Pickup Code” field) and “Voice Mail” (the value in the “Voice Mail Number” field).
There are three ways to intercept a call.
The important setting is “Return Code on Failure” and “Return Code at DND”, by default IP PBX returns SIP message 603 Decline, these messages can be changed to others, if necessary for correct interpretation of the reason for the release.
Tab “Forwarding VOIP calls”
The tab sets forwarding: unconditional, if the subscriber did not answer or the line is busy.
Menu “Phone Settings” -> “Advanced Settings”
Tabs “Sound - Basic” and “Sound - Advanced”
Here you can set the volume of the telephone, speaker and ringing tone. Also, the volume settings of the microphone tube and speakerphone. By default, when calling, the phone claims all possible codecs. If necessary, unused codecs can be disabled. Here you can enable echo cancellation and VAD. Moreover, you can download your own ringtone.
The “Line” tab of the “
Account ” menu can take the values ​​Account1 / Account2 / Any and becomes active if the button is assigned a dialing mode, for example, speed dial, DTMF or speed dial prefix.
Function Keys Tab
In this menu, you can assign an action to the function buttons of the phone if the actions they perform by default for some reason should be different. To do this, in the drop-down menu of the choice, specify the action that will be performed when you press the button.
Soft Keys
Allows you to manage the sets of soft buttons appearing on the phone screen depending on the state of the phone (the handset is on, the handset is off hook, connection, conversation, etc.) This is a very useful feature that allows you to control the functions available to the user.
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.
Menu "Phone Settings" -> "Basic"
You can copy configuration files using three different protocols. FTP, TFTP and HTTP - the choice of a particular protocol is a matter of taste and convenience. Software update is extremely simple, you need to select the firmware file, then click update. If the version of the downloaded firmware of the phone is lower than the installed one, a window will appear with the inscription “Filename is illegal”.In the menu, you can also restart the phone or reset it to the factory settings.
Menu “Phone Settings” -> “Advanced Settings”
To debug your phone, you can enable logging by specifying the necessary logs. You can view them in two ways:
Also in the phone it is possible to collect network dumps of packets into pcap files, which can then be analyzed using a sniffer, for example, Wireshark, this is a very effective debugging tool.
To start capturing packets, click the “Start” button. When finished, click the “Finish” button. To download the received dump, click the "Create backup" button.
Also on the “Automatic Update” tab, you can configure the phone's firmware update according to the TFTP / FTP / HTTP / HTTPS protocols.
“Security” menu
Here you can set a login and password for the administrator and the user of the phone, as well as download the SSL certificate.
Setting up connection to IP PBX Asterisk using web-interface
Suppose we need to configure two extensions (two SIP accounts).
For example, the first entry on the IP PBX Asterisk, the second on the virtual IP PBX:
IP address of the server with Asterisk = 10.10.10.1
UserID = 10
password = QOXZuTcZ38qlBsr
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 = QOXZuTcZ38qlBsr
dtmfmode = rfc2833
canreinvite = no
context = from-internal
host = dynamic
type = friend
nat = yes
port = 5060
qualify = yes
callgroup = 01
pickupgroup = 01
allow = g722
dial = SIP / 10
mailbox = 10 @ device
permit = 0.0.0.0 / 0.0.0.0
callerid = device <10>
callcounter = yes
faxdetect = no
Equivalently, when configured in the Free-PBX web interface, using the example of the first line:
To work with Asterisk, you only need to configure the Username (Username) = 10, password (Password) = QOXZuTcZ38qlBsr and SIP Server (SIP Server) = 10.10.10.1. You can add a label (Label) that will be displayed on the screen of the phone, in this case "L1 # 10".
You can reduce the re-registration time from standard 3600 seconds to 600 seconds, this is especially true if the IP PBX is located outside the office, for example, Virtual PBX. If the phone is on a local network and the IP PBX is on the Internet, no special settings are usually required to overcome NAT. Next, click the "Apply" button.
Exactly the same 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.
Corporate IP-phone Escene ES290 is one of the most profitable offers on the market, given its cost and functionality. In addition, the device has a stylish appearance, high quality plastic and good ergonomics.
The device is easily configured, it works stably, does not lose registration, has good sound quality, additional functions (transfer, retention, redirection, etc.) also work stably.
The key features of the phone include
When developing the devices, the engineers and designers of Escene used a unified approach to design, functional characteristics, as well as to control interfaces. The line includes models from basic to advanced phones, but they all share similar features - they meet the high requirements and have all the necessary functions of corporate IP-phones.
The phone is made in a strict style and is made of high quality materials, has a large bright screen, two independent lines, an intuitive interface in Russian, a huge set of functions, supports HD audio quality, two Ethernet ports and PoE power (optional).
The device is presented in two models: Escene ES290-PN with PoE support (powered over Ethernet) and Escene ES290-N without PoE support (equipped with an Escene AD200 power supply unit). For a model with a PoE power supply unit is not included, but if necessary it can be purchased separately.
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At the same time, the retail price of the models today is 2650 rubles, considering the capabilities of the phone, this is one of the most profitable offers on the market.
Positive features
- Part of a single corporate level lineup
- High quality body materials
- Large and clear graphic screen
- High ergonomics
- Two-position stand and wall mounting option
- Suitable for contact center operations.
- Easy setup due to clear interface
- Russified web-interface and OSD 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
- More functionality than most IP PBXs and carriers currently support.
Functionality
- Direct SIP connection to Virtual IP PBXs (for example, Broadworks, MFI RTU, Metaswitch, Alcatel-Lucent) and to office IP PBXs (for example, Asterisk, 3CX IP PBX, Avaya IP Office, Huawei)
- Two Ethernet ports (PC / LAN) with VLAN support and the ability to work in the switching or routing mode
- Easy installation and operation, advanced configuration options (including SIP and DVO functions) via the on-screen menu or via the web interface
- Supports two simultaneous calls on two independent SIP accounts
- 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
- Corporate Notebook Support via LDAP or XML or Personal Notebook
- Russified OSD menu and web-interface phone
- USB port for DC power supply with a voltage of 5 volts
- HTTP / TFTP / FTP auto setup, TR069
Specifications
VoIP
- RFC 3261 standard SIP server, Asterisk, Avaya, Cisco, Broadsoft, RTU MFI, 3CX IP PBX, Panasonic SIP-PBX, Huawei, Metaswitch, Alcatel-Lucent, Yeastar and others
- Encryption of SIPS signaling traffic and SRTP media traffic
- Audio codecs: G.711 u / a, G.722 (HD Voice), G.729a, G.723
- DTMF: In-Band, RFC2833, SIP Info, Auto
- QoS: TOS, Jiffer Buffer, VAD, CNG, G.168 (32ms)
- DNS SRV support
- Two SIP accounts with the possibility of registration on two independent SIP servers and the possibility of automatic switching in case of loss of registration
- Two simultaneous calls to the phone from any of the two SIP accounts
Data transfer
- 2 * RJ45 10 / 100M Ethernet interfaces (LAN / PC)
- Modes bridge / router PC port
- Support VLAN QoS (802.1pq) / QoS
- IP addressing: DHCP client or static IP destination.
- NAT Traversal: STUN mode
- Built-in VPN client L2TP or SSL VPN
- Network protocols HTTP, BOOTP, FTP, TFTP, IEEE 802.1Q, IEEE 802.1X
Physical parameters
- Monochrome LCD screen with backlight and the size of 132 * 64 characters with white backlight
- Additional headset connectors - connection is supported in one of two ways: Jack 3.5 mm jack or RJ9 connector
- Line status indicator (two-color LED)
- Full-duplex speaker and hands-free microphone (Full-duplex)
- Two buttons for selecting line 1 and line 2 with light indication of line status
- Buttons for adjusting the phone / ring volume
- 4 multifunction buttons below the screen
- 6 navigation multifunction buttons (4 navigation buttons, “OK” button and 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
- Voicemail button with light indication
- Button to switch to the headset with light indication
- RJ11 tube connector
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 repetition, auto answer
- Speed ​​dialing, start recording button for old-code conversation (if it supports IP PBX)
- Multilateral conference (if supports IP PBX), 3-way conference on the phone
- Do Not Disturb (DND)
- Voice mail (if the function is supported by IP PBX)
- Personal note book, corporate note book (LDAP or XML)
Control
- Protocols update: FTP / HTTPS / HTTP / TFTP / PnP auto-tuning (PnP auto-provision).
- Configuration: via phone's on-screen menu / web-interface / auto-provision
- SNMP V1 / 2, TR069
- Debugging: telnet / phone screen / web-interface.
Nutrition
- Adapter model AD200 (AV 220/110 Volt, output DC 5 Volt / 1.2A)
- Power over Ethernet LAN port (802.3af, class 0) for ES290-PN
- Power consumption 1.4-2.6 W
Package, appearance and packaging
Packaging
The phone is delivered in the company's traditional cardboard box with the Escene logo, on the side of the package there is a sticker with the model number and barcode of the device.
Phone kit
Opening the box, we will see that the phone is neatly packed in a soft film, nothing more.
Inside the standard equipment, which includes:
- Telephone set
- Handset
- Holder
- Phone stand
- Handset cord
- RJ45 patch cord for network connectivity
- Instruction and warranty card
- Escene AD200 power supply (for model ES290-N)
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.
- “Mail” for accessing voice mail, a button can be assigned another function. The voice mail button "Mail" with the image of the envelope, lights up in red if there are unread messages in the voice mail box
- “Handset” to switch to the headset, you can assign another function to the button. A very useful button (the headset icon is shown above it) to switch to the headset and back, the button also has an indicator light, which allows the operator to control whether the headset is on or off
To the left of this block, there are two buttons for adjusting the phone / ring volume.
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
Below this block, there is a microphone mute button, if the microphone is turned off, the button is lit in red.
The third block - line management and multifunctional screen buttons
The phone has two independent SIP accounts (two SIP lines). By default, outgoing calls are established from line 1, unless of course it is set up, if necessary to make a call from line 2, you need to press the line button, then dial the number — the phone will send the call through the second SIP account.
The phone can accept two simultaneous calls. The “Line 1” and “Line 2” buttons are the light indication, when a call arrives, the diode of the line to which the call is being received flashes. If the line is busy, the line button is lit in red; if it is flashing, an incoming call has arrived. If the line is green, an active call is on the line; if it is flashing, the call is on hold
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 - 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.
Back of phone
On the back of the phone is a standard sticker with the model number, serial number and MAC address. If for convenience it is necessary to bring the wires of the handset or headset to the top of the phone, they can be laid on the phone body, for this, two grooves are located on the panel. The device can be hung up on a wall or be established on a table. To hang the phone on the wall, it is necessary to remove from it the holder “Part 1” which closes one of the mounting holes.
The design for installation on the table consists of two parts. To install the phone on the table, you need to attach the holder of the “Part 1” stand to the back panel, and then attach the “Part 2” stand itself to it, which are supplied in the kit. The stand can be installed at two angles, as shown in the figure below.
Phone Interfaces & Connectors
The photo shows a block of interfaces. For AC power using a power adapter, there is a 5 volt socket on the panel, two Ethernet interfaces — a PC for connecting the phone to a computer and a LAN for connecting to a local network and PoE power, and two connectors for a handset and a headset with an RJ9 jack .
On the back there are jack jack 3.5 mm and USB.
Jack 3.5 mm are used to connect the headset. This is extremely convenient, since many headsets have such a connector.
The USB connector is designed to power devices, such as charging a phone with a USB cable.
Here is the panel with the connected wires.
Wires do not interfere, the phone is exactly on the table.
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.
Phone screen
It is worth noting a nice screen with good resolution. The phone has a monochrome LCD screen with backlight size 132 * 64, not large, but its size is enough to easily read the information from the screen.
This is how the phone screen with the registered line in Russian looks. “Line1” and “Line 2” is an arbitrary label that is configured in the “SIP Accounts” menu and is called “Label”.
Enter and dial numbers
When dialing a number, the button of the line through which the number is dialed lights up in red
Incoming call
The call line button flashes red.
Conversation state
During a call, the line button is green.
Call Log
View of the menu on the phone screen
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 3 (or the “OK” button), which corresponds to the choice of the “System Settings” menu, then select “Phone Setting” (number 3), then “Language” (number 1), using the up or down buttons Down "select" Russian "and click" 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 menu "System Settings" (or press number 3), number 2 - "Advanced Settings", the default password is empty, just click "OK". If you need to configure a VLAN (menu item 2 - “Network”, 3 - VLAN), go to the appropriate menu and configure its ID and priority. Next, select “Network”, then “LAN port”, by default, after loading the phone, a DHCP client is 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, 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, the IP port and 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, click "Menu" (or "OK"), press the number 1 - "View status", using the up or down navigation buttons, find the IP assigned to the phone, in my case the IP address assigned by DHCP: 192.168.1.14
Setting up additional phone 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 two independent SIP accounts, that is, registration on two different IP PBXs. At simultaneous registration of both lines, by default, the first line will be used. To switch to the second line (it should be configured) and return to the first one, use the “Line 1” and “Line 2” buttons.
I note that the phone supports two simultaneous calls, so to use simultaneous SIP registration on both lines in the SIP account settings for each line, you must set the "Number of lines used by the account" parameter to 1 (default value is 2). That is, the device supports only two lines, you can distribute them at your discretion: either assign both lines to the first SIP account, or distribute one line to each SIP account and register both at the same time.
As for the DVO - they all work correctly
- The 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 button - call transfer with consultation and blindly, also uses SIP 302 Moved Temporarily
- The Hold button (also Pickup) allows you to either put a call on hold during a call or pick up a call. By default, when you click on this button, the standard 123 combination is triggered, it can be reassigned via the web interface in the Advanced Settings -> Phone Settings menu, the Call Pickup Code option
- The "Redial" button allows you to redial the last number
- The "Speakerphone" button allows you to enable or disable the speakerphone, answer the call with the speakerphone on, or end the call if the conversation is held over the speakerphone
To access call logs
- Method 1: Click the "Journal". The call log contains records of recent outgoing, incoming, and missed calls.
- Method 2: press the "Menu" button or the "OK" button, then the number 5 (corresponds to the menu item "Call Log")
- Method 3: pressing the navigation button "Up" - opens all calls, "Down" - see missed calls, Button "Left" - see made calls, button "Right" - see log of received calls
Web Interface Overview
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 192.168.1.14
Default login and password:
root
root
We get to the main menu of the web configurator of the phone. For convenience of setting, we immediately select the Russian language in the bottom left menu.
The menu is divided into several groups.
- Network settings (interfaces, VLAN, VPN, etc.)
- VoIP settings (SIP records and additional functions for signaling and media traffic)
- Settings for additional phone functions (phonebook settings, programmable buttons, dial plan, sound, etc.)
- Service settings (logging, reset and reboot, configuration management and software updates, etc.)
Phone status
Menu "Phone Status"
The menu refers to the service settings, allows you to get detailed information about the status of settings and statistics of the phone, such as time in operation, the status and health of the network connection, the status of the registration of SIP-lines, the firmware version and others.
Network settings
Menu "Network" -> "LAN port"
Basic Settings Tab
You can set one of three connection methods: DHCP, static IP, or PPPoE.
Advanced Settings Tab
It is also an important setting - HTTP and Telnet ports. They should be made non-standard if the phone is on an untrusted network (for example, with an external IP address on the Internet).
Also here you can configure Paging - group notification.
Menu "Network" -> "PC port"
L2 switching is switched on by default between the LAN and PC ports of the telephone - the “Bridge” mode.The phone can switch to L3 routing mode — a NAT address translation will start on the LAN port, an IP address will need to be configured on the PC port, and if necessary, a DHCP server must be enabled in which to set the pool of IP addresses for clients.
Network menu -> Advanced settings
VPN settings tab
If you need to connect your phone via a secure VPN channel, this can be done directly from the phone, without buying additional equipment (for example, a VPN router), the phone supports L2TP and SSL VPN. This is a very useful feature for several reasons.
Firstly, if you have several phones that need to be delivered to a remote office, there is no need to buy a VPN hub at each remote location, you just need to configure the VPN client built into the phone. Further, through the tunnel to register his phone on the IP PBX in the central office.
Secondly, VPN improves security, more and more administrators are thinking about how to protect terminals that are on the Internet, two problems are becoming more acute: the danger of hacking the terminal and the difficulty of accessing telecoms operators to configure it, because often the terminal is behind NAT. Using a VPN client solves both of these problems, so this useful feature will become increasingly popular. In the example, using the VPN type L2TP, a connection to the vpn.ucexpert.ru server was created.
VLAN Settings tab
In a corporate network, it is recommended to isolate the traffic of a computer network from the voice network traffic; this is most often implemented using two VLANs. The phone supports VLAN on both ports.
VoIP Settings
The phone allows you to manage a large number of SIP signaling settings and settings for RTP media traffic.
Menu “SIP accounts” -> “Account 1”
Tab “Basic”
Besides the standard settings of the SIP account - User Name (UserID), Name (AuthID) and password, there is a “Tag” field, it allows you to insert an arbitrary line description, which will be displayed on the phone screen.
In addition to the primary IP address of the SIP server, you can add an additional IP SIP server. In case of unsuccessful registration during the timeout, which by default is 32 seconds, the address of the additional SIP server will be used for registration. The setting “Number of lines used by the account” should be equal to 1 if you need to use both lines, because the second line must be assigned to the second 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.
Calls tab
Here you can specify additional settings for overcoming NAT, enable DNS SRV.
Security tab The
phone supports RTP encryption and SIP signaling traffic over TLS.
Menu “Phone Settings” -> “Basic”
Tab “Basic”
Here you can configure various phone functions, such as “Hotline” - when you pick up the handset, the preset number is automatically dialed, you can turn on autosearch in the address book during dialing and auto answer the call.
The important setting is DTMF type - by default it is set to RFC2833.
Calls tab
In this menu, global functions for the phone are configured.
If you set up the SIP settings here, they will be applied to both lines automatically, except for the settings “Local SIP port” and “RTP port range”, which can be useful for properly configuring the firewall.
In case the call transfer is required to be performed with a special key combination (old code), instead of the standard SIP message 302, this can be specified in the setting “Special code for call transfer”. A useful setting that allows you not to break the connection in the conference, if it left the initiator. You can set call forwarding by condition (busy and “no answer”) and unconditional.
Here you can also set codes that will be transmitted when you press the “Pickup” buttons (the value in the “Call Pickup Code” field) and “Voice Mail” (the value in the “Voice Mail Number” field).
There are three ways to intercept a call.
- Pressing the "Hold" button - a combination for call pickup will be sent to the IP PBX, assigned in the "Call Pickup Code" field
- Assigning a speed dial combination to one of the extension panel buttons for call pickup
- Using the explicit dialing combination on the phone keypad
The important setting is “Return Code on Failure” and “Return Code at DND”, by default IP PBX returns SIP message 603 Decline, these messages can be changed to others, if necessary for correct interpretation of the reason for the release.
Tab “Forwarding VOIP calls”
The tab sets forwarding: unconditional, if the subscriber did not answer or the line is busy.
Menu “Phone Settings” -> “Advanced Settings”
Tabs “Sound - Basic” and “Sound - Advanced”
Here you can set the volume of the telephone, speaker and ringing tone. Also, the volume settings of the microphone tube and speakerphone. By default, when calling, the phone claims all possible codecs. If necessary, unused codecs can be disabled. Here you can enable echo cancellation and VAD. Moreover, you can download your own ringtone.
The “Line” tab of the “
Account ” menu can take the values ​​Account1 / Account2 / Any and becomes active if the button is assigned a dialing mode, for example, speed dial, DTMF or speed dial prefix.
Function Keys Tab
In this menu, you can assign an action to the function buttons of the phone if the actions they perform by default for some reason should be different. To do this, in the drop-down menu of the choice, specify the action that will be performed when you press the button.
Soft Keys
Allows you to manage the sets of soft buttons appearing on the phone screen depending on the state of the phone (the handset is on, the handset is off hook, connection, conversation, etc.) This is a very useful feature that allows you to control the functions available to the user.
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
Menu "Phone Settings" -> "Basic"
You can copy configuration files using three different protocols. FTP, TFTP and HTTP - the choice of a particular protocol is a matter of taste and convenience. Software update is extremely simple, you need to select the firmware file, then click update. If the version of the downloaded firmware of the phone is lower than the installed one, a window will appear with the inscription “Filename is illegal”.In the menu, you can also restart the phone or reset it to the factory settings.
Menu “Phone Settings” -> “Advanced Settings”
To debug your phone, you can enable logging by specifying the necessary logs. You can view them in two ways:
- In the same menu, enable sending logs to syslog server
- Download log file
Also in the phone it is possible to collect network dumps of packets into pcap files, which can then be analyzed using a sniffer, for example, Wireshark, this is a very effective debugging tool.
To start capturing packets, click the “Start” button. When finished, click the “Finish” button. To download the received dump, click the "Create backup" button.
Also on the “Automatic Update” tab, you can configure the phone's firmware update according to the TFTP / FTP / HTTP / HTTPS protocols.
“Security” menu
Here you can set a login and password for the administrator and the user of the phone, as well as download the SSL certificate.
Setting up connection to IP PBX Asterisk using web-interface
Suppose we need to configure two extensions (two SIP accounts).
For example, the first entry on the IP PBX Asterisk, the second on the virtual IP PBX:
IP address of the server with Asterisk = 10.10.10.1
UserID = 10
password = QOXZuTcZ38qlBsr
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 = QOXZuTcZ38qlBsr
dtmfmode = rfc2833
canreinvite = no
context = from-internal
host = dynamic
type = friend
nat = yes
port = 5060
qualify = yes
callgroup = 01
pickupgroup = 01
allow = g722
dial = SIP / 10
mailbox = 10 @ device
permit = 0.0.0.0 / 0.0.0.0
callerid = device <10>
callcounter = yes
faxdetect = no
Equivalently, when configured in the Free-PBX web interface, using the example of the first line:
To work with Asterisk, you only need to configure the Username (Username) = 10, password (Password) = QOXZuTcZ38qlBsr and SIP Server (SIP Server) = 10.10.10.1. You can add a label (Label) that will be displayed on the screen of the phone, in this case "L1 # 10".
You can reduce the re-registration time from standard 3600 seconds to 600 seconds, this is especially true if the IP PBX is located outside the office, for example, Virtual PBX. If the phone is on a local network and the IP PBX is on the Internet, no special settings are usually required to overcome NAT. Next, click the "Apply" button.
Exactly the same 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.
findings
Corporate IP-phone Escene ES290 is one of the most profitable offers on the market, given its cost and functionality. In addition, the device has a stylish appearance, high quality plastic and good ergonomics.
The device is easily configured, it works stably, does not lose registration, has good sound quality, additional functions (transfer, retention, redirection, etc.) also work stably.
The key features of the phone include
- Support of two independent SIP accounts per phone
- Existence of additional Ethernet port for connection to the computer and an opportunity to work in the IP routing mode
- The ability to connect the headset in one of two ways: through the RJ9 jack, 1x jack 3.5 mm
- Most of the functions are displayed on the hardware buttons.
- The ability to reconfigure the software and hardware buttons of the phone
- PoE support (model ES290-PN)
- Power supply via USB port
- Clear and bright backlit LCD screen
- The ability to configure in addition to network settings, SIP accounts, speed dial buttons and redirects directly from the phone screen.
- Advanced Debugging Features
Source: https://habr.com/ru/post/208530/
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