To move to configuration mode, select one of the following tab 'Network', 'VoIP' or 'System' depending on the configuration goals:

• In the 'Network' menu, the network settings of the device are configured;
• In the 'VoIP' menu, the following is configured: SIP settings, accounts settings, codecs installation, VAS and dialplan settings;
• In the 'User Interface' menu, function keys and sound volume for different device operation modes are configured;
• In the 'System' menu, system, time, access to the device via different protocols are configured, passwords can be changed, firmware can be updated.

Configuration mode elements:

'Network' menu

In the 'Network' menu, the network settings of the device are configured.

'Internet' submenu

In the 'Internet' submenu, you can configure LAN via DHCP, Static and No IP.

Common settings

• Hostname — device network name.
• Speed and Duplex — specify data rate and duplex mode for LAN Ethernet port of the device:
  • Auto — automatic speed and duplex negotiation;
  • 100 Half — 100 Mbps data transfer rate with half-duplex mode is supported;
  • 100 Full — 100 Mbps data transfer rate with duplex mode is supported;
  • 10 Half — 10 Mbps data transfer rate with half-duplex mode is supported;
  • 10 Full — 10 Mbps data transfer rate with duplex mode is supported.

LAN

• Protocol — select the protocol that will be used for device LAN interface connection to a data network:
  • Static — operation mode where IP address and all the necessary parameters for LAN interface are assigned statically;
  • DHCP — operation mode where IP address, subnet mask, DNS address, default gateway and other necessary settings for network operation are automatically obtained from DHCP server;
  • No IP — operation mode when IP address is not assigned to the interface.

Static protocol

When 'Static' type is selected, the following parameters will be available for editing:

• IP Address — specify the device LAN interface IP address in the data network;
• Netmask — external subnet mask;
• Default gateway — address that the packet will be sent to, when route for it is not found in the routing table;
• 1st DNS Server, 2nd DNS Server  — domain name server addresses (allow identifying the IP address of the device by its domain name). You can leave these fields empty, if they are not required;
• MTU — maximum size of the data unit transmitted on the network.

DHCP protocol

When 'DHCP' type is selected, the following parameters will be available for editing:

• Alternative Vendor ID (Option 60) — when selected, the device transmits Vendor ID (Option 60) field value in Option 60 DHCP messages (Vendor class ID). When not selected, a default value is transmitted in Option 60 in the following format:
  • [VENDOR: device vendor][DEVICE: device type][HW: hardware version][SN: serial number][WAN: WAN interface MAC address][LAN: LAN interface MAC address][VERSION: firmware version]
    Example: [VENDOR:Eltex][DEVICE:VP-17P][HW:2.0][SN:VI23000118] [WAN:A8:F9:4B:03:2A:D0][LAN:02:20:80:a8:f9:4b][VERSION:#1.3.1].
• Vendor ID (Option 60) — option 60 value (Vendor class ID) which is transmitted in DHCP messages. When the field is empty, option 60 is not transmitted in DHCP messages;
• 1st DNS Server, 2nd DNS Server  — domain name server addresses (allow identifying the IP address of the device by its domain name). Addresses, which are specified statically, have the higher priority than addresses obtained via DHCP;
• MTU — maximum size of the data unit transmitted on the network.

You can manually assign the list of used DHCP options on each network interface.

No IP protocol

When this mode is selected, IP address will not be assigned to the network interface. This mode is used when IP telephony operates in an allocated VLAN.

Use VLAN

VLAN (virtual local area network) is a group of hosts united in a network not depending on the physical location. The devices grouped to a VLAN have the same VLAN identifier (ID). 

• Use VLAN — when selected, VLAN identifier specified below is used to enter the network:
  • VLAN ID — VLAN identifier which is used for the network interface;
  • 802.1P — 802.1P attribute (also called CoS — Class of Service) attached to egress IP packets from the interface. The value is from 0 (the least priority) to 7 (the highest priority). 

'QoS' submenu

In the 'QoS' submenu, Quality of Service functions configuring is available.

DSCP Configuration

• RTP — value of the DSCP field of the IP packet header for voice traffic;
• SIP — value of the DSCP field of the IP packet header for SIP signaling traffic.

The setting are common for the first and second accounts.

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

'MAC management' submenu

In the 'MAC management' submenu you can change MAC address of the device LAN interface.

• MAC — MAC address that will be assigned to the device network interface.

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button

'VoIP' menu

In the 'VoIP' menu you can configure VoIP (Voice over IP):

• Account configuration;
• Codec installation;
• VAS and dialplan configuration.

'SIP Accounts' submenu

Use drop-down 'Account' menu to select account for editing.

You can assign own SIP server addresses, registration servers, voice codecs, individualized dialing plan and other parameters for each account.

General settings

• Enable — when selected, account is active;
• Account Name — an account tag, which will be used for identifying active account or account by default;
• Phone — subscriber number assigned to the account;
• SIP Port — UDP port for incoming SIP message reception for this account, and for outgoing SIP message transmission from this account. It can take values from 1 to 65535 (default value: 5060);
• Voice Mail Number — a number which a call will be established to when subscriber selects 'Call' (to listen voice mail messages) in voice mail menu.

Authentication

• Login  — user name used for subscriber authentication on SIP server and on registration server;
• Password — password used for subscriber authentication on SIP server and on registration server.

SIP parameters

Use 'SIP Parameters' section to configure SIP parameters of the account.

• Proxy Mode — you can select SIP server operation mode in the drop-down list:
  • Off;
  • Parking — SIP-proxy redundancy mode without main SIP-proxy management;
  • Homing — SIP-proxy redundancy mode with main SIP-proxy management.

The phone can operate with a single main SIP-proxy and up to three redundant SIP-proxies. For exclusive operations with the main SIP-proxy, 'Parking' and 'Homing' modes are identical. In this case, if the main SIP-proxy fails, it will take time to restore its operational status.

For operations with redundant SIP-proxies, 'Parking' and 'Homing' modes will work as follows:

The gateway sends INVITE message to the main SIP-proxy address when performing outgoing call, and REGISTER message when performing registration attempt. If on expiration of 'Invite Total Timeout' there is no response from the main SIP-proxy or response 408 or 503 is received, the phone sends INVITE (or REGISTER) message to the first redundant SIP-proxy address. If it is not available, the request is forwarded to the next redundant SIP-proxy and so forth. When available redundant SIP-proxy is found, registration will be renewed on that SIP-proxy.

Next, the following actions will be available depending on the selected redundancy mode:

In the 'Parking' mode, the main SIP-proxy management is absent, and the phone will continue operation with the redundant SIP-proxy even when the main proxy operation is restored. If the connection to the current SIP-proxy is lost, querying of the subsequent SIP-proxies will be continued using the algorithm described above. If the last redundant SIP-proxy is not available, the querying will continue in a cycle, beginning from the main SIP-proxy.

In the 'Homing' mode, three types of the main SIP-proxy management are available: periodic transmission of OPTIONS messages to its address, periodic transmission of REGISTER messages to its address, or transmission of INVITE request when performing outgoing call. First of all, INVITE request is sent to the main SIP-proxy, and if it is unavailable, then the next redundant one, etc. Regardless of the management type, when the main SIP-proxy operation is restored, the gateway will use it to renew its registration. The gateway will begin operation with the main SIP-proxy.

• Home Server Check Method — select availability control method for the primary SIP server in 'Homing' mode:

  • Invite — control via transmission of INVITE request to its address when performing an outgoing call;

  • Register — control via periodic transmission of REGISTER messages to its address;

  • Options — control via periodic transmission of OPTIONS messages to its address.

• Home Server Keepalive Timeout — periodic message transmission interval in seconds; used for home SIP server availability check.

• Transport  — select protocol for SIP messages transport;

• Invite Initial Timeout, ms — a time interval between first INVITE transmission and the second one in case there is no answer on the first INVITE (ms). For the following INVITE requests (third, forth, etc.) the interval will be increased twice (i.e. if the value is 300 ms, the second INVITE will be sent in 300 ms, the third — in 600 ms, the forth — in 1200 ms, etc.);

• Invite Initial Max Timeout, ms  — the maximum time interval for retransmitting non-INVITE requests and responses on INVITE requests;

• Invite Total Timeout, ms — common timeout of INVITE requests transmissition (ms). When the timeout is expired, it is defined that the route is not available. INVITE requests retranslation is limited for availability definition as well;

• Subscribe for MWI — when selected, the subscription request on 'message-summary' events is sent. After obtaining such request, subscription server will notify the device on new voice messages through sending NOTIFY requsts;

• Subscriprion Server — a network address, to which SUBSCRIBE requests are sent for subscription on 'message-summary' and 'dialog' events. It is possible to specify IP address as well as domain name (after colon, specify a UDP port of SIP server, default value is 5060).

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

Proxy Addresses

To add the main SIP proxy and registration server, enter the following settings:

• Proxy Server — network address of the main SIP server. You can specify IP address as well as the domain name (specify SIP server UDP port after the colon, default value is 5060);
• Registration Server — network address of the main registration server (specify UDP port after the colon, default value is 5060). You can specify IP address as well as the domain name.

To add redundant SIP proxy and registration server, click 'Add' button and enter the following settings:

• Proxy Server — network address of redundant SIP server. You can specify IP address as well as the domain name (specify SIP server UDP port after the colon, default value is 5060);
• Registration Server — network address of redundant registration server (specify UDP port after the colon, default value is 5060). You can specify IP address as well as the domain name. If the 'Proxy Server' checkbox is selected, the redundant server registration is enabled.

To delete the redundant SIP proxy and registration server, select the checkbox next to the specified address and click 'Remove' button.

Additional SIP Properties

• SIP Domain — domain where the device is located (fill in, if needed);
• Use Domain to Register — when selected, apply SIP domain for registration (SIP domain will be inserted into the 'Request-Line' of REGISTER requests);
• Use Domain to Subscribe —  when selected, apply SIP domain for subscription (SIP domain will be inserted into the 'Request-Line' of SUBSCRIBE requests);
• Outbound Mode:
  • Off — calls will be routed according to the dialplan;
  • Outbound — dialplan is required for outgoing communications; however, all calls will be routed via SIP server; if there is no registration, PBX response will be sent to the subscriber in order to enable subscriber service management (VAS management);
  • Outbound with Busy — dialplan is required for outgoing communications; however, all calls will be routed via SIP server; if there is no registration, VoIP will be unavailable — error tone will be transmitted to the phone headset.
• Expires, s — valid time of account registration on SIP server. At the average, account registration renewal will be performed after 2/3 of the specified period;
• Registration Retry Interval, s — time interval between unsucceful attempt of SIP server registration and the next try;
• Subscription Expires, s — valid time of subscription on events. The subscription renewal is usually performed in 2/3 of the specified period;
• Subscription Retry Interval, s — time interval between unsucceful attempt of subscription on events and the next try;
• Public IP Address — this parameter is used as an external address of the device when it operates behind the NAT (gateway). As a public address, you can specify an external interface address (WAN) of a gateway (NAT) that the IP Phone operates through. At that, on the gateway (NAT), you should forward the corresponding SIP and RTP ports used by the device;
• Ringback at 183 Progress — when selected, 'ringback' tone will be sent upon receiving '183 Progress' message (w/o enclosed SDP);
• Reliable provisional responses (1xx) (100rel) — use reliable provisional responses (RFC3262):
  • Supported — reliable provisional responses are supported;
  • Required — reliable provisional responses are required;
  • Off — reliable provisional responses are disabled.

SIP protocol defines two types of responses for connection initiating requests (INVITE) — provisional and final. 2хх, 3хх, 4хх, 5хх and 6хх-class responses are final and their transfer is reliable, with ACK message confirmation. 1хх-class responses, except for 100 Trying response, are provisional and transferred unreliable, without confirmation (RFC3261). These responses contain information on the current INVITE request processing step, therefore loss of these responses is unacceptable. Utilization of reliable provisional responses is also stated in SIP (RFC3262) protocol and defined by 100rel tag presence in the initiating request. In this case, provisional responses are confirmed with PRACK message.

100rel setting operation for outgoing communications:

• • • Supported — send the following tag in INVITE request — supported: 100rel. In this case, communicating gateway can transfer provisional responses reliably or unreliably — as it deems fit;
    • Required — send the following tags in INVITE request — supported: 100rel and required: 100rel. In this case, communicating gateway should perform reliable transfer of provisional replies. If communicating gateway does not support reliable provisional responses, it should reject the request with message 420 and provide the following tag — unsupported: 100rel. In this case, the second INVITE request will be sent without the following tag — required: 100rel;
    • Off — do not send any of the following tags in INVITE request — supported: 100rel and required: 100rel. In this case, communicating gateway will perform unreliable transfer of provisional replies.

100rel setting operation for incoming communications:

• • • Supported,  Required — when the following tag is received in INVITE request — supported: 100rel, or required: 100rel — perform reliable transfer of provisional replies. If there is no supported: 100rel tag in INVITE request, the gateway will perform unreliable transfer of provisional replies;
    • Off — when the following tag is received in INVITE request — required: 100rel, reject the request with message 420 and provide the following tag — unsupported: 100rel. Otherwise, perform unreliable transfer of provisional replies.
• Timer Enable — when selected, the 'timer' (RFC 4028) extension support is enabled. When connection is established, and both sides support 'timer' extension, one of them periodically sends re-INVITE requests for connection monitoring purposes (if both sides support UPDATE method, wherefore it should be specified in the 'Allow' header, the session update is performed by periodic transmission of UPDATE messages);
• Min SE, s — minimal time interval for connection health checks in seconds (90 to 1800 s, 120 s by default);
• Session Expires, s — period of time in seconds that should pass before the forced session termination if the session is not renewed in time (90 to 80000 s, recommended value — 1800 s, 0 — unlimited session);
• Keepalive NAT Sessions Mode — select SIP server polling method:
  • Off — SIP server will not be polled;
  • Options — SIP server polling with OPTIONS message;
  • Notify — SIP server polling with NOTIFY message;
  • CLRF — SIP server polling with an empty UDP packet.

• Rejecting SIP Response — select SIP response on incoming call rejection;
• Use Alert-Info Header — process INVITE request 'Alert-Info' header to send a non-standard ringing to the subscriber port;
• Check RURI User Part Only — when selected, only subscriber number (user) will be analyzed, and if the number matches, the call will be assigned to the subscriber port. When cleared, all URI elements (user, host and port — subscriber number, IP address and UDP/TCP port) will be analyzed upon receiving an incoming call. If all URI elements match, the call will be assigned to the subscriber port;
• Send IP Address in Call‑ID Header — when selected, during outgoing communications, device custom IP address will be used in 'Call-ID' header in 'localid@host' format.

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

Codecs

• Codec 1..5 — you can select codecs and an order of their usage. The highest priority codec should be dragged to the top of the list. For operation, you should select the checkbox 'Enable' at least for one codec:
  • G.711a — use G.711A codec;
  • G.711u — use G.711U codec;
  • G.729 — use G.729 codec;
  • G.726-24 — use G.726 codec with the rate of 24 kbps;
  • G.726-32 — use G.726 with the rate of 32 kbps.

• Params:
  
  • Packet Time — amount of voice data in milliseconds (ms) transmitted in a single RTP packet for the corresponding codec G.711А, G.729, and G.726;
  • Payload Type — payload type of G.726-24 or G.726-32 codec (acceptable values are in the range from 96 to 127).

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

Service settings

• Call Waiting — when selected, the subscriber will accept incoming calls while being in a call state, otherwise '484 Busy here' reply will be sent;

• DND — when selected, temporary restriction is placed for incoming calls (DND service — Do Not Disturb);

• Stop Dial At # — when selected, use '#' button on the phone unit to end the dialing, otherwise '#' will be recognized as a part of the number;

• CLIR — when selected, limitation of caller number identification:

  • SIP:From — Anonymous sip: anonymous@unknown.host will be transmitted in the 'From' header of SIP messages;

  • SIP:From and SIP:Contact — Anonymous sip:anonymous@unknown.host will be transmitted in the 'From' and 'Contact' headers of SIP messages.

• Hotline — when selected, 'Hotline' service is enabled. This service enables an outgoing connection automatically without dialing the number after the phone handset is picked up with the defined delay (in seconds). When selected, fill in the following fields:

  • Hot Number — phone number that will be used for connection establishment upon 'Hot timeout' expiration after the phone handset is picked up (in SIP profile being used, a prefix for this direction should be defined in the dilaplan);

  • Hot Timeout, s — time interval that will be used for connection establishment with the opposite subscriber, in seconds.

• Allow Receiving Intercom Call — when cleared, incoming intercom calls are declined automatically:

  • Generate Tone — short sound signal is played before automatic answering to an incoming intercom call;

  • Intercom Сall Priority — when selected, an incoming intercom call has higher priority than an active call. Before answering to incoming intercom call, an active call is put on hold. When the option is disabled, the function of automatic answering to intercom calls during active call is disabled;

• Allow Auto Call Answering — when selected, all incoming calls will be answered automatically:
  • Notify Me Before Auto Answer — short audio signal is played before automatic answering;
  • Auto Call Answering Priority — when selected, an incoming call has higher priority than an active call. Before answering to incoming call, an active call is put on hold. When the option is disabled, the function of automatic answering to incoming calls during active call is disabled;
  • Auto Call Answering Delay, s — time interval in seconds between the incoming call and the automatic answer to it.
• Allow Call Pickup — when selected, pressing the BLF key will initiate the interception of the incoming call to the subscriber on which the BLF key is configured;
  • Call Pickup Mode — the way the call is intercepted:
    • Replaces — call pickup using the Replaces header;
    • Feature Code — call pickup using the prefix added to the number of the subscriber on which the BLF key is configured:
      • Call Pickup Code — prefix which will be added to the number of the subscriber to which the BLF key is configured;
      • Sign '#' terminates the number — adding the '#' symbol when intercepting a call after the number of the subscriber to which the BLF key is configured.
• Remote Call Control — when selected, processing of 'Event' headers in 'SIP Notify' is allowed in accordance with the Broadsoft: SIP Access Side Extensions Interface specification. When receiving the 'event: talk' header, the phone itself can answer the call, and when receiving the 'event: hold' header, the phone itself will send a request to hold the call. This is important for the 'line shifting' function, since the phone itself will remove the call on hold;
  • Generate Tone Before Answer — short audio signal is played before remote control answering;
  • Remote Call Answer Priority — when selected, a new incoming call has higher priority than an active call. The active call will be put on hold before the call is automatically answered. When the option is disabled, automatic call answering will not work when there is an active call.

Forwarding

• CFU — when selected, CFU (Call Forwarding Unconditional) service is enabled — all incoming calls will be forwarded to the specified CFU Number:
  • CFU Number — number that all incoming calls will be forwarded to when CFU service is enabled (in SIP profile being used, a prefix for this direction should be defined in the dialplan).
• CFB — when selected, CFB (Call Forwarding Busy) service is enabled — call forwarding to the specified CFNR Number, when the subscriber is busy:
  • CFB Number — number that incoming calls will be forwarded to when the subscriber is busy and CFB service is enabled (in SIP profile being used, a prefix for this direction should be defined in the dialplan).
• CFNR — when selected, CFNR (Call Forwarding No Reply) service is enabled — call forwarding, when there is no answer from the subscriber:
  • CFNR Number — number that incoming calls will be forwarded to when there is no answer from the subscriber and CFNR service is enabled (in SIP profile being used, a prefix for this direction should be defined in the dialplan);
  • CFNR Timeout — time interval that will be used for call forwarding when there is no answer from the subscriber, in seconds.

When multiple services are enabled simultaneously, the priority will be as follows (in the descending order):

1. CFU;
2. DND;
3. CFB, CFNR.

Three-party conference

• Mode — operation mode of three-party conference. Two modes are possible:
  • Local — conference assembly is performed locally by the device after pressing 'CONF' button;
  • Remote (RFC 4579) — conference assembly is performed at the remote server; after pressing 'CONF' button, 'Invite' message will be sent to the server using number specified in the 'Conference server' field. In this case, conference operation complies with the algorithm described in RFC 4579.

• Conference Server — in general, address of the server that establishes conference using algorithm described in RFC 4579. Address is specified in the following format SIP-URI: user@address:port. You can specify the 'user' URI part only — in this case, 'Invite' message will be sent to the SIP proxy address.

Additional Parameters

• DTMF Transfer — mode of DTMF signal transmission:
  • Inband — inband transmission;
  • RFC2833 — according to RFC2833 recommendation as a dedicated payload in RTP voice packets:
    • RFC2833 Payload Type — payload type for packet transmission via RFC2833 (possible values: from 96 to 127);
    • Use the Same PT Both for Transmission and Reception — option is used in outgoing calls for payload type negotiation of events sent via RFC2833 (DTMF signals). When selected, event transmission and reception via RFC2833 is performed using the payload from 200Ok message sent by the opposite side. When cleared, event transmission is performed via RFC2833 using the payload from 200Ok being received, and reception — using the payload type from its own configuration (specified in the outgoing Invite).
  • SIP info — transfer messages via SIP in INFO requests.
• RTCP — when selected, use RTCP for voice link monitoring:
  • Sending Interval — RTCP packet transmission period, in seconds;
  • Receiving Period — RTCP message reception period measured in transmission period units; if there is no single RTCP packet received until the reception period expires, the IP Phone will terminate the connection.
• Silence Detector — when selected, enable voice activity detector.

RTP

• Min RTP Port — lower limit of the RTP ports range used for voice traffic transmission;
• Max RTP Port — upper limit of the RTP ports range used for voice traffic transmission.

SRTP

• Enable — when selected, RTP flow encryption is used. Thus, the RTP/SAVP profile will be specified in SDP of outgoing INVITE requests. Also, the SDP of incoming requests will be scanned for the RTP/SAVP profile. If the RTP/SAVP profile is not found, the call will be rejected;
  • Crypto Suite 1-2 — allows to choose encryption and hashing algorithms to be used. A suite with the highest priority should be specified in 'Crypto Suite 1' field. You have to specify at least one crypto suit:
    • AES_80 — according to AES_CM_128_HMAC_SHA1_80;
    • AES_32 — according to AES_CM_128_HMAC_SHA1_32;
    • Off – RTP encryption will not be used.

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

Dialplan

To define a dialplan, use regular expressions in the 'Dialplan configuration' field. The structure and format of regular expressions that enable different dialing features are listed below.

Structure of regular expressions:

S xx , L xx  (Rule1 | Rule2 | ... | RuleN)

where:

• хх — arbitrary values of S and L timers;
• () — dialplan margins;
• | — delimiter for dialplan rules;
• Rule1, Rule 2, Rule N — numbers templates which are allowed or forbidden to be called.

Routing rules structure: 

Sxx Lxx prefix@optional(parameters)

where:

• хх — arbitrary value of S and L timer. Timers inside rules could be dropped; in this case, global timer values, defined before the parentheses, will be used.
• prefix — prefix part of the rule;
• @optional — optional part of the rule (might be skipped);
• (parameters) — additional options (might be skipped).

Timers

• Interdigit Long Timer ('L' character in a dialplan record) — entry timeout for the next digit, if there are no templates that correspond to the dialed combination.
• Interdigit Short Timer ('S' character in a dialplan record) — entry timeout for the next digit, if the dialed combination fully matches at least one template and if there is at least one template that requires an extension dialing for the full match.

The timers values might be assigned either for the whole dialplan or for a certain rule. The timers values specified before round brackets are applied for the whole dialplan. 

Example: S4 (8XXX.) or S4, L8 (XXX)

If the values of timers are specified in a rule, they are applied to this rule. The value might be located at any position in a template.  

Example: (S4 8XXX. | XXX) or ([1-5] XX S0) — an entry requests instant call transmission when 3-digit number dialing; a number should begin with 1,2, … ,5.

Prefix part of the rule

Prefix part might consist of the following elements: 

Optional part of rules 

The optional part of a rule might be omitted. This part might consist the following elements: 

Additional parameters

Format: (param1: value1, .., valueN; .. ;paramN: value1, .., valueN)

• param — parameter name; several parameters are semicolon-separated and all parameters are enclosed in parentheses;

• value — parameter value; several values of one parameter are comma-separated.

Valid parameters and their values:

Examples

Example 1: ( 8 xxx xxxxxxx ) — 11-digit number beginning with 8.

Example 2: ( 8 xxx xxxxxxx | <:8495> xxxxxxx ) — 11-digit number beginning with 8; if 7-digit number is dialed, add 8495 to the number being sent.

Example 3: (0[123] | 8 [2-9]xx [2-9]xxxxxx) — dialing of emergency call numbers and unusual sets of long-distance numbers.

Example 4: (S0 <:82125551234>) — quickly dial the specified number, similar to 'Hotline' mode.

Example 5: (S5 <:1000> | xxxx) — this dialplan allows you to dial any number that contains digits, and if there was no entry in 5 seconds, dial number '1000' (for example, it belongs to a secretary).

Example 6: (8, 10x.|1xx@10.110.60.51:5060) — this dialplan allows you to dial any number beginning with 810 and containing at least one digit after '810' (after entering '8', 'station reply' tone will be generated) as well as 3-digit numbers beginning with 1. Subscriber calls with 3-digit numbers beginning with 1 will be sent to IP address 10.110.60.51 and port 5060. 

Example 7: (S3 *xx#|#xx#|#xx#|*xx*x+#) — managment and usage of VAS.

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

'Phone Book' submenu

Local phone book management

Download Phone Book From Device

Use the section to download a phone book stored on the device. 

• File Format — select a format of the file you want to download. The following formats are available:
  • csv — text file format where all the conacts are written in the table. The values in the table are separated by the selected separator;
    
    • Separator — the symbol for separating data in the table in csv format;
    • Add Header — when selected, downloaded csv file will have a header — the first line.
• xml — an eXtensible Markup Language.

Upload Phone Book To Device

This section is used to configure parameters of restoring a phone bool from the backup copy.

• Phone Book File — choose file;
• Add Mode — when selected, the conacts from the uploaded file will be added to existing ones.

Clear Phone Book File

To clean the phone book, click 'Clear' button.

LDAP Phone Book management

In the 'Phone book' submenu, you can set up the connection to LDAP server and search parameters.

• Enable LDAP — when selected, the phone book is accessible via display menu;
  • LDAP Server Address — domain name or IP address of LDAP server;
  • LDAP Server Port — port of LDAP server transport protocol;
  • Base — indicates the location of base directory, that contains the phone book, and from which the search begins, in the LDAP directory. Specifying this parameter narrows the search and thereby reduces the time it takes to search for a contact;
  • Login — username that will be used when authorizing on LDAP server;
  • Password — password that will be used when authorizing on LDAP server;
  • Protocol Version — LDAP protocol version of formed requests;

  • Max Hits — the parameter indicating the maximum amount of search results that will be returned by LDAP server;

    Too big ‘Max Hits’ value reduces the LDAP search rate, that is why the parameter is to be configured according to the available bandwidth.

  • Name Attributes — the parameter that indicates the name attribute of each record returned by the LDAP server;

  • Number Attributes — the parameter that indicates the number attribute of each record returned by the LDAP server;
  • Display Name Attributes — the parameter that indicates the display name attribute of each record returned by the LDAP server;
  • Name Filter — the filter used to lookup for the names. The '*' character in the filter indicates any character. The '%' character in the filter indicates the input string used as the filter condition prefix;
  • Number Filter — the filter used to lookup for the number. The '*' character in the filter indicates any character. The '%' character in the filter indicates the input string used as the filter condition prefix;

  • Lookup For Incoming Call — when selected, lookup for a name using a number during incoming calls;

  • Lookup For Outcoming Call — when selected, lookup for a name using a number during outcoming calls.

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

Remote Phone Book management

• Enable Remote PhoneBook — when selected, remote phonebook is loaded automatically;
  • PhoneBook URL — a full path to the remote phonebook — is set in URL format (the following protocols are available to be used for phonebook loading through: TFTP, FTP, HTTP and HTTPS);
  • User Name — a name which is used for authentication on FTP/HTTP/HTTPS server for phonebook loading;
  • Password — a password which is used for authentication on FTP/HTTP/HTTPS server for phonebook loading;
  • File Format — select a format of the file you want to download. The following formats are available:
    • csv — text file format where all the conacts are written in the table. The values in the table are separated by the selected separator;
      
      • Separator — the symbol for separating data in the table in csv format;
      • Add Header — when selected, downloaded csv file will have a header — the first line.
  • xml — an eXtensible Markup Language.
  • Provisioning Mode — select a mode for phonebook loading:
    • Periodically — the device phone book will be automatically updated after defined period of time;
      • PhoneBook Update Interval, s — time interval between phonebook updates. If the parameter is set to 0, the phonebook is updated once — right after device loading.
    • Scheduled — the device phone book will be automatically updated at specific time and on specific days:
      • Days Of PhoneBook Update — weekdays when the phonebook will be automatically updated;
      • Time Of PhoneBook Update — time in 24-hours format, when the phonebook will be automatically updated.

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

Phone Book Priority management

In the 'Priority' submenu, you can configure the priority for displaying the subscriber’s name on the display.

• Local Contacts — displaying of names from local pnonebook;
• SIP Display Name — displaying of names received via SIP protocol;
• Remote Contacts — displaying of names from remote pnonebook;
• LDAP Contracts — displaying of names from LDAP pnonebook.

The caller's name will be displayed according to the selected priority. For example, in this case, if the local phone book has the name of the caller, the display will show the name from the local phone book, if not — the name designated in the SIP protocol. If the name is not designated in the SIP protocol, it will be displayed from the remote phone book, etc.

To apply a new configuration and store settings into the non-volatile memory, click 'SaveOrder' button. To discard changes, click 'Cancel' button.

'Call History' submenu

In the 'Call History' submenu you can configure call history logging.

• File Format — select a format of the file you want to download. The following formats are available:
  • csv — text file format where call history is written in a table. The values in the table are separated by the selected separator;
  • txt — text file format that contains call history organized by lines.
• Download Call History File — to save 'voip_history' file on a local PC, click 'Download' button;
• Clear Call History — to clear call history, click 'Clear' button.

To view the call history, follow the View 'Call History' link. For parameter monitoring description, see section 'View call history'.

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

'User Interface' menu

'Common Settings' submenu

In the 'Common Settings' submenu you can change phone user settings.

• Language of Phone screen menu — select screen menu language: Russian or English;
• Error display time and busy signal duration — the option is used to define the time interval in seconds during which an error will be displayed and a busy signal will be played.

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

'Buttons' submenu

You can choose actions for each button to be performed on pressing. The settings are presented as a table with the following columns:

1. Button;
2. Action — select action to be performed on the button pressing. The followings are available:
   1. No Action Selected — pressing this button will not be processed;
   2. Screen — open a screen selected in the additional parameters;
   3. Call — call the number selected in the additional parameters;
   4. Switch Account — change the account by default;

   5. BLF — pressing the button in stand-by mode initiates a call. In conversation mode, pressing the button redirects the call to the selected subscriber. 

      BLF — only for buttons with LED indicator. LED indicates line status of the subscriber selected in the additional settings.

      To activate BLF function, you should specify subscription server in SIP account settings.

   6. Account — open the dialer for the specified account;
   7. Forward — activate forwarding to a specified number;
   8. DND — temporary ban on incoming calls for all accounts;
   9. Custom DND — temporary ban on incoming calls for the specified account.
3. Label — button label, which is displayed on the screen next to the button;
4. Additional parameters— select additional parameters for the button (options depend on the action selected).

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

'System LED' submenu

In the 'System LED' submenu you can configure the operation of the system indicator and the priority for possible events. The indicator first displays the signal of the event that is placed higher in the priority table than the others. In the screenshot below, the highest priority event is 'Incoming Call', the lowest priority event is 'Power On'.

Possible indicator modes:

• Disabled;
• Green;
• Red;
• Blink green;
• Blink red;
• Blink green (fast);
• Blink red (fast);
• Alternately green, red;
• Alternately green, red (fast).

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

'Notifications' submenu

In the 'Notifications' submenu you can manage the notifications that are displayed on the device screen.

• Notify of Missed Calls — when selected, the display shows notifications of missed calls;
• Notify of Forwarded Calls — when selected, the display shows notifications of forwarded calls;
• Notify of Unread Messages — when selected, the display shows notifications of unread text messages;
• Notify of Unheard Voice Messages — when selected, the display shows notifications of unheard voice messages.

'Ringtones' submenu

In 'Ringtones' submenu, you can upload audio file and set it as ringtone. You can assign different ringtones for accounts.

This tab consists of 3 parts:

• a block for audio file uploading;
• drive free space indicator and total drive memory size for audio files storage;
• list of uploaded audio files.

The list of uploaded audio files is shown in a table with the following columns:

• Ringtone Name — name of the audio file;
• Account 1 — assignment of the ringtone to the Account 1;
• Account 2 — assignment of the ringtone to the Account 2;
• Size — the size of the file before being compressing;
• Actions — a button to play/pause audio file on the device. When the key is pressed, the audio file will be played.

'Audio' submenu

In the 'Audio' submenu you can configure the volume in various device operation modes.

Volume Settings

• Handsfree — speakerphone volume during conversation;
• Handset — handset volume during conversation;
• Headset — headset volume during conversation;
• Ringtone — ringtone volume.

Input Gain Control

• Handsfree — specifies the value by which a signal from the speakerphone will be amplified (valid values -9, … 9 dB, at a pitch of 1.5 dB);
• Handset — specifies the value by which a signal from the handset will be amplified (valid values -9, … 9 dB, at a pitch of 1.5 dB);
• Headset — specifies the value by which a signal from the headset will be amplified (valid values -9, … 9 dB, at a pitch of 1.5 dB).

Jitter Buffer

Jitter is a deviation of time periods dedicated to packet delivery. Packet delivery delay and jitter are measured in milliseconds. Jitter value is of great importance for real time data transfers (e.g. voice or video data). 

In RTP, there is a field for precision transmission time tag related to the whole RTP stream. Receiving device uses these time tags to learn when to expect the packet and whether the packet order has been observed. On the basis of this information, the receiving side will learn how to configure its settings in order to evade potential network problems such as delays and jitter. If the expected time for packet delivery from the source to the destination for the whole call period corresponds to the defined value, e.g. 50ms, it is fair to say that there is no jitter in such a network. But packets are delayed in the network frequently, and the delivery time period can fluctuate significantly (in the context of time-critical traffic). If the audio or video recipient application will play packets in the order of their reception time, voice (or video) quality will deteriorate significantly. For example, if the voice data is being transferred, there will be interruptions and interference in the voice.

The device features the following jitter buffer settings:

• Min Delay, ms — minimum expected IP package network propagation delay;
• Max Delay, ms — maximum expected IP package network propagation delay;
• Deletion Threshold (DT) — maximum time for voice package removal from the buffer. The parameter value should be greater or equal to maximum delay;
• Jitter Factor — parameter used for jitter buffer size optimization. The recommended value is 0.

Advanced

• Echocanceller — when selected, use echocanceller.

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

'System' menu

In the 'System' menu you can configure settings for system, time and access to the device via various protocols, change the device password and update the device firmware.

'Time' submenu

In the 'Time' submenu you can configure time synchronization protocol (NTP).

• Time Zone — select a timezone from the list according to the nearest city in your region;
• Time format — set time format: Hour 24 or Hour 12;
• NTP Server — time synchronization server IP address/domain name. Manual entering of server address or selection from a list are available;
• Period — the device time will be automatically updated after the specified period of time;
• Priority — allows selection of priority of obtaining the NTP server address:
  • DHCP — when selected, the device uses the NTP server address from DHCP messages in option 42 (Network Time Protocol Servers). DHCP protocol must be set for the main interface;
  • Config — when selected, the device uses the NTP server address from 'NTP Server' parameter.

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

'Access' submenu

In the 'Access' submenu you can configure the device access via web interface, Telnet and SSH protocols.

Access Ports

In this section you can configure TCP ports for the device access via HTTP, HTTPS, Telnet, and SSH.

• HTTP port — number of the port that allows for the device web interface access via HTTP, default value is 80;
• HTTPS ports — number of the port that allows for the device web interface access via HTTPS (HTTP secure connection), default value is 443;
• Telnet port — number of the port that allows for the device access via Telnet, default value is 23;
• SSH port — number of the port that allows for the device access via SSH, default value is 22.

You can use Telnet and SSH protocols in order to access the command line (Linux console). Username/password for console connection: admin/password.

Device Access

To get device access from the Internet service interfaces, set the following permissions:

Web

• HTTP — when selected, connection to the device web configurator is enabled via HTTP (insecure connection);

• HTTPS — when selected, connection to the device web configurator is enabled via HTTPS (secure connection).

Telnet — a protocol that allows you to establish mechanisms of control over the network. Allows you to remotely connect to the gateway from a computer for configuration and management purposes. To enable the device access via Telmet protocol, select the appropriate checkboxes.

SSH — a secure device remote control protocol. However, as opposed to Telnet, SSH encrypts all traffic, including passwords being transferred. To enable the device access via SSH protocol, select the appropriate checkbox.

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

'Log' submenu

In the 'Log' submenu you can configure output for various debug messages intended for device troubleshooting. Debug information is provided by the following device firmware modules:

• Configd Log — deals with the configuration file operations (config file reads and writes from various sources) and the device monitoring data collection;
• Networkd Log — deals with network configuration;

• VoIP Log — deals with VoIP functions operation;

• Interface Manager Log — deals with the device user interface operation (such as keyboard, display, speaker phone, handset etc.);

• Auto-update Log — deals with auto-updating;
  
• Remote phone book update log — deals with LDAP phone book updating.
  

Syslog Settings

If there is at least a single log is configured for Syslog output, it is necessary to enable Syslog agent that will intercept debug messages from the respective manager and send them to remote server or save them to a local file in Syslog format.  

• Enable — when selected, user Syslog agent is launched;
• Mode — Syslog agent operation mode:
  • Server — log information will be sent to the remote Syslog server (this is the 'remote log' mode):

    • Syslog server address — Syslog server IP address or domain name (required for 'Server' mode);

    • Syslog server port — port for Syslog server incoming messages (default value is 514; required for 'Server' mode).

  • Local File — log information will be saved to the local file:
    • File Name — name of the file to store log in Syslog format (required for 'Local File' mode);
    • File Size, kB — maximum log file size (required for 'Local File' mode).

  • Server and File — log information will be sent to the remote Syslog server and saved to the local file;

  • Console — log information will be sent to the device console (connection via a COM port adapter is required).

Configd Log, Network Log, VoIP Log, Interface Manager Log, Auto-update Log, Remote phone book update log

• Error — select this checkbox, if you want to collect 'Error' type messages;
• Warning — select this checkbox, if you want to collect 'Warning' type messages;
• Debug — select this checkbox, if you want to collect debug messages;
• Info — select this checkbox, if you want to collect information messages.
• SIP trace level — defines output level of VoIP SIP manager stack messages.

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

'Passwords' submenu

In the 'Passwords' submenu you can define passwords for administrator.

When signing into web interface, administrator (default password: password) has the full access to the device: read/write any settings, full device status monitoring.

• Password, Confirm — enter administrator password in the respective fields and confirm it.

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

'Configuration Management' submenu

In the 'Configuration management' submenu you can save and update the current configuration.

Backup Configuration

• Full — download full device configuration archive;
• Partial — download partial device configuration archive, which contains only user configuration.

To save the current device configuration to a local PC, click 'Download' button.

Restore Configuration

Select configuration file stored on a local PC. To update the device configuration, click 'Choose File' button, specify a file (in .tar.gz format) and click 'Upload' button. Uploaded configuration will be applied automatically without device reboot.

Reset to Default Configuration

To reset the device to default settings, click 'Reset' button.

'Firmware Upgrade' submenu

In 'Firmware upgrade' submenu you can update the firmware of the device.

• Active Version of Firmware — installed firmware version;

You can upgrade the device firmware manually by downloading the firmware file from the web site 
https://eltex-co.com/support/downloads/ and saving it on the computer. To do this, click the 'Choose File' button and specify path to firmware .tar.gz format file.

To launch the update process, click 'Upload file' button. The process can take several minutes (its current status will be shown on the page). The device will be automatically rebooted when the update is completed.

'Reboot' submenu

In the 'Reboot' submenu you can reboot the device.

Click the 'Reboot' button to reboot the device. Device reboot process takes approximately 1 minute to complete.

'Autoprovisioning' submenu

In the 'Autoprovisioning' submenu you can configure DHCP-based autoprovisioning algorithm.

Common Settings

• Parameters Priority from — this parameter manages names and locations of configuration and firmware files:
  • Static settings — paths to configuration, firmware, and manifest files are defined by the 'Configuration File' and 'Firmware File', and 'Manifest File' settings;
  • DHCP options — paths to configuration and firmware files are defined by the DHCP Option 43, 66, and 67 (it is necessary to select DHCP for the Internet service).

Automatic configuration updates

Configuration

• Provisioning Mode — to update configuration, you can specify one of the several update modes:

  • Periodically — the device configuration will be automatically updated after defined period of time;

    • Configuration Update Interval, s — time period in seconds that will be used for periodic device configuration update; if 0 is selected, device will be updated only once — immediately after startup;

  • Scheduled — the device configuration will be automatically updated at specific times and on specific days:

    • Time of Configuration Update — time on 24-hour format that will be used for configuration autoupdate;
    • Days of Configuration Update — week days with defined time that will be used for configuration autoupdate.
• Configuration File — full path to configuration file; defined in URL format:
  • http://<server address>/<full path to cfg file>
  • https://<server address>/<full path to cfg file>
  • ftp://<server address>/<full path to cfg file>
  • tftp://<server address>/<full path to cfg file>

where <server address> — HTTP, HTTPS, FTP or TFTP server address (domain name or IPv4),

          < full path to cfg file > — full path to configuration file on server.

Automatic software updates

Firmware

• Provisioning Mode — to update firmware, you can separately specify one of the several update modes:

  • Periodically — the device firmware will be automatically updated after defined period of time;

    • Firmware Update Interval, s — time period in seconds that will be used for periodic device firmware update; if 0 is selected, device will be updated only once — immediately after startup;

  • Scheduled — the device firmware will be automatically updated at specific times and on specific days:

    • Time of Firmware Update — time on 24-hour format that will be used for firmware autoupdate;
    • Days of Firmware Update — week days with defined time that will be used for firmware autoupdate.
• Firmware File — full path to firmware file; defined in URL format:
  • http://<server address>/<full path to firmware file>
  • https://<server address>/<full path to firmware file>
  • ftp://<server address>/<full path to firmware file>
  • tftp://<server address>/<full path to firmware file>

where <server address> — HTTP, HTTPS, TFTP or FTP server address (domain name or IPv4),

          <full path to firmware file> — full path to firmware file on server.

• Manifest File — full path to manifest file; defined in URL format. The use of the manifest file is due to the large size of the firmware file, which is downloaded periodically using the firmware auto-update algorithm. To reduce load on the network in such cases, it is recommended to use the Manifest file.
  The file structure is a line that specifies the firmware version identifier that is available for downloading and updating.
  For example, the contents of the Manifest file could be as follows: '1.2.0-b100'.

There is an optional ability to control the data integrity of the Manifest file, which consists of adding a line with an MD5 checksum to the file. If the Manifest file is specified, but an error occurs during network transmission and an incorrect checksum is received, then after a timeout specified in the configuration, an attempt will be made to obtain the Manifest file again. In this case, the contents of the Manifest file could be as follows:

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To discard changes, click 'Cancel' button.

'Certificates' submenu

'Certificates' submenu allows to view, download and upload certificates for using in protected TLS connections. To configure a certificate, click the needed certificate type.

Select the certificate and click 'Remove' button below the table to delete it.

Server certificate

Server certificate is used when accessing to the device web configurator via HTTPS.

• Certificate — information about certificate:

  • Serial Number — serial number of the selected certificate;

  • Not valid before — valid-from date;

  • Not valid after — valid-to date;

  • Key Length — amount of encryption symbols in bits.

• Subject — information about the certificate recipient (Сommon name, Organization, Subject alternative name);

• Name of the certification authority — information about the certification authority (Сommon name, Organization);

• Operation with certificate — possible actions to be done with  the certificate:
  • Download certificate — to save the certificate click 'Download' button;
  • Upload certificate — to update the current certificate choose the file by clicking 'Choose File' and then click 'Upload'.

Click 'Back' button to return to the list of certificates.

Client certificate

Client certificate is used with outbound connections via SIP with use of TLS.

• Certificate — information about certificate:

  • Serial Number — serial number of the selected certificate;

  • Not valid before — valid-from date;

  • Not valid after — valid-to date;

  • Key Length — amount of encryption symbols in bits.

• Subject — information about the certificate recipient (Сommon name, Organization, Subject alternative name);

• Name of the certification authority — information about the certification authority (Сommon name, Organization);

• Operation with certificate — possible actions to be done with the certificate:
  • Download certificate — to save the certificate click 'Download' button;
  • Upload certificate — to update the current certificate choose the file by clicking 'Choose File' and then click 'Upload'.

Click 'Back' button to return to the list of certificates.

Root certificate

Root certificate is used to authenticate certificates with incoming connections. This certificate must be signed by the certification authority.

• Certificate — information about certificate:

  • Serial Number — serial number of the selected certificate;

  • Not valid before — valid-from date;

  • Not valid after — valid-to date;

  • Key Length — amount of encryption symbols in bits.

• Subject — information about the certificate recipient (Сommon name, Organization, Subject alternative name);

• Name of the certification authority — information about the certification authority (Сommon name, Organization);

• Operation with certificate — possible actions to be done with  the certificate:
  • Download certificate — to save the certificate click 'Download' button;
  • Upload certificate — to update the current certificate choose the file by clicking 'Choose File' and then click 'Upload'.

Click 'Back' button to return to the list of certificates.

'Advanced' submenu

Use the menu to configure additional device settings.

Settings LLDP

• Enable LLDP — use LLDP when selected;
• LLDP transmit interval — time interval for messages transmission through LLDP. Default value is 60 seconds. 

To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button.To discard changes, click 'Cancel' button.