Escene CC800 Contact Center IP Phone Overview
Today we look at the compact and ergonomic IP-phone Escene CC800 specially designed for comfortable work of the contact center and the constant handling of a large number of calls.
This device has a compact case and takes up little space on the table, its panel contains all the hardware buttons necessary for the contact center operator, and they are arranged so that it is as convenient to work with the phone. The phone has several connectors to choose how to connect the headset, on the side panel there is a special connector for monitoring the line by the supervisor.
If necessary, the phone can be optionally retrofitted with a headset stand. The headset itself is not included in the phone kit for several reasons; first, customers prefer to purchase them separately, the choice of headset is an individual matter, and depending on the specific tasks, the cost and modifications can vary significantly. Secondly, headsets can already be purchased from the customer, and finally, thirdly, the lack of a headset in the kit reduces the cost of an IP phone kit.
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The device is represented by two models: Escene CC800-PYN with PoE (power over Ethernet) and CC800-N without PoE.
Practice has shown that the CC800 series phones are always used near the computer and it is almost always more convenient to power them via USB. Therefore, the power supply was replaced with a USB extension cable, but if necessary it can be purchased separately.
At the same time, the retail price of a model with PoE is 2200 rubles and 2000 rubles without PoE, 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
The phone comes in a simple white cardboard box, on the side 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
Separately to the phone, you can order a headset stand (English U-Shaped Rack), which is mounted on the bottom of the phone.
The Escene AD200 power supply (5 volts) is not included in the delivery package; it can be ordered separately if necessary. Also, the phone can be powered via a micro-USB cable.
Conventionally, the buttons on the phone can be divided into 4 blocks.
The first 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 button flashes, then an incoming call has arrived. During a call - if the line is lit in red - an active call is on the line, if it is flashing - the call is held on the line.
Each of the multi-function buttons displays a function depending on the state of the line (dialing / conversation / hang up), for example, a new call, end a call, Do Not Disturb, Call Transfer, and others.
The second block is the management of additional functions.
There are all the necessary buttons here, just the ones that are used by contact center agents most often.
The third block - multifunctional navigation keys
When activating the IP phone menu, buttons 2, 4, 5, 6, 8 are used as navigation keys.
The block is used primarily for easy navigation through the menu, without placing special buttons on the telephone’s panel.
On the back of the phone is a standard sticker with the model number, serial number and MAC address. Despite its small size, the phone stands firmly on the table and does not move while working with the headset and keypad.
Also here are the point clips for mounting the headset stand, which must be ordered separately. It looks like the image below.
On the side panels there are RJ9 connectors for connecting an agent headset and an additional connector for monitoring the supervisor on the line. Of course, many contact centers allow you to join in a conversation using the functions of the contact center itself, but this is not always convenient, especially if the supervisor wants to be close to the agent, for example, for training and assistance.
If there is no headset with an RJ-9 connector, on the other side panel of the IP phone you can connect via a 3.5-inch Jack.
The back panel features an interface block for power and network connectivity.
For AC power supply using a power adapter, there is a 5 volt socket on the panel, here is also a micro USB connector, also for power.
Two Ethernet interfaces - PC for connecting the phone to a computer and LAN for connecting to a local network and power over PoE (for Escene CC800-PYN model).
Here is the panel with the connected wires.
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.
The phone has a monochrome LCD screen with a backlight of 128 * 64 characters, 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. “L1 # 10” is an arbitrary label that is configured in the “SIP Accounts” menu and is called “Label”.
Dialing a number
Incoming call
Conversation state
Call Lists
Menu view
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 available from the phone menu in addition to the standard settings, 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.
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, a menu will open. Use the “2” or “5” buttons to navigate the menu, and use the “Back” button on the OSD to return to the previous item. Next, select the “Language” menu, press the “Enter” button, then use the “2” or “5” navigation buttons to select “Russian” and press the “OK” button. Then press the back button until you exit the menu.
Now you need to configure the network settings:
Click "Menu", then select the menu "System Settings", then, item 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 want to use a static IP address, select item 1 - “Type”, select “Static” and click “OK”. By default, a DHCP client is configured on the phone, use the menu buttons and navigation keys to change the IP address, mask, gateway and DNS settings, and the phone will reboot after saving the settings. 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", select "View status", then item 1 - "Network", in my case, the IP address assigned by DHCP: 192.168. 1.208
All these settings are made in the "Menu" -> "Functions"
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.
Soft buttons become active on the phone screen during a call.
You can add or reassign soft buttons through the web-interface in the Programmable buttons -> Programmable keys menu.
To access the call logs, press the “Menu” button, then item 3 (corresponds to the “Call log” menu item). The call log contains records of the last outgoing, incoming, and missed calls. Each log stores up to 600 recent calls.
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 192.168.1.208
192.168.1.68
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 "Advanced Settings" -> "VPN Settings"
If you need to connect your phone via a secure VPN channel, this can be done directly from the phone, without buying additional equipment (for example, a VPN router), the phone supports L2TP and SSL VPN. This is a very useful feature for several reasons.
Firstly, if you have several phones that need to be delivered to a remote office, there is no need to buy a VPN hub at each remote location, you just need to configure the VPN client built into the phone. Further, through the tunnel to register his phone on the IP PBX in the central office.
Secondly, VPN improves security, more and more administrators are thinking about how to protect terminals that are on the Internet, two problems are becoming more acute: the danger of hacking the terminal and the difficulty of accessing telecoms operators to configure it, because often the terminal is behind NAT. Using a VPN client solves both of these problems, so this useful feature will become increasingly popular. In the example, using the VPN type L2TP, a connection to the vpn.ucexpert.ru server was created.
The phone allows you to manage a large number of SIP signaling settings and settings for RTP media traffic.
Menu “SIP accounts” -> “Account 1”
In addition to the standard SIP account settings - User Name (UserID), Name (AuthID) and password, there is a “Tag” field, it allows you to insert an arbitrary description of the line that will be displayed on the phone screen .
In addition to the primary IP address of the SIP server, you can add an additional IP SIP server. In case of unsuccessful registration during the timeout, which by default is 32 seconds, the address of the additional SIP server will be used for registration. The setting “Number of lines used by the account” should be equal to 1 if you need to use two lines, because the second line must be assigned to the second account. If you leave the value equal to 2, then when you apply the settings of the second line, the phone will display a message that there are not enough lines.
The phone supports encryption of RTP and SIP signaling traffic using TLS protocol.
Menu "Programmable buttons" - "Line"
In this menu you can configure the mode of operation of any of the two buttons on the phone line.
The following modes are available:
This can be useful if the first line is used, then the button of the second line can be programmed in one of the modes.
Menu "Programmable buttons" - "Programmable keys"
Allows you to manage the sets of soft buttons that appear on the screen, the phone, depending on the state of the phone (hang up, off hook, connect, talk, etc.). This is a very useful feature that allows you to control functions available to the user.
“Sound” menu
By default, during a call, the phone claims all possible codecs. If necessary, unused codecs can be disabled.
In the menu, you can adjust various volume parameters: handset, ringer, microphone, speakerphone. You can enable echo cancellation and VAD. Moreover, you can download your own ringtone.
Menu “Advanced Settings” -> “Global SIP Settings”
If you set the SIP settings here, they will be applied to all lines automatically, except for the “Local SIP Port” and “RTP Port Range” settings, which can be useful for correct network configuration. screen.
Menu “Advanced settings” -> “Phone settings”
In this menu, additional functions are configured. Such as the “Hotline” when you remove the headset, 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.
You can intercept a call in two ways - by assigning a speed-dial combination to one of the line buttons or by using the explicit dialing combination on the telephone 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.
Call forwarding is set in this block: unconditional if the subscriber has not answered or the line is busy.
Tab "Forwarding VOIP calls"
Menu "Phonebook"
The phone has a built-in phone book, and quite advanced. It allows you to store up to 800 contact records, each of which can store up to 3 phone numbers. Entries can be made via the phone's on-screen menu using the web-based interface.
To download or save an already-created phone book in XML format, use the Phone Service menu -> Update over HTTP -> XML Phone Book; here you can save or load the phone book in XML format.
If your company uses an LDAP server, you can connect a phone to it and synchronize corporate contacts. 2 and 3 protocol versions are supported, as well as using the “LDAP Lookup For Incoming Call” and “LDAP Lookup For PreDial / Dial” settings, you can search for the contact name for an incoming and outgoing call. If the contact is in the LDAP directory, then its name will be automatically added to the number.
The phone also supports blacklists or ban lists: an unwanted phone number is added to such a list and can no longer reach you.
Menu "Phone service" - "Log"
To debug the phone, you can enable logging by specifying the necessary logs. You can view them in two ways:
Also, the phone has the ability to collect network dumps of packets into pcap files, which can then be analyzed using a sniffer, for example, Wireshark, this is an extremely effective debugging tool.
To start capturing packets, click the “Start” button. 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.
Menu "Phone maintenance" -> "Automatic update".
Using this menu, you can configure the phone to automatically load the configuration, firmware and address book into the 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 with the inscription “Filename is illegal” will appear.
Software Backup and 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 on the registration status of SIP accounts, for example, in our case two SIP accounts are registered - 10 and 11, with the capacity for one simultaneous call.
The phone will display the default line 10, when you click on the “line 2” button, the second line will be activated, which corresponds to account 11. The “Registered” status will be displayed in the lines “Account 1 and 2”.
In the Network Status menu box, the current network settings are displayed.
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 sip.conf configuration, Asterisk will be equivalent to:
[ 10]
deny = 0.0.0.0 / 0.0.0.0
secret = QOXZuTcZ38qlBsr
dtmfmode = rfc2833
canreinvite = no
context = from-internal
host = dynamic
type = friend
= yes
port = 5060
qualify = yes
callgroup = 01
pickupgroup = 01
allow = g722
dial = SIP / 10
mailbox = 10 @ device
permit = 0.0.0.0 / 0.0.0.0
callerid = device <10>
callcounter = yes
faxdetect = no
Equivalent when configured in the Free-PBX web interface, using the first line as an example:
To work with Asterisk is enough 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 “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, the 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) of the additional setting not required.
The key features of the phone include
The Escene CC800 model is a good choice if you need to organize workplaces for contact center operators, the small dimensions of the phone, while maintaining ergonomics, make it easy to find a place on the table, powered by USB or PoE (optional) will help reduce the number of wires and the cost of connecting .
This device has a compact case and takes up little space on the table, its panel contains all the hardware buttons necessary for the contact center operator, and they are arranged so that it is as convenient to work with the phone. The phone has several connectors to choose how to connect the headset, on the side panel there is a special connector for monitoring the line by the supervisor.
If necessary, the phone can be optionally retrofitted with a headset stand. The headset itself is not included in the phone kit for several reasons; first, customers prefer to purchase them separately, the choice of headset is an individual matter, and depending on the specific tasks, the cost and modifications can vary significantly. Secondly, headsets can already be purchased from the customer, and finally, thirdly, the lack of a headset in the kit reduces the cost of an IP phone kit.
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An important feature of the phone is the possibility of power supply from the USB interface, without any additional power sources, for example, if there is no extra outlet in the workplace, you can connect the phone to the USB port of the computer.
The device is represented by two models: Escene CC800-PYN with PoE (power over Ethernet) and CC800-N without PoE.
Practice has shown that the CC800 series phones are always used near the computer and it is almost always more convenient to power them via USB. Therefore, the power supply was replaced with a USB extension cable, but if necessary it can be purchased separately.
At the same time, the retail price of a model with PoE is 2200 rubles and 2000 rubles without PoE, considering the capabilities of the phone, this is one of the most profitable offers on the market.
Positive features
- Designed specifically for contact center operation.
- USB power supply
- High quality body materials
- High-quality and contrast display
- The ability to enable / disable registration with one button
- One button auto answer setting
- High ergonomics and compact body
- Comfortable headset stand (optional)
- 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 (RJ9 connector or two 3.5-inch Jacks, for the headset of the contact center operator and an additional RJ9 connector for the supervisor's headset)
- Caller ID (Caller ID), call hold, 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
- Auto configuration via HTTP / TFTP / FTP, TR069
Specifications
VoIP
- RFC 3261 standard SIP server, Asterisk, Avaya, Cisco, Broadsoft, RTU MFI, 3CX IP PBX, Panasonic SIP-PBX, Huawei, Metaswitch, Alcatel-Lucent, Yeastar and others
- Encryption of SIPS signaling traffic and SRTP media traffic
- Audio codecs: G.711 u / a, G.722 (HD Voice), G.729a, G.723, iLBC
- 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 (VLAN), IEEE 802.1X
Physical parameters
- Monochrome LCD screen with backlight and the size of 128 * 64 characters with white backlight
- 4 multifunction buttons below the screen
- Headset or handset connectors - connects in one of two ways: two jack jacks 3.5 mm for headphones and a microphone or an RJ9 connector
- RJ9 connector for connecting a supervisor headset
- Two buttons of line 1 and line 2 with light indication of line status (two-color LED)
- Multi-function button to adjust the volume of the phone / microphone / ring tone
- Buttons of additional services: auto answer, hold, redial / mute microphone
- Answer / end button with light indication
- Micro USB connector for power supply
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, AON.
- Multilateral conference (if supports IP PBX), 3-way conference on the phone
- Enabling the "Auto Answer" mode with one button
- Possibility to enable / disable line registration with one button
- 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 / 1A)
- From the micro-USB connector (for example, a computer in the workplace)
- Power over Ethernet (802.3af, class 0) support per LAN port for CC800-PYN
- Power consumption 2.5-3.5 W
Package, appearance and packaging
The phone comes in a simple white cardboard box, on the side 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
- USB cable <-> Micro USB
- RJ45 patch cord for network connectivity
- Instruction and warranty card
Separately to the phone, you can order a headset stand (English U-Shaped Rack), which is mounted on the bottom of the phone.
The Escene AD200 power supply (5 volts) is not included in the delivery package; it can be ordered separately if necessary. Also, the phone can be powered via a micro-USB cable.
Front panel and hardware buttons
Conventionally, the buttons on the phone can be divided into 4 blocks.
The first 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 button flashes, then an incoming call has arrived. During a call - if the line is lit in red - an active call is on the line, if it is flashing - the call is held on the line.
Each of the multi-function buttons displays a function depending on the state of the line (dialing / conversation / hang up), for example, a new call, end a call, Do Not Disturb, Call Transfer, and others.
The second block is the management of additional functions.
There are all the necessary buttons here, just the ones that are used by contact center agents most often.
- “Auto Ans” - when you press the button, it lights up in red, which indicates that the automatic answer mode is activated when an incoming call is received
- “Transfer” - call transfer during a call.
- “Hold / Mute ringing” - during a call, when you press the button, the call will be put on hold, and the line button will flash. If the phone is free, when you press the button, it lights up in red, which means that now, when an incoming call arrives, the phone will send a ringing signal to the speaker, that is, “ring” loudly. The fact is that in the normal mode of the contact center, in order not to distract agents, an audible notification of an incoming call is sent to the headset.
- “Redial / Mute microphone” - the button is used to redial the last number or mute the microphone during a call.
- The headset button serves to answer an incoming call or hang up when pressed during an active call. If either of the two lines of the phone is active, the button is lit in red.
- The "Vol" button is used to activate the ringer volume, microphone, and headset volume control modes.
- The "vol-" and "vol +" buttons are used to change the volume level in the corresponding mode.
The third block - multifunctional navigation keys
When activating the IP phone menu, buttons 2, 4, 5, 6, 8 are used as navigation keys.
The block is used primarily for easy navigation through the menu, without placing special buttons on the telephone’s panel.
Back of phone
On the back of the phone is a standard sticker with the model number, serial number and MAC address. Despite its small size, the phone stands firmly on the table and does not move while working with the headset and keypad.
Also here are the point clips for mounting the headset stand, which must be ordered separately. It looks like the image below.
Interfaces and connectors phone.
On the side panels there are RJ9 connectors for connecting an agent headset and an additional connector for monitoring the supervisor on the line. Of course, many contact centers allow you to join in a conversation using the functions of the contact center itself, but this is not always convenient, especially if the supervisor wants to be close to the agent, for example, for training and assistance.
If there is no headset with an RJ-9 connector, on the other side panel of the IP phone you can connect via a 3.5-inch Jack.
The back panel features an interface block for power and network connectivity.
For AC power supply using a power adapter, there is a 5 volt socket on the panel, here is also a micro USB connector, also for power.
Two Ethernet interfaces - PC for connecting the phone to a computer and LAN for connecting to a local network and power over PoE (for Escene CC800-PYN model).
Here is the panel with the connected wires.
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
The phone has a monochrome LCD screen with a backlight of 128 * 64 characters, 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. “L1 # 10” is an arbitrary label that is configured in the “SIP Accounts” menu and is called “Label”.
Dialing a number
Incoming call
Conversation state
Call Lists
Menu view
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 available from the phone menu in addition to the standard settings, 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, a menu will open. Use the “2” or “5” buttons to navigate the menu, and use the “Back” button on the OSD to return to the previous item. Next, select the “Language” menu, press the “Enter” button, then use the “2” or “5” navigation buttons to select “Russian” and press the “OK” button. Then press the back button until you exit the menu.
Now you need to configure the network settings:
Click "Menu", then select the menu "System Settings", then, item 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 want to use a static IP address, select item 1 - “Type”, select “Static” and click “OK”. By default, a DHCP client is configured on the phone, use the menu buttons and navigation keys to change the IP address, mask, gateway and DNS settings, and the phone will reboot after saving the settings. 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", select "View status", then item 1 - "Network", in my case, the IP address assigned by DHCP: 192.168. 1.208
Setting up additional phone features
All these settings are made in the "Menu" -> "Functions"
- 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.
- Call forwarding (clause 6) allows you to set conditional and unconditional call forwarding to specified numbers
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.
Soft buttons become active on the phone screen during a call.
- The Conf button allows you to initiate a conference.
- Transfer button - call transfer with consultation and blindly (immediately after pressing Transfer, an additional SlPer button will appear), also uses SIP 302 Moved Temporarily.
- The Hold button allows you to or put a call on hold during a call.
You can add or reassign soft buttons through the web-interface in the Programmable buttons -> Programmable keys menu.
To access the call logs, press the “Menu” button, then item 3 (corresponds to the “Call log” menu item). The call log contains records of the last outgoing, incoming, and missed calls. Each log stores up to 600 recent calls.
Web Interface Overview
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 192.168.1.208
192.168.1.68
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 "Advanced Settings" -> "VPN Settings"
If you need to connect your phone via a secure VPN channel, this can be done directly from the phone, without buying additional equipment (for example, a VPN router), the phone supports L2TP and SSL VPN. This is a very useful feature for several reasons.
Firstly, if you have several phones that need to be delivered to a remote office, there is no need to buy a VPN hub at each remote location, you just need to configure the VPN client built into the phone. Further, through the tunnel to register his phone on the IP PBX in the central office.
Secondly, VPN improves security, more and more administrators are thinking about how to protect terminals that are on the Internet, two problems are becoming more acute: the danger of hacking the terminal and the difficulty of accessing telecoms operators to configure it, because often the terminal is behind NAT. Using a VPN client solves both of these problems, so this useful feature will become increasingly popular. In the example, using the VPN type L2TP, a connection to the vpn.ucexpert.ru server was created.
VoIP Settings
The phone allows you to manage a large number of SIP signaling settings and settings for RTP media traffic.
Menu “SIP accounts” -> “Account 1”
In addition to the standard SIP account settings - User Name (UserID), Name (AuthID) and password, there is a “Tag” field, it allows you to insert an arbitrary description of the line that will be displayed on the phone screen .
In addition to the primary IP address of the SIP server, you can add an additional IP SIP server. In case of unsuccessful registration during the timeout, which by default is 32 seconds, the address of the additional SIP server will be used for registration. The setting “Number of lines used by the account” should be equal to 1 if you need to use two lines, because the second line must be assigned to the second account. If you leave the value equal to 2, then when you apply the settings of the second line, the phone will display a message that there are not enough lines.
The phone supports encryption of RTP and SIP signaling traffic using TLS protocol.
Menu "Programmable buttons" - "Line"
In this menu you can configure the mode of operation of any of the two buttons on the phone line.
The following modes are available:
- – .
- – .
- DTMF – DTMF.
- SIP URI – , , sip:ignat@ucexpert.ru
- Paging – .
- Call Park – .
- Intercom –
This can be useful if the first line is used, then the button of the second line can be programmed in one of the modes.
Menu "Programmable buttons" - "Programmable keys"
Allows you to manage the sets of soft buttons that appear on the screen, the phone, depending on the state of the phone (hang up, off hook, connect, talk, etc.). This is a very useful feature that allows you to control functions available to the user.
“Sound” menu
By default, during a call, the phone claims all possible codecs. If necessary, unused codecs can be disabled.
In the menu, you can adjust various volume parameters: handset, ringer, microphone, speakerphone. You can enable echo cancellation and VAD. Moreover, you can download your own ringtone.
Menu “Advanced Settings” -> “Global SIP Settings”
If you set the SIP settings here, they will be applied to all lines automatically, except for the “Local SIP Port” and “RTP Port Range” settings, which can be useful for correct network configuration. screen.
Menu “Advanced settings” -> “Phone settings”
In this menu, additional functions are configured. Such as the “Hotline” when you remove the headset, 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.
You can intercept a call in two ways - by assigning a speed-dial combination to one of the line buttons or by using the explicit dialing combination on the telephone 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.
Call forwarding is set in this block: unconditional if the subscriber has not answered or the line is busy.
Tab "Forwarding VOIP calls"
Menu "Phonebook"
The phone has a built-in phone book, and quite advanced. It allows you to store up to 800 contact records, each of which can store up to 3 phone numbers. Entries can be made via the phone's on-screen menu using the web-based interface.
To download or save an already-created phone book in XML format, use the Phone Service menu -> Update over HTTP -> XML Phone Book; here you can save or load the phone book in XML format.
If your company uses an LDAP server, you can connect a phone to it and synchronize corporate contacts. 2 and 3 protocol versions are supported, as well as using the “LDAP Lookup For Incoming Call” and “LDAP Lookup For PreDial / Dial” settings, you can search for the contact name for an incoming and outgoing call. If the contact is in the LDAP directory, then its name will be automatically added to the number.
The phone also supports blacklists or ban lists: an unwanted phone number is added to such a list and can no longer reach you.
Service Settings
Menu "Phone service" - "Log"
To debug the phone, you can enable logging by specifying the necessary logs. You can view them in two ways:
- In the same menu, enable sending logs to syslog server
- Download log file
Also, the phone has the ability to collect network dumps of packets into pcap files, which can then be analyzed using a sniffer, for example, Wireshark, this is an extremely effective debugging tool.
To start capturing packets, click the “Start” button. 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.
Menu "Phone maintenance" -> "Automatic update".
Using this menu, you can configure the phone to automatically load the configuration, firmware and address book into the 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 with the inscription “Filename is illegal” will appear.
Software Backup and 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 on the registration status of SIP accounts, for example, in our case two SIP accounts are registered - 10 and 11, with the capacity for one simultaneous call.
The phone will display the default line 10, when you click on the “line 2” button, the second line will be activated, which corresponds to account 11. The “Registered” status will be displayed in the lines “Account 1 and 2”.
In the Network Status menu box, the current network settings are displayed.
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 sip.conf configuration, Asterisk will be equivalent to:
[ 10]
deny = 0.0.0.0 / 0.0.0.0
secret = QOXZuTcZ38qlBsr
dtmfmode = rfc2833
canreinvite = no
context = from-internal
host = dynamic
type = friend
= yes
port = 5060
qualify = yes
callgroup = 01
pickupgroup = 01
allow = g722
dial = SIP / 10
mailbox = 10 @ device
permit = 0.0.0.0 / 0.0.0.0
callerid = device <10>
callcounter = yes
faxdetect = no
Equivalent when configured in the Free-PBX web interface, using the first line as an example:
To work with Asterisk is enough 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 “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, the 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) of the additional setting not required.
findings
The key features of the phone include
- Support for two independent SIP-accounts on the phone.
- Compact body and high ergonomics especially for work in the contact center.
- Specially adapted for the continuous processing of a large number of calls.
- : RJ9, 2x jack 3.5 .
- RJ9 .
- USB
- 128x64
- /
- ()
- web- .
- , SIP
The device is easily configured, stably working, does not lose registration, has good sound quality (including support for HD), additional functions (transfer, retention, forwarding, etc.) also work stably. All this significantly increases the convenience and efficiency of working with the phone.
The Escene CC800 model is a good choice if you need to organize workplaces for contact center operators, the small dimensions of the phone, while maintaining ergonomics, make it easy to find a place on the table, powered by USB or PoE (optional) will help reduce the number of wires and the cost of connecting .
Source: https://habr.com/ru/post/212229/
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