To configure the device, establish connection in the web-browser (hypertext document viewer), such as Google, Firefox, Internet Explorer, etc. Enter device IP address into address bar of the web browser. 

Upon entering IP address, the device will request username and password. it is also pissible to choose the language that will be used in the interface. 

After logging in with the default password, the system will ask you to create and enter a new password that meets the security criteria shown to the right of the input fields. The system automatically checks the password’s security; requirements that are met are marked with , that are not are marked with – .

If the password has expired, after successfully logging in with the old password, the user will see a message and be prompted to enter a new password that meets the criteria described above.

When web configurator access is established, the «System information» page will be displayed. 

The figure below shows web configurator navigation elements.

Figure 45 – Web configurator navigation elements

The user interface is divided into several areas:

To prevent unauthorized access to device in the future, it is recommended to change password (see 'Users: Management' menu).

System settings

System settings → Basic settings

• Device name – name of the device. This name is used in the device web configurator header; 
• Backup unsaved changes – if checked, the device creates a backup copy of unsaved configuration changes every 60 seconds for further recovery. For example, there were unsaved changes on the device and the power restarted. If the option was enabled after the device started, the window will be displayed in the web interface asking you to restore unsaved changes; 
• Local disk drive for traces – the debugging information (traces) can be saved on the device in random access memory (RAM) or on the installed drive: 
  
  • default – debug information is stored in RAM; 
    
  • /mnt/sdX – path to local storage device; setting is displayed when the storage device is installed. When the drive is selected, a logs directory will be created on it, which will contain trace files. 
    
• Active dial plan count – quantity of simultaneously active dial plans; up to 16 (up to 255 on SMG-2016 and SMG-3016 if there is a VAS license) independent dial plans can be configured with an ability to add subscribers and create custom call routing table;  
• Numbering plan wait for applying – if checked, SMG will not apply changes in the numbering plan without confirming. Setting this option helps to operate with long dial plans. It allows avoiding long processing of dial plans after every setting change; 
• Local disk drive for alarm logging – select the drive used for critical alarm message storage into nonvolatile memory. This option may be necessary when finding out the reasons for restarting or equipment failure: 
  
  • /mnt/sdX – select path to a local storage device. When this option is enabled, the file 'alarm.txt' containing alarm data will be created on the storage device. 
    

Example of alarm.txt file

1. 24/09/13 20:03:22. Software started.
2. 24/09/13 20:03:22. state ALARM. Sync from local source, but sync source table not empty
3. 24/09/13 20:03:22. state OK. PowerModule#1. Unit ok! or absent
4. 24/09/13 20:03:31. state OK. MSP-module lost: 1
5. 24/09/13 20:03:34. state OK. MSP-module lost: 2
6. 24/09/13 20:03:38. state OK. MSP-module lost: 3
7. 24/09/13 20:03:42. state OK. MSP-module lost: 4

File format description:

0, 1, 2… – event sequence number;

24/09/13 – event occurrence date;

20:03:22 – event occurrence time;

ALARM/OK – event current state (ОК – alarm is resolved, ALARM – alarm is active).

Table 19 — Alarm message examples

• SM-VP submodules usage – select SM-VP submodules, which will be in operation. 

Alarm indication

• Fans operation – when checked, fault indication will appear in case of cooling fan failure (ALARM LED will light up, alarm will be added to alarm log). 
• CPU load – when checked, fault indication will appear in case of high CPU utilization (ALARM LED will light up, alarm will be added to alarm log). 
• RAM usage – when checked, fault indication will appear in case of high RAM utilization (more than 75% of the total RAM amount) (ALARM LED will light up, alarm will be added to alarm log). 
• Local disk drive free space – when checked, fault indication will appear, if the utilization of a single external storage device with capacity less than 5GB exceeds 80% (or there is less than 1024MB of free space on an external storage device with capacity exceeding 5GB) (ALARM LED will light up, alarm will be added to alarm log).
• Alarms from slave device – when checked, the main device will receive alarms of the backup device; 
• Slave device connection – when checked, in the absence of communication with the slave on a global or local link, there will be an indication of an accident (the device will light up ALARM indicator, the accident will be recorded in the accident log).

Autoupdate settings

System settings → Autoupdate settings

SMG can automatically receive configuration and software version files from the auto-configuration server (hereinafter referred to as the “server”) with a specified period.

After downloading the configuration, SMG will wait for all active calls to end, after which will apply the new configuration or before a reboot.

Firmware version description file contains information about firmware versions on the server: versions and file names. In this file, the time to update can be also set. The file format should be as follows: 

<Firmware version>;<Firmware file name>;<Permitted update time, hour>

• Firmware version is specified completely up to the assembly version; 
• Firmware file name should have .bin extension; 
• The permitted update time can be unspecified. In this case, SMG will be updated in the near future, when there are no active calls. If a time interval is specified, then SMG will be updated only at the specified time interval. 

Example of firmware description file:

3.7.0.1944;smg1016m_firmware_3.7.0.1944.bin

3.8.0.2050;smg1016m_firmware_3.8.0.2050.bin;9-13

• Enable autoupdate – enable automatic firmware and configuration update; 
• Source – server information source:
  
  • Static – information about server is written and saved on SMG; 
  • DHCP (interface name) – server information will be received on the selected interface via DHCP protocol from option 66, information about the firmware file name and the configuration file name is obtained via option 67; 
• Protocol – выбор протокола для соединения с сервером;
• Authentication – use authentication to get access to the server (for FTP, HTTP, HTTPS); 
• Username – name (login) for access to the server; 
• Password – password for access to the server; 
• Server – IP address or domain name of server. Available if Static Sourcу is selected;
• Configuration update – allows configuration update from server; 
• Configuration file – configuration file name. The name should have .cfg extension and contains up to 64 symbols; 
• Configuration update interval, min – frequency of server validation for configuration update; 
• Firmware upgrade – enable firmware upgrade from server; 
• Firmware versions file – file name with firmware versions. The name should have .manifest extension and contains up to 64 symbols. 
• Firmware upgrade interval, min – frequency of server validation for firmware upgrade.

Upload configuration

System settings → Upload configuration

SMG can upload a configuration to FTP/TFTP/SCP-server automatically each time it is saved to nonvolatile memory. 

• Enable autoupload – enable the function of automatic configuration upload; 
• Protocol – select a protocol for uploading. FTP, TFTP, SCP are supported; 
• Server – IP address of the server for uploading the configuration; 
• Port – port of the server through which the uploading will be implemented; 
• Path to file – directory located on the server where the configuration will be stored; 
• Username – name for authentication in case of FTP using; 
• Password – password for authentication in case of FTP using. 

Ringback settings

System settings → Ringback settings

RingBack settings allow replacing the standard ringing sound with any other one, similarly operation of the “Replace the horn” service.

• Local disk – path to the external drive where audio files will be stored; 
• Directory name – the name of the folder on the external drive where the audio files are stored; 
• File name – desired file to play; 
• Mode:
  • RingBack – standard ringback sound; 
  • Audio file – a special file selected as audio for RBT. 

The ‘Overview’ submenu allows you to upload, select and delete the audio files you want to use as RBT:

System settings → RingBack settings → Browse 

• Upload — uploading an audio file of a specific format; 

• Apply — selecting the desired audio file; 
• Cancel — exit from the ‘Browse’ submenu 

When setting up RBT from the ‘System parameters’ item, the audio file is applied to all subscribers and system trunk groups. 

There are several levels of settings, each next “more detailed” level has priority over previous: 

1. RBT system settings 
2. RBT settings for Trunk groups and PBX profiles;
3. RBT settings for subscribers.

Monitoring

Monitoring → Telemetry

Telemetry

This section contains information on the device telemetric sensor readings as well as the information on power supplies and fans installed. 

Temperature sensors:

For SMG-1016M, SMG-3016:

• Sensor #0 – CPU temperature; 
• Sensor #1 – RAM module temperature. 

For SMG-2016 :

• Sensor #0 – CPU temperature.

Power supply:

• Power module #0 – status of power supply installed in slot 0; 
• Power module #1 – status of power supply installed in slot 1.
  

Possible power supply states:

• Installed – power supply is installed; 
• Not installed – power supply is not installed; 
• In operation – power supply is energized; 
• Not in operation – power supply is de-energized.

Fans ¹:

• Fan #N –  information on fan N and its rotation speed (e.g. 9600 rpm). 

Voltage ¹:

• Intenal voltage (+12V) – 12V voltage sensor status details. 

Current voltage ²:

• +12.0 V – 12V voltage sensor status details; 
• +5.0 V – 5V voltage sensor status details; 
• +3.3 V – 3.3V voltage sensor status details; 
• +2.5 V – 2.5V voltage sensor status details; 
• +1.8 V – 1.8V voltage sensor status details; 
• +1.5 V – 1.5V voltage sensor status details; 
• +1.2 V – 1.2V voltage sensor status details; 
• +1.0 V – 1V voltage sensor status details; 
• CPU – CPU voltage status details; 
• CPU Vcore – CPU core voltage status details; 
• RTC battery – real-time clock battery voltage status details. 

CPU load:

• USR – percentage of CPU time utilization by user applications; 
• SYS – percentage of CPU time utilization by core processes; 
• NIC – percentage of CPU time utilization by applications with modified priority; 
• IDLE – percentage of unused CPU resources; 
• IO – percentage of CPU time spent on I/O operations; 
• IRQ – percentage of CPU time spent on hardware interruptions' processing; 
• SIRQ – percentage of CPU time spent on software interruptions' processing. 

E1 streams

The section displays information about installed chips on C4E1 submodules, as well as E1 stream monitoring and statistics. 

Monitoring → E1 streams

For E1 chips, the table indicates the position number in which it is installed (see Submodule installation), chip name and identifier. 

Stream parameters:

• State – stream status:
  • WORK – stream is in operation; 
  • LOS – signal is lost;
  • OFF – stream is disabled in configuration; 
  • NONE – submodule is not installed; 
  • AIS – alarm state indication signal (signal that contains all ONEs); 
  • LOMF – multi-frame alarm state indication signal; 
  • RAI – remote alarm indication; 
  • TEST – stream test indication (PRBS test, local or remote loop). 
• D-channel state– state of D-channel, service management channel:
  
  • up – D-channel is in operation; 
  • down – D-channel is not in operation; 
  • no – there is no management channel for the stream; 
  • off – signalling is disabled for the stream; 
  • KPD1/KPD2 down – KPD1/KPD2 is not in operation. 
• Statistics collection time, sec – statistics collection period in seconds; 
• Slip up – number of positive bit slips for the stream; 
  
• Slip down – number of negative bit slips for the stream; 
• RX bytes – number of bytes received from the stream; 
• TX bytes – number of bytes sent to the stream; 
• Short packets – number of packets received which size is less than standard; 
• Big packets – number of packets received which size is bigger than standard; 
• RX Overflow – buffer overrun error counter; 
• CRC errors – CRC error counter; 
• TX underrun – stream transmission failure counter; 
• Сode violation counter – signal code sequence failure counter; 
• CRC Error Counter / PRBS – CRC error quantity (in 'PRBS test' mode); 
• Bit error rate – number of bit errors for the stream.

The following buttons are below the table: 

• Reset counters – when checked, click 'Reset' button to reset the collected statistics for the selected stream; 
• Remote Loop – Е1 path test mode, where signal received from the connected Е1 stream by the unit is transmitted into the same stream; 
• PRBS test – enables pseudorandom sequence output to the output port of the unit (transmitted into the connected Е1 stream); at that, error detection mode will be enabled at the unit input port (Е1 72 SMG Digital Gateway stream reception) for this sequence in order to evaluate the signal transmission quality. Number of errors and analysis time counter will be displayed in the stream information window; 
• PRBS test with Local Loop – Е1 path test mode, where external line is disabled and the signal transferred by the unit is transmitted into the input of the same unit. Pseudorandom sequence output will be enabled to the unit output port; input port will operate in the error detection mode; 
• Stop test – disable test mode. 
  

E1 channel monitoring

This section contains information on E1 stream channel status. In the upper part of the field, there is E1 stream channel matrix, where channel numbers are defined in rows and stream numbers are defined in columns (their assigned signalling protocol listed in parentheses). In the lower part of the field, there are information tables and the management table. 

Information tables

Connection information for stream # and channel #:

• Port/channel – this section is divided into two parts: 
  
  • Signalling protocol (PRI/SS7);
  • Port location Stream #: Channel #. 
• Connected port/channel – this section is divided into two parts: 
  
  • Linked port signalling protocol (PRI/SS7/VoIP);
  • Linked port location Stream: # Channel:  # for PRI/SS7 or VoIP submodule: # VoIP channel.
• Connected Callref – call identifier for linked channel; 
• State – channel state:
  • Off – channel is disabled; 
  • Block – port is blocked; 
  • Init – channel initialization; 
  • Idle – channel is in initial state;
  • In-Dial/ Out-Dial – incoming/outgoing call dialing; 
  • In-Call/ Out-Call – incoming or outgoing occupation; 
  • In-Busy/ Out-Busy – sending 'busy' tone; 
  • Talk – channel is in call state; 
  • Release – channel release; 
  • Wait-Ack – waiting for acknowledgement; 
  • Wait-CID – waiting for CgPN (Caller ID);
  • Wait-Num – waiting for call dialing; 
  • Hold – subscriber is on hold.
• State timer – channel last known state duration; 
• Incoming SS7 category – SS7 category of an incoming call before modification; 
• Incoming CdPN – callee number before modification; 
• Incoming CgPN – caller number before modification; 
• Outgoing SS7 category – SS7 category of an incoming call after modification; 
• Outgoing CdPN – callee number after modification; 
• Outgoing CgPN – caller number after modification.

Stream state – information table with matrix symbol interpretations:

• State – stream status:
  • NONE – missing С4Е1 submodule;
  • OFF – поток выключен в конфигурации;
  • ALARM – C4E1 submodule initialization error; 
  • LOS – signal is lost; 
  • AIS – alarm state indication signal (signal that contains all ONEs); 
  • LOMF – multi-frame alarm state indication signal; 
  • WORK/RAI – remote alarm indication; 
  • WORK/SLIP – SLIP indication for the stream; 
  • WORK – stream is in operation; 
  • TEST – stream test indication (PRBS test, local or remote loop). 

Channel state – information table with matrix symbol interpretation:

• State – channel status:
  • Off – channel is disabled in configuration; 
  • Idle – channel is in initial state; 
  • Block – channel is blocked; 
  • Incoming dialing – incoming call dialing; 
  • Outgoing dialing – outgoing call dialing; 
  • Incoming alerting – incoming occupation, callee is disengaged; 
  • Outgoing alerting – outgoing occupation, callee is disengaged; 
  • Busy, Release – channel release, sending 'busy' tone; 
  • Talk, Hold – channel is in call state, on hold; 
  • Waiting – waiting for response from the opposite party (waiting for occupation
    acknowledgement, waiting for Caller ID, waiting for call dialing); 
  • 3way, Conference – conference mode (3-WAY conference or conference Add-on). 

If one of the C4Е1 submodules is missing, the message "C4E1 submodule is not installed, channel monitoring is unavailable ".

Channel state updates in 5 seconds interval. 

Link management

To enable stream management, left-click the stream name. The field will become highlighted, for example, the screenshot below shows the information for Stream 1 (SS7). Next, in 'SS7 link management' table, select the field with the required action and left-click it. Pop-up informational message about the command execution will be shown on screen. 

Monitoring → E1 channels

SS7 link management – SS7 signal link management:

• Send LUN – send Link uninhibit signal;
• Send LIN – send Link inhibit signal;
• Send LFU – send Link forced uninhibit signal;
• Send congestion state – set signal link overload state; 
• Clear congestion state – cancel signal link overload state; 
• Set local processor outage; 
  
• Clear local processor outage; 
  
• Invoke normal link restart; 
  
• Invoke emergency link restart; 
  
• Stop link.

Channel management

To enable management for a channel in a stream, left-click its icon. The field will become highlighted, for example, the screenshot below shows the information for Channel 11 in Stream 0 (SS7). Next, in 'SS7 channel management' table, select the field with the required action and left-click it. Pop-up informational message about the command execution will be shown on screen. 

Monitoring → E1 channels

SS7 channel management – SS7 (CIC) channel management table:

• Block channel (send BLO) – send BLO message to block channel; 
• Unblock channel (send UBL) – send UBL message to unblock channel; 
• Reset channel (send RSC) – send RSC message; 
• Local block – block channel locally without BLO message transmission; 
• Local unblock – cancel local block; 
• Release (send REL) – send REL message; 
• Release complete (send RLC) – send RLC message; 
• Run continuous-check test (send CCR) – run continuous-check test by sending CCR message; 
• Stop continuous-check test – stop channel continuity test; 
• Show continuous-check test state – show current continuous-check test state.

CPU utilization chart

This section contains information on CPU utilization in real time (10-minute interval). Statistics charts are based on average data for each 3-second device operation interval. 

Monitoring → CPU load graph

To navigate between specific parameters in monitoring charts, use buttons and . To facilitate visual identification, all charts have different colors. 

• TOTAL – total CPU utilization percentage; 
• IO – percentage of CPU time spent on I/O operations;
• IRQ – percentage of CPU time spent on hardware interruptions' processing; 
  
• SIRQ – percentage of CPU time spent on software interruptions' processing; 
• USR – percentage of CPU time utilization by user applications; 
• SYS – percentage of CPU time utilization by core processes; 
• NIC – percentage of CPU time utilization by applications with modified priority. 

SFP module monitoring

This section contains status indication and optical line parameters. 

Monitoring → SFP modules

• SFP port Х status – optical module status: 
  
  • miniGBIC presence – indication of module installation (module is installed; module is not installed); 
  • Signal status – signal loss indication (signal lost, in operation); 
  • Temperature, °C – optical module temperature; 
  • Voltage, V – optical module power supply voltage, V; 
  • Tx bias current, mА – ток смещения при передаче, мА;
  • Input power, mW – receiving signal power, mW; 
  • Output power, mW – transmitting signal power, mW. 

Front ports monitoring

This section contains information about physical switch port state – link state, committed data rate and mode of transmission. Dual port (copper and optical connectors) is marked with ‘SFP’ label near its number. There is no label, if dual port is active and connected with copper cable. 

Monitoring → Front-ports

• Link – cable connection state on port (UP/DOWN);
• Speed – committed data rate on port;
• Duplex – data transmission mode (half-/full-duplex).
• LACP group – LACP channel including the port and its state (UP/DOWN);
• LACP state – port mode (active/backup);
• Rx bytes – storage counter of received packets, including different types of received packets;
• Tx bytes – storage counter of transmitted packets, including different types of transmitted packets.
• Fault indication¹ – when checked, the event of the global/local link loss and provided active backup interface on the relevant port, a fault message will be displayed in the header of the web interface (No WAN/LAN connection on front port x/x);
• Apply¹ – save the fault indication settings.

VoIP submodule monitoring

This section contains information on SM-VP submodules installed and their channel state. 

Monitoring → VoIP submodules

• № – SM-VP submodule sequential number;
• Type – installed submodule type;
• State:
  • Not Present – not installed; 
  • No init – not initialized, no initialization attempts; 
    
  • Off – disabled, no submodule load attempts; 
    
  • Wait Ack – waiting for acknowledgement from CPU after submodule load; 
  • Failed – no response from submodule; 
  • Work – submodule normal operation; 
    
  • Recovery – no control packets coming from submodule; 
  • Reserved – submodule is reserved for SORM needs; 
  • SSW.Sorm – submodule is used by SORM agent. 
• Active counts – number of active connections on the submodule at the given moment; 
• Payload – submodule resource utilization percentage at the given moment. 
  

For channel state monitoring, left-click the row containing the required submodule number. To hide the information, left-click the row again. 

Monitoring → VoIP submodules→ Type (M82359)

Channel info#:

• Port/channel – port/channel data: 
  
  • signaling protocol (VoIP);
  • Port location VoIP submodule #: Channel #. 
• Callref – internal call identifier; 
• Connected port/channel – linked port/channel data: 
  
  • Linked port signaling protocol (PRI/SS7/VoIP); 
  • Linked port location Stream #:Channel # for PRI/SS7 or VoIP submodule #:VoIP channel#. 
• Connected Callref – call identifier for linked channel; 
• State – channel state:
  • Off – channel is disabled; 
  • Block – channel is blocked;
  • Init – channel initialization; 
  • Idle – channel is in initial state; 
  • In-Dial/ Out-Dial – incoming/outgoing call dialing 
  • In-Call/ Out-Call – incoming or outgoing engagement; 
  • In-Busy/ Out-Busy – sending 'busy' tone; 
  • Talk – channel is in conversational state; 
  • Release – channel release; 
  • Wait-Ack – waiting for acknowledgement; 
  • Wait-CID – waiting for CgPN (Caller ID); 
  • Wait-Num – waiting for call dialing; 
  • Hold – subscriber is on hold. 
• State timer – channel last known state duration; 
• Incoming SS7 category – SS7 category of an incoming call before modification; 
• Incoming CdPN – callee number before modification; 
• Incoming CgPN – caller number before modification; 
• Outgoing SS7 category – SS7 category of an incoming call after modification; 
• Outgoing CdPN – callee number after modification; 
• Outgoing CgPN – caller number after modification. 

Channels state:                                                                       

• Idle (grey) – initial state, channel is ready to serve the call; 
• Active (green) – active state, channel is engaged with active call; 
• Reserved (yellow) – channel is reserved for service needs (sending 'busy', 'ringback', 'PBX response' tone) or for a new call. 

To view detailed channel information, left-click to select it from the table.  

Call IP info# submodule#: 

• State – channel state (see description above); 
• Codec – used codec (Payload Type is defined in square brackets); 
• Status – media information transfer status, options: 
  
  • Good – channel is in operation; 
    
  • Loss of RTP – loss of the opposite RTP stream (when 'RTP packet timeout' expires); 
    
  • VBD – communication in data transfer mode has been established through the channel; 
    
  • T38 – fax connection with Т.38 protocol has been established through the channel. 
    
• Mode – media channel operating mode: 
  
  • sendrecv – channel operates in duplex mode (reception and transmission); 
    
  • sendonly – channel operates in simplex mode, transmission only; 
    
  • recvonly – channel operates in simplex mode, reception only; 
  • inactive – channel is not active, reception and transmission are inactive;
    
• SSRC – SSRC (Synchronization Source) field value for outgoing device RTP stream; 
  
• IP:port remote – remote IP address and port of RTP;
  
• IP:port local – local IP address and port of RTP stream source; 
  
• MAC remote – remote MAC address of RTP stream source; 
  
• MAC local – local MAC address of RTP stream source. 
  

There is the 'Disconnect the call' button below the tables with channel status, which allows one to forcibly terminate the connection. 

Fault alarms. Alarm events list

When a failure occurs, related information containing the fault stream number, SS7 link set, signal link or faulty module will be displayed on the web configurator header. If there are multiple active alarms, the most critical alarm at the given moment will be shown in the web configurator header.

When there are no alarms, the message 'No alarms' will be shown. 

Table 20 — Alarm message examples

The 'Alarm events list' menu displays a list of emergency events, ranked by date, time and events. 'Only active' events show current accidents on the device in this moment. 'All events' display all available alarm information. Also there is a 'Clear' button, which deletes all information from the current log. 

Monitoring → Alarm events list → Only active

Monitoring → Alarm events list → All events

Alarm events list:

• Clear – delete the current alarm events table; 
• № – alarm sequential number; 
• Time – alarm occurrence time in HH:MM:SS format; 
• Date – alarm occurrence date in DD/MM/YY format; 
• Type – alarm type:  
  
  • CONFIG – critical fault, configuration file fault; 
  • SIPT-MODULE – critical fault, failure of a software module responsible for VoIP operation; 
  • LINKSET – critical fault, SS7 link set is not in operation; 
  • STREAM – critical fault, E1 stream is not in operation; 
  • SM-VP DEVICE – fault, SM-VP module failure; 
  • SS7LINK – SS7 signal channel failure; 
  • SYNC – synchronization fault, synchronization source is missing; 
  • STREAM-REMOTE – warning, E1 stream remote fault; 
  • CDR_UPSERVER – alarm or warning, error of sending a CDR file to a remote storage; 
  • TRUNK-CPS – permitted number of calls per second is exceeded for a trunk group; 
  • SORM-KPD – alarm, KPD1/KPD2 in not in operation; 
  • SIP-DUPLICATE – duplication failures of INVITE message received from emergency call service node; 
• State – fault state status::
  • Critical fault, flashing red icon – alarm requires immediate intervention of the service personnel, affects device operation and provisioning of communication services; 
  • Fault, red icon – non-critical alarm, also requires intervention of the service personnel; 
  • Warning, yellow icon – alarm does not affect provisioning of communication services; 
  • OK, green icon – alarm is resolved. 
• Parameters – text description of alarm details. Depending on the alarm type, may appear as follows: 
  
  • CONFIG;
  • SIPT-MODULE – no communication with SIP module; 
  • LINKSET – SS7 link set XX is not in operation, where ХХ is SS7 link set number; 
  • STREAM – E1 stream XX failure, where ХХ is stream number; 
  • SM-VP DEVICE – no communication with VoIP submodule XX, where ХХ is SM-VP submodule number; 
  • SS7LINK – SS7 link failure Linkset XX, E1 stream YY, where ХХ is SS7 link set number, YY is a signal channel number in SS7 group; 
  • TRUNK-CPS – 'XX' trunk group exceeds CPS threshold, where XX is a trunk group name; 
  • SORM-KPD – KPD1/KPD2 stream 'XX' in not in operation, where XX — E1 stream number; 
  • SIP-DUPLICATE – SIP interface 'XX'. INVITE duplication to the '<YY>' server failure, where XX — SIP interface name, on which failure was occurred; YY — duplication server address, on which failure was occurred. 

Network interface monitoring

This section allows monitoring of network interfaces (tagged/untagged/VPN) and viewing users connected to VPN device. 

Monitoring → Network interfaces

• Ethernet – Ethernet interface name;
• Network name – name that the current network settings are associated with; 
• VLAN ID – virtual network identifier (for tagged interface); 
• DHCP – DHCP usage status, allows to obtain network settings automatically (DHCP server is required in the operator network); 
• IP address, Broadcast, Network mask – interface network settings (if DHCP is not used). 

VPN/pptp interfaces

• PPP interface – name of the interface; 
• Network name – name that the current network settings are associated with; 
• PPTPD IP – PPTP server IP address used for connection; 
• Username – username identifier; 
• IP address, P-t-P, Network mask – interface network settings. 

Local disk drives

This section contains information on the connected storage media. 

• Eject – click this link to safely remove the storage device. 

Monitoring → Local disk drive

The names of external drives are linked to the interface ports: 

V5.2 interfaces

The state of V5.2 interfaces is displayed in this section¹.

• Red — the interface is out of the operation; 
• Green — the interface is on operation. 

Queue statistics

This section displays queue operation statistics. 

• ID queue — queue identifier;

• Total calls — total number of calls received in the queue;

• Answered — number of successful calls ending with an operator response;

• Unaswered — number of calls in which the caller hung up without waiting an operator response;

• Maximum queue length (hour/day/workday) — maximum queue length per last hour/day/working day. Last hour/day is a periodic time interval, repeating every hour/24 hours respectively, the beginning of the first interval is necessary to count from the moment the software starts. Time intervals of the working day are set in the group settings call; 
  

• Callback failure — number of unsuccessful attempts to call back the subscriber, when using the callback option²;

• Queue overflow — number of calls rejected due to queue overflow; 
  

• Waiting time — average waiting time for an operator response, a response is generated based on this value. To clear queue statistics, check ‘Select’ opposite those queues whose statistics need to be cleared, and click the appeared ‘Clear selected’ button. 
  

  

VNS tasks

Monitoring → VNS tasks

This section displays the status of running voice notification systems¹.

• Task name – VNS task name;
• State – displays the state of a running task for an alert: 
  
  • Waiting;
  • Reserved;
  • Prepared;
  • Launched;
  • Error;
  • Requires completion;
  • Stopped;
  • Completed.
• Start time – time to start the notification task in the format Hours:Minutes:Seconds Day.Month.Year; 
• Percent done – task procent done (ratio of processed calls number to all calls in this task); 
• Idle – number of inactive calls in a task. Example: 30(40) – 30 from 40 (total numbers in the task); 
• Active – number of active calls in a task. Example: 15(40) – 15 from 40 (total numbers in the task);
• Failed – number of unsuccessful calls in a task. Example: 5(40) – 5 from 40 (total numbers in the task); 
• Done – number of completed calls in the task. Example: 35(40) – 35 из 40 (total numbers in the task); 
• Stop – force completion of a calling task. 

Domains monitoring

This section displays a list mapping domain names to their IP addresses (local DNS cache).

Monitoring → Domains monitoring

The SMG Domain Name System is designed such that, upon the first request to the DNS server, the mapping between the domain name and its IP address is stored in the local DNS cache. To speed up the processing of subsequent queries for this domain name, SMG uses the existing local cache. The duration for which records are stored in the cache corresponds to the TTL parameter received from the DNS server.

There is no time limit on the storage of records from the ‘Domain Name List’ table; the TTL for these records is always zero.

SS7 stack monitoring

This section¹ displays the status of the netlinks configured on the device.

See more about netlink configuration in Appendix M. SS7 stack monitoring.

Monitoring → SS7 stack monitoring

Е1 streams

The signalling and parameters of each E1 stream are configured in this section. 

Synchronization sources

To synchronize the device with multiple sources, priority list algorithm has been implemented. Its meaning is as follows: when sync signal from the current source is lost, the list lookup is performed to identify active signals from the lower priority sources. When the higher priority signal is restored, the system will switch to that signal. Also, you may use multiple sources of the same priority; at that, when the same priority signal is restored, the system will not switch to that signal.

Up to 16 synchronization sources can be specified (each of 16 E1 streams and 2 external sources).

The ports receiving external signals have the impedance of 120 Ohm. The incoming signal should have the parameters given in ITU-T G.703 recommendation, section 15, 2048 kHz syncronization interface (T12). 

E1 streams → Synchronization sources

To generate the list, use the following buttons:

 – «Add source»; 

  – «Remove».

To change the source priority, use "Up"/"Down" buttons located next to each source. The highest priority value is 0, the lowest priority value is 15.

• Signal loss timeout, sec – time interval during which the system does not switch to a lower priority synchronization source when the signal is lost. If the signal is restored during this interval, there will be no switching; 
• Signal presece timeout, sec – time interval during which the signal restored from a higher priority
  synchronization source should be active before the system switches to that signal. 

Signaling protocol selection

The signaling protocol used on the stream is selected in the drop-down list ‘Signaling protocol’. 

E1 stream → Stream 0 (Q.931-U) → Physical settings/Q.931

The device supports the following signals:

• Q.931 (User);
• Q.931 (Network);
• SS7;
• SORM;
• SORM-TRANSIT;
• V5.2 (LE);
• M2UA²;
• IUA (User)¹ ;
• IUA (Network)¹.

V5.2 interfaces:

• Media Gateway¹.

Physical settings

Physical settings:

• Title – Е1 stream name;
• Enable – enable the stream;
• Framing:
  • doubleframe – CRC4 is disabled;
  • CRC multiframe – CRC4 checksum generation at transmission and control at the reception. 
    
• Equalizer – when checked, transmitted signal is amplified; 
• Alarm indication – when checked, in case of local alarm an alarm indication will be on the stream (the ALARM indicator will light up, the accident will be recorded in the alarm events list); 
• Remote Alarm indication – when checked, in case of remote alarm an alarm indication will be on the stream (the ALARM indicator will light up, the accident will be recorded in the alarm events list); 
• Line code – type of information encoding in the channel (HDB3, AMI); 
• Slip indication – when checked, in case of slip detection in the receiving path, an alarm indication will take place; 
• Slip detection timeout – the frequency of polling the flow parameters of the board, if the slip is detected in the stream, then during this timeout the gateway will indicate an alarm. 

E1 streams → Stream 0 (Q.931-U) → Physical settings/Q.931

Signaling protocol settings DSS1 (ISDN PRI Q.931)

"Physical settings/Q.931" tab

E1 streams → Stream 0 (Q.931-U) → Physical settings/Q.931

Q.931 LAPD – LAPD channel level settings of Q.931 protocol

• Т200, х100 ms – transmission timer. This timer defines time period for frame response reception that will enable the following frames' transmission. This time period should be greater than the time required for frame transmission and its acknowledgement reception; 
• Т203, х100 ms – maximum time during which the device may not exchange frames with the remote device; 
• N200 – quantity of frame retransmission attempts. 

Q.931 settings

• Trunk group – name of a trunk group, that includes the Е1 stream;
• PRI profile – selects a PRI profile for servicing PRI subscribers;
• Scheduled routing profile – selects scheduled routing profile from the list of existing profiles;
• Access category – selects access category;
• Dial plan – defines dial plan that will be used for routing of the call received from this port (necessary for dial plan negotiation);
• Numbering plan type – defines ISDN dial plan type. To use common dial plan E.164, select 'ISDN/telephony';
• Calling party category – Caller ID category assigned to calls received from this port;

• Send calling part category – enables Caller ID category transmission as the first digit of a number in CgPN information element of the SETUP message.

  For proper operation, support for this mode on the opposite side is necessary.

  
  

• ‘End of dial’ message – produces 'Sending Complete' informational element upon 'End of dial' event (such event arrives from the linked channel side, achieved maximum quantity of digits according to prefix, dialing timeout for the next digit);
• Do not send RESTART for interface – when checked, gateway will not send RESTART message into the line when the stream is restored (channel level LAPD is established);
• Do not send RESTART for channel – when checked, gateway will not send RESTART message upon the expiration of T308 timer. This timer activates when RELEASE message is sent into the channel and resets when it receives RELEASE COMPLETE message as a response. If RELEASE COMPLETE message is not received during T308 timer active state, RESTART message is transmitted in order to release the channel;
• Channels selection order – defines the order of the physical channel provisioning when performing outgoing call. You may select one of four types: sequential forward, sequential back, from the first and forward, from the last and back. To minimize conflicts during communication with neighboring PBXes, we recommend to set inverse channel engagement types;
• DialTone for incoming overlap-seize – when checked, gateway will send DialTone into the line during incoming overlap seize ('PBX response' ready signal). In this case, overlap seize is a reception of SETUP message without 'sending complete' indication;
• Process PI ‘In-Band’ in DISCONNECT – when checked, field PI In-Band contained in DISCONNECT message will be processed for call release voice message transmission, otherwise this field is ignored;
• Handle PROCEEDING as ALERTING – when checked, upon receiving a PROCEEDING message, it will be processed as an ALERTING and a RBT will be issued;
• Process PI in SETUP – when checked, adds the ability to change the Progress Indicator in a SETUP message. It is possible to change to:
  • Transit – transmit without change;
  • 1 – Not end-to-end ISDN;
  • 2 – Dest addr is non ISDN;
  • 3 – Orig addr is non ISDN;
  • 4 – Return to ISDN;
  • 5 – Interworking occurred;
  • 8 – In-band information.

• Replace symbol ‘?’ by ‘D’ in CgPN – when checked, if a received SETUP message in CgPN receives a ‘?’, it will be replaced by ‘D’.

• ISUP Location Number transit – when checked, if the Location Number parameter is passed in the incoming message SS7/SIPT, it will be transferred to the Calling Party Number parameter in the outgoing message SETUP Q.931.
• Send Redirection to FACILITY — enable the use of the facility in the event of call forwarding:
  • if the option is active, then upon receiving a FACILITY with CheckRestrictions, we parse it and forward it to subscriber A (Diversion for SIP, CheckRestrictions for 931, provided the option is also active on the outgoing leg);
  • when forwarding on the device, FACILITY with CheckRestrictions and number C to subscriber A is generated;
  • when forwarding on the device, Facility with ActivateDiversion and number B to subscriber C is added in the SETUP;
  • when receiving a SETUP, if there is no Redirection, we parse the Facility with ActivateDiversion and store it for further use in the call.

'Calling name translation settings' tab

This tab is used to configure the way of name reception/transmission and coding of received/transmitted name.

E1 streams → Stream 0 (Q.931-U) → Calling name and translation settings

• Name transmission:
  • Not set – name transmission is disabled;
  • Q.931 DISPLAY – transmission in Q.931 Display element with Codeset 5;
  • QSIG-NA – transmission via QSIG-NA (ECMA-164) protocol;
  • CORNET – transmission via Siemens CorNet protocol;
  • CORNET HICOM-350 – transmission via Siemens CorNet protocol with additional info for Hicom PBX;
  • AVAYA DISPLAY – transmission in Q.931 Display element with Codeset 6;
  • QSIG-NA (Ericsson) – transmission in facility and user-user information.
• Name coding:
  • Transit – no recoding is carried out (by default the name is assumed to be accepted in UTF-8);
  • CP 1251 – coding of Windows-1251;
  • Siemens adaptation – coding of Siemens PBX;
  • AVAYA adaptation – coding of AVAYA PBX;
  • Latin transliteration – Russian names will be transliterated into Latin letters.
• Straight direction only – send subscriber name only in forward direction messages.

The method selected for name reception/transmission and coding of received/transmitted name works only in a configurable E1 stream. Transmission between streams differing by the settings of name transmission parameters is possible. In case of such transmission, the SMG performs recoding by itself to harmonize the sides.

'Channel settings' tab

E1 streams → Stream 0 (Q.931-U) → Channel settings

This menu is used to enable/disable E1 stream channel. To do that, select/clear checkbox against the corresponding channel. ‘Trunk group’ column displays number of group where these channels are configured (used only when trunk group is assigned to channels, not to the whole stream).

In the ‘Access Category’ column, it is possible to select the required access category for each E1 stream channel individually.

SS7 signalling protocol configuration

Physical settings/ss7

E1 streams → Stream 0 (SS7) → Physical settings/SS7

SS-7 settings:

• SS-7 Linkset – linkset selection (SS7 linkset);
• Channel ID (SLC) – signal line identifier in SS7 linkset;
• DPC-MTP3 – destination point code of the signaling transition point (STP). It is used during SMG operation in quasi-associated mode. If quasi-associated mode is not required, set value 0. At that, MTP3 opposite code is equal to DPC-ISUP value defined in configuration (see Linkset);

• D-channel – number of the channel interval that will be used for signaling transmission;

  Move to 'Channel settings' tab after changing the number of D channel on a stream with SS7 and set the appropriate CIC for the same channel timeslot that you are already set for D channel.

  
  

• Bit D in LSU – set value 1 for bit D in status field (SF) of a signal unit LSSU (D–F bits in status field SF are reserved).

'Channel settings' tab

E1 streams → Stream 0 (SS7) → Channel settings

• ISUP CIC – channel identifier code – setting conversation channel numbers (CIC).

To automatically number conversation channels, click the ‘Set’ button.

E1 streams → Stream 0 (SS7) → Channel settings → Set

The following menu will be displayed:

• Starting value – number of the first conversation channel;
• Numbering step – channel numbeing step. Each subsequent channel will be assigned a number with “numbering step” larger relative to the previous channel;
• Last value – number of the last conversation channel in the selected range;
• Channels range – selecting values in this block allows one to assign numbering for all stream channels or for a specified range of channels.

V5.2 signalling protocol configuration

The assignment of a stream to the V5.2 interface is made in the V5.2 interface parameters.

This section displays the current V5.2 interface to which this stream is assigned for reference, as well as the stream identifier inside the V5.2 interface.

E1 streams → Stream 0 (SS7) → Physical settings/V5.2

SORM signalling protocol configuration

E1 streams → Stream 0 (SS7) → Physical settings/SORM

• Enable command-avating timer (10 min) – enable/disable timeout for receiving commands from the SORM control panel (реализовано согласно пункту 1.5 Приказа №70 Госкомсвязи России от 20.04.1999);
• Activity control – control of message exchange activity at the L1 level, if within 15 seconds, no packets were received on at least one of the channels, a reset will occur and re-initialization of the E1 stream framer;
• No VAS-number prefix – when ordering VAS by the subscriber, VAS-number prefix will be not transmitted to the SORM remote control. For example, when ordering the ‘Unconditional forwarding’ service and dialing the number *21*2728331# to the SORM remote control, the message 44 will have the number 2728331 only, which forwarding is assigned to;
• No extended error codes – when checked, in response to a command with incorrect parameters, the message will be sent about non-acceptance or non-compliance commands only with the characteristics defined in order No. 268. Otherwise the command non-compliance signs of the manufacturer will be used, allowing to more accurate determine the reason for the command failure. List of standard and manufacturer codes is given in Appendix D;
• No operator-selection code – when monitoring a subscriber, the prefix for selecting a telecom operator for long-distance or international calls is not taken into account (more details in Appendix D);
• Control by Redirecting number – use the number from the Redirecting number field (or diversion in the SIP protocol) for transmission to the control panel. Upon receiving a call with a Redirecting number (or diversion in the SIP protocol) the number from the Calling Party Number field is initially compared with the numbers on the control, then, if a match is not found, with the number from Redirecting number fields (or diversion in the SIP protocol). When unchecked, the comparison with Redirecting number (or diversion in the SIP protocol) is not performed;
  • No – there is no comparison with the redirecting number (or ‘diversion’ in the SIP protocol);
  • Ordinary call – if the number displayed on the control panel matches the redirected number, the information is sent to the control unit as in a normal call;
  • Redirected call – If the number on the control panel matches the redirected number on the control unit, information regarding the call and the use of the VAS call forwarding feature is sent (message 44).
• Skip msg 1.1 for incomplete number;
• Station type – communication node type transmitted in the last byte of message 11 (station firmware version);
• Protocol – selection of the SORM specification according to which the device will operate:
  • RUS Order 70 – SORM specification for the order of the State Committee for Communications of Russia dated April 20, 1999 No. 70;
  • RUS Order 268 – SORM specification for the order of the Ministry of Telecom and Mass Communications of Russia dated November 19, 2012 No. 268;
  • KZ – SORM specification for the Republic of Kazakhstan.

• Connection mode:

  • X25 – signal channels are organized via the X25 protocol, using 30-31 channels of E1 stream;

  • TCP – signal channels are organized via the TCP protocol. The settings are active only when selecting ‘TCP connection mode’):

    • Port 1 – virtual TCP port to organize the signal channel of command and control center 1;

    • Port 2 – virtual TCP port to organize the signal channel of command and control center 2;

    • Network interface – selecting the device network interface.

Channel operation mode:

• Channel 1 – block for setting parameters of the control information transmission channel from the SORM control panel;
• Channel 2 – block for setting parameters of the channel for transmitting information about the controlled connections from SMG-1016M.

Channel settings:

• Channel mode:
  • DTE – when checked, the device type is DTE (information transmitter);
  • DCE – when checked, the device type is DCE (receives the data from DTE devices).
• Send SABM – when checked, a message about the beginning of the connection initialization procedure is transmitted to the channel;
• Send RESTART (L3) – transmission of ‘level 3 restart’ message when establishing connections with SORM control panel;
• Send INITIAL_RESET (L3) – transmission of a ‘level 3 reset’ message when establishing connections with SORM control panel.

Frames addresses:

• TxCmd Addr – command frame address;
• TxResp Addr – response frame address.

• DTE/DCE mode adjustment – option to automatically adjust DTE/DCE mode, by default: enabled. If the device and the remote side are set to the same mode (DTE-DTE or DCE/DCE) and the adjustment option is enabled, the SMG will automatically change the mode to the correct one.

Modifiers of incoming numbers – selecting a table of modifiers intended for analysis and modification of the subscriber's telephone number in SORM messages received from the console.

Modifiers of outgoing number – selecting a table of modifiers intended for analysis and modification of the subscriber's phone number in SORM messages sent to the remote control.

Always modify B-number – an option required to modify all B-numbers, the outgoing number modifier has been not previously applied for the number dialed by the local subscriber.

Modifiers of controlled numbers – selecting a table of modifiers intended for analysis and modification of the subscriber's phone number before selecting it for sending to the SORM control panel.

Dial plans

This section is used to configure the device dial plan.

The device has up to 16 independent dial plans (up to 255 for SMG-2016 and SMG-3016 with VAS license). Each dial plan can have its own subscribers and prefixes.

The number of active plans is configured in the System settings.

There are 4 criteria by which calls are routed on the device:

• search by calling party number – CgPN (Calling Party Number);
• search by called party number – CdPN (Called Party Number);

• search by calling number – CgPN (Calling Party Number) and called party – CdPN(Called Party Number);

• search in the database of subscribers configured on the device.

Upon entering a call into the dial plan, its routing begins, initially search for a match with CgPN number masks takes place. If there is a prefix with ‘AND’ logic (masks are specified by CgPN and CdPN, and a match was found for both parameters) and a prefix with the same mask is found according to CgPN, then if the “Priority” parameter is equal, the call will go through the prefix with the ‘AND’ logic, because this mask is considered to be more accurate. If the priority of the prefix with ‘AND’ is lower, then the call will go by prefix with ‘OR’.

If, when searching by CgPN, two prefixes with ‘AND’ logic are found, and the CgPN mask is the same, then CdPN is compared and the call is routed using a prefix with a more accurate mask.

Далее происходит поиск по базе сконфигурированных на устройстве абонентов. В случае нахождения совпадения по префиксу с маской CgPN или по базе абонентов происходит маршрутизация вызова и дальнейший поиск прекращается.

Searching and routing a call through the database of configured subscribers is carried out even when the call parameters match the CgPN number masks.

If the call parameters do not match the CgPN masks and the subscriber number, a search occurs across all CdPN masks configured in the dial plan.

Dial plans → Dial plan # 0 'NumberPlan#0'

Dial plan settings:

• Name – dial plan name.

Check dial plan by number – checks if routing is possible for the number entered into this field.

The check is performed by callee and caller masks and thorugh the configured SIP subscriber database.

• ST – when checked, the search recognizes the end dial marker.

Search mask – prefix search by number pattern, name, direction, prefix type, trunk direction, trunk group.

The check provides information on routing capability for this number:

• calling-table – routing by the calling table;
• called-table – routing by the called table;
• NOT found in – routing by this table is not possible;
• found in – routing by this table is possible;
• Abonent 'SIP' idx[4] – SIP subscriber [entry number for this subscriber in the database];
• Prefix [6] – routing by a prefix [prefix number in the list].

Copying prefixes to another dial plan:

• Copy all prefixes to the dial plan – this option allows copying the selected prefixes to another dial plan. It is used similarly to copying dedicated prefixes, but does not require prefix selection;
• Copy selected prefixes to the dial plan – this option appears when prefix is selected in the table. It allows copying selected prefixes to another dial plan. For use, select prefixes, target dial plan and click ‘Copy’.

  

Creating a prefix in the dial plan

To create a new prefix, open the ‘Objects’ menu and click ‘Add an object’ or click the button located below the list, and enter prefix parameters in the opened form:

Dial plans→ Dial plan # 0 'NumberPlan#0' →  

Common prefix settings:

• Title – prefix name;
• Dial plan – selects a dial plan;
• Access category – selects an access category;
• Check access category – when this option is selected, it checks the possibility of call routing by the prefix based on the rules determined by access categories;
• Prefix type – selects the prefix type: 
  • TrunkGroup – transition to a trunk group;
  • Trunk Direction –  transition to a trunk direction;
  • Change dial plan –  this option allows you to enter another dial plan when this prefix is dialed. When this prefix type is selected, the New Dial plan option becomes available, where you should specify the dial plan for transition;
  • Subscribers pool – enables setting the subscriber capacity of the device. If the number is present in the subscriber capacity but not yet assigned to any subscriber, then a call to such a number will trigger a call release message with the cause code: 1 – Unallocated (unassigned) number;
  • VAS prefix – used to manage VAS services from the telephone set;
  • Pickup group – used to configure the pickup group transition prefix;
  • IVR scenario – used to configure the IVR script transition prefix.
  • VNS¹ – if dialled, VNS task is activated.

Parameters of the ‘Trunk Group and Trunk Direction’ Prefix:

Common Prefix Parameters:

• Trunk group – a trunk group to which the call will be routed by this prefix;
• Direction – a trunk group access type: local, emergency, zone, department, national, international. The prefix is used when enabling SORM function in the network, as well as to restrict a connection if a failure occurs during the data exchange with the RADIUS server (see section RADIUS configuration);
• Caller ID request – indicates the need for caller ID information (caller number and category) to access the trunk group specified in the “Trunk group” field. Upon receiving a call from an interacting node and the absence of Caller ID information in this call, a Caller ID request will be sent to the node (INR message via SS7 signaling);
• Caller ID mandatory –  indicates that Caller ID information is required when accessing the direction. If Caller ID information cannot be obtained from the calling party, then connection establishment process is interrupted;
• Dial mode – a method of number transmission:
  • enblock – after collection of all address information;
  • overlap – without waiting for collection of all address information.
• Do not send end-of-dial (ST) – when this option is active, the end dial marker is not sent (ST in SS or sending complete in PRI);
• Priority – if there are some overlapping masks in the dial plan, the call will be made into the prefix with a higher priority. The value of 0 is the highest priority, 100 is the lowest priority;
• Max session time (sec) – limit duration of calls passed through this prefix;
• Session warning time (sec) – activates when using the option ‘Max session time (sec)’, an audible signal is issued, which warns about the end of the call for a specified number of seconds before the end of the call. If the specified time is more than 60 seconds, an additional warning signal will sound 5 seconds before the end of the call. If the specified time is less than 60 seconds, there will be no additional signal;

• Logical operator:

  – OR – if CgPN and CdPN masks are on the prefix, there is no simultaneous analysis by CgPN and CdPN numbers;

  – AND – simultaneous analysis by CgPN and CdPN number is performed.

  For correct operation of prefixes with the logical operator ‘AND’, it is necessary to configure a mask for CgPN and CdPN. If one of the masks is missing, the prefix does not work.

  

CdPN Settings:

• Number type – a callee number type: unknown, subscriber number, national number, international number,network specific, не изменять. The selected number type will be sent in SS7, ISDN PRI, SIP-I/T signaling messages during an outgoing call by a prefix (‘no change’ means that the number type will not be converted, i. e. it will be sent in the form it has been received from the incoming channel);
• Numbering plan type – a callee dial plan type; it may take the following values: unknown, isdn/telephony, national, privat, не изменять. The selected dial plan type will be sent in IDSN PRI signaling messages during an outgoing call by a prefix (‘no change’ means that the number type will not be converted, i. e. it will be sent in the form it has been received from the incoming channel);
• Skip first digits – number of digits removed from the callee number, starting from the first.

Direct route timers (used when trunk groups are directly connected without prefix mask analysis – the Direct Prefix function in trunk group settings).

These timers work only when dialling in the overlap mode:

• Short timer – time interval in seconds when the digital gateway waits for further dialing if a part of address information has already been received. Default value: 5 seconds;
• Duration – a timer for number dialing duration. Default value: 30 seconds.

Parameters of the ‘Change dial plan’ Prefix:

• New dial plan – a dial plan to which a call will be transferred;
• New access category – a category assigned to the caller after switching to another dial plan;

• Priority – if there are some overlapping masks in the dial plan, the call will be made into the prefix with a higher priority. The value of 0 is the highest priority, 100 is the lowest priority;

• Max session time (sec) – limit duration of calls passed through this prefix;

• Sesssion warning time (sec)  – activates when using the option ‘Max session time (sec)’, an audible signal is issued, which warns about the end of the call for a specified number of seconds before the end of the call. If the specified time is more than 60 seconds, an additional warning signal will sound 5 seconds before the end of the call. If the specified time is less than 60 seconds, there will be no additional signal;
• Logical operator:

  – OR – if CgPN and CdPN masks are on the prefix, there is no simultaneous analysis by CgPN and CdPN numbers;

  – AND – simultaneous analysis by CgPN and CdPN number is performed.

  For correct operation of prefixes with the logical operator ‘AND’, it is necessary to configure a mask for CgPN and CdPN. If one of the masks is missing, the prefix does not work.

  

Modifiers when changing the dial plan: 

• CdPN modifiers – intended for modifications based on the analysis of the called number;
• CgPN modifiers – intended for modifications based on the analysis of the calling number.

Parameters of the ‘VAS Prefix’

Number masks for VAS prefix always should be ended with # symbol.

• VAS type – selecting the Supplementary Service type to manage it from the subscriber's telephone:
  • CFU — Call Forwarding Unconditional;
  • CFB — Call Forwarding Busy;
  • CFNR — Call Forwarding No Reply;
  • CFOOS — Call Forwarding Out of Service;
  • Call forward time;
  • Call pickup — call pickup;
  • Conference — conference call;
  • Clear All — canceling all services;
  • Intercom — intercom call (with an automatic answer from party B);
  • Paging — similar to Intercom, but with a call to conference numbers;
  • Password — setting a password;
  • Password access — password activation;
  • Password use;
  • Outgoing calls restriction;
  • Follow me;
  • Follow me no response;
  • DND;
  • Blacklist;
  • Call park set;
  • Call park get;
  • Voice mail local;
  • Voice mail remote;
  • Voice message recording;
  • Anonymous call;
  • Reject anonymous call;
  • Reminder;
  • Call Waiting;
  • Do not disturb in call group;
  • Autoredial;
  • Autoredial with callback;
  • VNS.
• Action – selecting an action for the service:
  • Configure — enabling a Supplementary Service;
  • Cancel — canceling a Supplementary Service;
  • Control — a Supplementary Service activity control;
  • numberAdd— add a number;
  • numberDel — delete a number.

Parameters of the ‘Pickup Group’ Prefix

• Pickup group – a pickup group in which a call pickup is performed when this prefix is dialed. If ‘Any’ is chosen, pickup will be enabled for all groups;
• CallerID request – defining the Caller ID information necessity (caller number and category) for transition to the trunk group specified in ‘Trunk group’ field. When a call arrives from the communication node and the Caller ID information is missing in that call, Caller ID request will be directed to that node (INR message from SS7 signaling);
• CallerID mandatory – indicating that Caller ID information is mandatory during the direction transition. If Caller ID information cannot be received from the calling party, connection establishment process is interrupted;
• Priority –  configuring prefix priority in the range from 0 to 100. Prefix which parameter value is lower has a greater priority (0 — the highest priority, 100 — the lowest priority);
• Max session time (sec) – limit duration of calls passed through this prefix;
• Session warning time (sec)  – activates when using the option ‘Max session time (sec)’, an audible signal is issued, which warns about the end of the call for a specified number of seconds before the end of the call. If the specified time is more than 60 seconds, an additional warning signal will sound 5 seconds before the end of the call. If the specified time is less than 60 seconds, there will be no additional signal;
• Logical operator:

  – OR – if CgPN and CdPN masks are on the prefix, there is no simultaneous analysis by CgPN and CdPN numbers;  

  – AND – simultaneous analysis by CgPN and CdPN number is performed.

  For correct operation of prefixes with the logical operator ‘AND’, it is necessary to configure a mask for CgPN and CdPN. If one of the masks is missing, the prefix does not work.

Direct route timers

• Short timer – time in seconds that the digital gateway will wait for further dialing, if the already dialed number matches any pattern in the numbering plan, but there is opportunity to obtain more digits, which will lead to a match with another pattern. Default value: 5 seconds;
• Duration – dialing duration timer. Default value: 30 seconds.

Parameters of the ‘IVR Scenario’ Prefix

• IVR scenario – an IVR scenario to which a call will be routed to on the basis of this prefix;
• CallerID request – indicates the need for caller ID information (caller number and category subscriber). When a call comes from a collaborating node and there is no Caller ID information, a caller ID request will be sent to the node (INR message via SS7 signaling);
• CallerID mandatory – indicates that Caller ID information is mandatory when accessing the direction. If Caller ID information cannot be obtained from the calling party, then connection establishment process is interrupted;
• Priority – configuring prefix priority in the range from 0 to 100. Prefix which parameter value is lower has a greater priority (0 —the highest priority, 100 —the lowest priority);
• Max session time (sec) – limit duration of calls passed through this prefix;
• Session warning time (sec) – activates when using the option ‘Max session time (sec)’, an audible signal is issued, which warns about the end of the call for a specified number of seconds before the end of the call. If the specified time is more than 60 seconds, an additional warning signal will sound 5 seconds before the end of the call. If the specified time is less than 60 seconds, there will be no additional signal;
• Logical operator:

  – OR – if CgPN and CdPN masks are on the prefix, there is no simultaneous analysis by CgPN and CdPN numbers;

  – AND –  simultaneous analysis by CgPN and CdPN number is performed.

  For correct operation of prefixes with the logical operator ‘AND’, it is necessary to configure a mask for CgPN and CdPN. If one of the masks is missing, the prefix does not work. 

Direct route timers:

• Short timer – time in seconds that the digital gateway will wait for further dialing, if the already dialed number matches any pattern in the numbering plan, but there is opportunity to obtain more digits, which will lead to a match with another pattern. Default value: 5 seconds;
• Duration — dialing duration timer. Default value: 30 seconds.
• Duration – dialing duration timer. Default value: 30 seconds.

Common prefix settings:

• VNS – VNS task triggered when an incoming call is routed to this prefix;

• CallerID request – indicate the need for Caller ID information (the number and category of the calling subscriber). When a call is received from an interconnecting node and the call lacks Caller ID information, a Caller ID request will be sent to the node (an INR message via the SS7 signalling system);
• CallerID mandatory – indicate that caller ID information is mandatory when establishing a connection. If caller ID information cannot be obtained from the calling party, the connection establishment process is terminated;
• Priority – set the prefix priority within a range of 0 to 100. A prefix with a lower value for this parameter has a higher priority (0 – highest priority, 100 – lowest priority);
• Max session time (sec) – limit the duration of calls made using this prefix;
• Session warning time (sec) – when the ‘Call duration limit (secs)’ option is enabled, an audible signal is emitted to warn that the call will end a specified number of seconds before the conversation concludes. If the specified time is more than 60 seconds, an additional warning tone will sound 5 seconds before the end of the call. If the specified time is less than 60 seconds, no additional tone will be played; 

• Logical operator:

  – OR – if CgPN and CdPN masks are on the prefix, there is no simultaneous analysis by CgPN and CdPN numbers;

  – AND – simultaneous analysis by CgPN and CdPN number is performed.

  For correct operation of prefixes with the logical operator ‘AND’, it is necessary to configure a mask for CgPN and CdPN. If one of the masks is missing, the prefix does not work.

Direct route timers:

• Short timer – time in seconds that the digital gateway will wait for further dialing, if the already dialed number matches any pattern in the numbering plan, but there is opportunity to obtain more digits, which will lead to a match with another pattern. Default value: 5 seconds;
• Duration – dialing duration timer. Default value: 30 seconds.

Masks lists:

For created dial plans, the ‘Masks List’ section allows configuring the masks of numbers for routing by this prefix.

Dial plans → Dial plan # 0 'NumberPlan#0' → Object

To generate the list, use the following buttons:

 – Add mask; 

– Edit mask;

– Remove mask;

– View mask.

Using green arrows to the left of the created mask, the entries can be moved in the table by prioritizing them.

Dial plans → Dial plan # 0 'NumberPlan#0' → Object →

• Mask – a template or a set of templates, which is compared to the calling or called number received from the incoming channel. It is used for further call routing (for mask syntax, see section Description of Number Mask and Its Syntax); 
• Type – mask type. Defines the number for the call routing – caller number (calling) or callee number (called);
• Long timer – the time interval in seconds when the digital gateway will wait for the next digit dialling until a match to a sample from the dial plan is established. Default value: 10 seconds;
• Short timer – the time interval in seconds when the digital gateway will wait for further dialling if the dialed number already matches a sample in the dial plan, but additional digits may be also dialed, which will result in a match to another sample. Default value: 5 seconds;
• Duration – the timer for number dialling duration. Default value: 30 seconds.

To edit a prefix, double-click the prefix row in the prefix table with the left button or select the prefix and click thebutton below the list.

To delete a prefix, select the prefix and click the button below the list or open the ‘Objects’ menu and select ‘Remove Object’.

Description of Number Mask and Its Syntax

Number mask is a set of templ (templates) delimited by the special character ‘|’. The mask should be enclosed into parentheses. (templ) is equal to (templ1|templ2|...|templN).

Syntax:

• X or х – any sign of the followings "0-9*#";
• * – an asterisk *;
• # – a pound key #;
• 0-9 – digits from 0 to 9;
• D – character D;
• .  – the special symbol ‘dot’ means that the preceding character may be repeated any number of times (30 characters max. for one number), e. g.:
  • (34x.) – all possible number combinations that begin with “34”.
• [ ] – defines a range (with a hyphen) or an enumeration (w/o spaces, commas, and other characters between the digits) of prefixes, e. g.:
  

  • the range ([1-5]ХХХ) –  all 4-digit numbers that begin with 1,2,3,4 or 5;

  • the enumeration ([138]xx) – all 3-digit numbers that begin with 1,3 or 8).

• {min, max} – defines the number of repetitions for the character outside the parentheses, e. g.:
  • (1x{3,5}) – means that there may be from 3 to 5 arbitrary digits (х) and it corresponds to the mask (1ххх|1хххх|1ххххх); 
• | – vertical bar. Logical OR – separates templates in a mask;
• ! – exclamation mark. When used before a template, it indicates a negation, that is a mismatch between the number and the template, for example;
  
  • (1xx|!123) – means that the number ‘123’ is excluded when any 3-digit number beginning with ‘1’ is added to the numbering plan.

• (-) – the mask used only in CgPN number modifier tables for calls without caller number. Allows the caller number to be added if it was missing and also specifies indicators for that number.

  Negation excludes only the specified pattern; it does not allow all numbers except the one specified. Negation only works in conjunction with other patterns.

  
  

Mask Operation Examples

Example 1.

(#XX#|*#XX#|*XX*X.#|112|011|0[1-4]|6[2-9]ХХХ|5[24]XXXXX|810X{11, 15})

The mask contains 9 templates:

1. #XX# – dialling a 4-character number that begins and ends with #; the 2nd and the 3rd digits of the number may take any values from 0 to 9, as well as * and #. In general, this template disables VAS using a phone set.
2. *#XX# – dialling a 5-character number that begins with *# and ends with #, the 3rd and the 4th digits of the number may take any values from 0 to 9, as well as * and #. In general, this template is used to control VAS from the phone set.
3. *XX*X.# – dialling an N-character number which begins with * followed by two arbitrary characters of the number (digits from 0 to 9, as well as * and # characters), then followed by *, and then by any number of characters (digits from 0 to 9, or *) until # is met. In general, this template is used to order VAS using a phone set.
4. 112 – dialling the specific 3-digit number (112).
5. 011 – dialling the specific 3-digit number (011).
6. 0[1-4] – a 2-digit number that begins with 0 and ends with 1, 2, 3, or 4, i. e. 01, 02, 03, or 04.
7. 6[2-9]ХХХ – a 5-digit number that begins with 6, with the second digit of the number being any digit from 2 to 9, and the last three digits being any digits from 0 to 9, as well as * and #.
8. 5[24]XXXXX – a 7-digit number that begins with 5, with the second digit of the number being 2 or 4, and the last five digits being any digits from 0 to 9, as well as * and #.
9. 810X{11, 15} – a number that begins with 810 followed by 11 to 15 arbitrary digits from 0 to 9, as well as * and #. Taking into account the first three digits, the length of the number according to this rule is from 14 to 18 digits.

Example 2.

A dial plan configuration is required to allow all numbers that begin with 1 and have the length of 3, to be routed to Trunk0, and number 117 to be individually routed to Trunk1.

To solve this task, configure the following prefixes:

1. the first prefix with the mask (117) to Trunk1;
2. the second prefix with the mask (11[0-689]|1[02-9]x) to Trunk0.

Templates of the second prefix overlap all “1xx” numbers except for 117.

Example 3.

It is required to configure a dial plan by deleting a few numbers from the group. Number group: 2340000-2349999, excluded numbers: 2341111, 2341112, 2341113, 2341114, 2341115, 2341234.

Such mask is set as follows: (234xxxx|!234111[1-5]|!2341234)

Timer operation examples

Consider an example of timer operation for dialling with 011 number overlap (example 1 from the previous section). Let us assume that the timer has the following values set:

L=10 seconds

S=5 seconds

Receiving the first digit — 0. A mask for such a dial matches to 2 rules: 011 and 0[1-4]. The first received digit does not provide any complete match to any of the rules, therefore the L-timer is activated (10 seconds) to wait for the next digit. If the next digit does not come in 10 seconds, a timeout will be registered. Since there are no matches to the rules, the timeout will result in dial error.

Receiving the second digit — 1. Receiving the second digit results in a match to rule 6: 0[1-4] (prefix 01). Since the match is found, but there may also be a further match to rule 5 (that is 011), the S-timer is activated (5 seconds) to wait for the next digit. If the next digit does not come in 5 seconds, a timeout will be registered. Since there is a match to a rule, the call will be successfully directed according to this mask.

Receiving the third digit — 1. There is no match to rule 6 anymore, but the number matches rule 5 now. This match is final, since the mask has no more rules for further matches. The call is immediately routed according to rule 5.

Configuration example of prefix with ‘subscribers pool’ type

Objective 

The following range of numbers is allocated to SMG: 26000 – 26199. However, not all numbers can be assigned to subscribers immediately. When an unassigned call arrives to a number in this range, SMG will reject it with release cause 3 – No route to destination. But since this numbering is local to the gateway, it should have sent release cause 1 – Unallocated (unassigned) number.

Solution 

For correct release cause transmission, local numbering should be created — configure a ‘subscribers pool’ type prefix. 

To do this, in the Dial plans section, add a new prefix with subscribers pool as the Prefix Type parameter value. In the prefix settings, add a list of prefix masks of the Called type. For the number range 26000–26199 specified in the objective, the mask will be as follows: (26[0-1]xx).

Call routing

Trunk groups

Call routing → TrunkGroup

A trunk group is a set of connection lines (trunks), which can be as follows: E1 stream channels, data transmission bandwidth (IP channels). E1 stream channels are used for Q.931 and SS7. IP channel interfaces are SIP/SIP-T/SIP-I/H.323. To edit a trunk group double-click the corresponding row in the group table with the left mouse button or select the group and click thebutton below the list.

To delete a trunk group, select the group and click thebutton below the list or open the Objects menu and select Remove Object.

Up to 255 trunk groups can be created.

‘Basic settings’ tab

To add a trunk group click the , button, then fill in the following fields:

Call routing → TrunkGroups →  

• Title – trunk group name;
• Description – trunk group description;
• TrunkGroup members – trunk group members:
  • Stream with Q.931 signaling, SS linkset, SIP or H323 interface;
  • E1 stream channels – E1 stream channels with Q.931, SS7 signaling protocols;
  • E1 streams from SS7 Linkset.
• E1 stream – E1 stream selection to assign the trunk group to E1 stream channels, this menu is active only when ‘E1 channels’ value is selected for ‘TrunkGroup members’ field.

Call routing → TrunkGroups → → Basic settings

• SS7 Linkset  – SS7 link set for selecting E1 streams. This menu is available only when you choose ‘SS7 Linkset lines’ in ‘TrunkGroup members’ menu;
• Channels selection order – channel selection order in E1 streams. This menu is available only when ‘SS7 Linkset lines’ is chosen in ‘TrunkGroup members’ menu;

• Local direction – when checked, subscribers of this direction are considered local. Subscribers of this direction are set under SORM control with the type and number sign as ‘subscriber of this station’;
• Play music on hold (MOH) – enabling Music On Hold option;
• Voice switch delay – forced voice switching path delay after the subscriber's answer.

‘Incoming calls’ tab

Call routing → TrunkGroups → → Incoming calls

• Disable ingress calls – when this option is checked, the incoming calls are prohibited. Setting the call prohibition does not terminate any of the established connections;
• Direct routing prefix – the prefix will be used without caller or callee number analysis. It enables switching of all calls in a single trunk group to another group regardless of the dialed number (without mask creation in prefixes). When a number is dialed in the overlap mode, direct dialling timers are used, which are configured in the direct prefix;
• Block when direct prefix is unavailable – option is available if the trunk group includes E1 circuits and a direct prefix has been selected. If this option is enabled, should the remote party (to which routing via the direct prefix is directed) fail, the E1 circuit from which the initiating call originated is deactivated. In this way, the initiating party understands that the circuit is no longer active, and failover is triggered on the side of the operator that initiated the call over the circuit;
• Use voice messages – when this option is selected, pre-recorded voice messages stored in the device memory will be played upon the occurrence of specific events. For detailed description, see Appendix G. Voice messages and music on hold (MOH); 
• No Connected number transit – disable the transmission of the Connected number field;
• Copy CgPN into Redirecting number – when this option is checked, if there is no Redirecting number in the incoming call, it will be generated from the CgPN number;
• Use Redirecting number for routing – when this option is checked, the Redirecting number field is used when using SS7 or Q.931 signaling protocols, the SIP diversion field is used to route the incoming call in the dial plan using CgPN number masks;
• CallerID request – specify the need of a caller's information (number and category) to call the trunk group. If a call is received from an interacting node and do not contain CallerID information, the CallerID request will be sent to the calling node (INR messages via SS7);
• Alarm CPS value – the number of calls per second after which a failure will be indicated in the log. ‘0’ value – the fault indication is turned off. Fault indication time — 10 minutes after exceeding the specified threshold of CPS;
• Max CPS value – the maximum number of calls per second that can be received by a trunk group. ‘0’ value – turning off the CPS limit. The CPS value is calculated as the moving average for the last 3 seconds. For example, if 3xCPS calls arrive within the first second, they will be accepted, but if there are any additional calls within the next two seconds, they will be rejected;
• RADIUS profile – selecting the RADIUS profile to use (profiles are configured in the RADIUS Configuration/Profile List menu);
• List of reasons for call recovery after outbound leg failure – if a call received via a trunk group with this setting enabled is not answered on the incoming end, the SMG will attempt to re-establish the connection without interrupting the call on the A side, either by making a callback or by using alternative routes if the primary route is unavailable.

Incoming calls modifiers

• CdPN modifiers – intended for modifications based on the analysis of the called number received from the incoming channel;
• CgPN modifiers – intended for modifications based on the analysis of the calling number received from the incoming channel.

‘Outgoing calls’ tab

Call routing → TrunkGroups → → Outgoing calls

• Disable egress calls – when this option is active, transmitting outgoing calls is forbidden. Setting the call prohibition does not terminate any of the established connections;
  
• Replace CgPN by Redirecting – when this option is active, the CgPN number is replaced with Redirecting;
• Check access category – when this option is active, it checks the possibility of call routing based on the rights determined by access categories;
• Reserve TrunkGroup – specifying a trunk group to which a call will be routed when routing to the current trunk group is not possible (all channels are engaged or inoperable);
• Q.850 release causes list for switching to reserve TG – selecting the Q.850 release causes table to configure the Q.850 release causes for switching to the reserve trunk group;
• RADIUS profile – selecting the RADIUS profile to use (profiles are configured in the RADIUS Configuration/Profile List menu).

Outgoing calls modifiers

• CdPN modifiers – intended for modifications based on the analysis of the called number received from the incoming channel;
• CgPN modifiers – intended for modifications based on the analysis of the calling number received from the incoming channel;
• Original CdPN – intended for modifications based on analysis of the original called number transmitted to the outgoing channel;
• RedirPN modifier – intended for modifications based on the analysis of the redirecting number transmitted to the outging channel;
• GenericPN – intended for modifications based on the analysis a special number (generic number) transmitted to the outgoing channel;
• LocationNumber – are intended for modifications based on the analysis location number transmitted to the outgoing channel.

To create, edit, or remove groups (as well as other objects), use the ‘Objects’ — ‘Add object’, ‘Objects’ — ‘Edit object’ and ‘Objects’ — ‘Remove object’ menus and the following buttons:

– Add a truck group;

– Edit trunk group parameters; 

– Remove a trunk group.

RingBack settings 

Mode:

• Default – the option corresponds to the default settings;
• RingBack – play the standard ringback tone, ignore the default settings;
• Audio file – change the standard ringback tone to a chosen one which has been downloaded in System settings (an individual sound for the direction).

SS7 Linkset

Call routing → SS7 Linkset

 SS7 Linkset is a set of signaling links in one direction. To create, edit, or remove linkset, use the ‘Objects’ — ‘Add object’, ‘Objects’ — ‘Edit object’ and ‘Objects’ — ‘Remove object’ menus and the following buttons:

– Add SS7 Linkset;

– Edit SS7 Linkset; 

– Remove SS7 Linkset.

Call routing → SS7 Linkset →

SS7 Linkset;

• Title – SS7 linkset name;
• TrunkGroup – name of a trunk group that SS7 linkset operates with;name of a trunk group that SS7 linkset operates with;
• Access category – selects access category;
• Dial plan – defines dial plan that will be used for routing in this group (necessary for dial plan negotiation);
• Scheduled routing profile – selects 'scheduled routing' service profile, configured in the 'Internal resources' section;
• Toll – means that the signal link is connected to ALDE. This parameter allows for the correct operation with the long-distance type calls (used for CAS transits);
• Alarm indication – when checked, fault indication will appear in case of SS7 signal link fault (ALARM LED will light up, alarm will be added to alarm log);
• Channel selection – channel engagement order for the outgoing calls. Available options:
  • Successive forward;
  • Successive backward;
  • From first forward;
  • From last backward;
  • Successive forward (even);
  • Successive back (even);
  • Successive forward (odd);
  • Successive back (odd).

• Reserve SS7 Linkset – redundant SS7 linkset selection. When the main SS7 linkset is not available, the whole signalling message exchange will be performed through the redundant SS7 linkset;
• Combined mode – Combined Linkset mode that will enable the exclusive utilization of voice streams in the current SS7 link set and signalling transfer through the signal channels of SS7 primary and secondary groups;
• Primary SS7 Linkset – selects SS7 link set, that will perform the exchange of signalling messages related to this particular SS7 link set, by the signal D-channels;
• Secondary SS7 Linkset – selects the second SS7 link set, that will perform the exchange of signaling messages related to this particular SS7 link set, by the signal D-channels;

• SS7 Timers profile – selects the timer profile that will be used for the current SS7 linkset;

• Stream order by SLC – affects the operation of the Order of channel engagement setting. With this option enabled, the order of engaged E1 streams is determined by the SLC number (sorted from a smaller SLC to a larger one), with this option disabled the order is determined by the E1 stream index.

MTP2 Layer settings

• Emergency alignment for a single link – enabling emergency phasing procedure during SS7 linkset commissioning, if this SS7 linkset has a single signal link.

 Service information (SIO)

• Network ID – indicates the network type: international, federal, local network or spare (usually on RF networks the value “Local network” is used).

 Routing label

• OPC – own code of the signaling point;
• DPC-ISUP – destination point code of the ISUP subsystem.

ISUP subsystem

• Channels initialization mode – device operations during stream recovery:
  • Remain in block – channels remain blocked (BLO);
  • Individual unblock – sending unblock command for each channel (UBL);
  • Group unblock – sending channel group unblock command (CGU);
  • Group reset – group reset command (GRS).
• Send REL on receiving SUS – sending Release message in response to Suspend message;
• Add a digit in IAM for overlap – sending a single digit of the number to Called Party number of IAM message if overlap dialing method is used;
• Restrict CdPN in IAM to 15 digits – when active, up to 15 digits of CdPN number will be sent in IAM message, other digits will be sent in SAM message;
• Control receiving Redirecting/Original Called for incoming redirection – this checkbox enables controlling the presence of Redirecting/Original Called fields with redirection information in incoming IAM message; when this option is active, the call will be rejected if these fields are absent;
• Ignore HOLD indication – when checked, SMG will ignore the CPG messages with remote hold or remote retrieval signs;
• Transmit Global Callref – when there is no Global Call Reference (GCR) field in an incoming leg, SMG forms it automatically;
• Hop counter – setting rules for operation with hop counter field:
  • Decrement;
  • No change;
  • Present;
  • Don't send.
• Hop counter to Max Forwards conversion rate – specifies the multiplier by which the value of the Hop counter (SS7) is multiplied; the resulting value is then rounded and transferred to Max Forwards (SIP) in SS7–SIP calls. The reverse conversion from Max Forwards to Hop counter in SIP–SS7 calls is performed as follows:
  • if Max Forwards <= 31, the value is copied into the Hop counter as is;
  • if 31 < MF <= 70, the value is divided by 2.25 and rounded;
  • if 70 < MF <= 140, the value is divided by 4.5 and rounded;
  • if 140 < MF <= 255, the value is divided by 8.25 and rounded;
• Use extra 3 bits for CIC – allow CIC values greater than 4095 to be used;
• Send Generic Number field – transmiting Generic Number in SS7 when a call is received via SIP and a Remote-Party-ID is present.

IAM indicators

• Transmission medium requirements – indicates the information type that should be transmitted via transmission medium; when transit type is selected, the value of the field is taken from the incoming connection leg. If this field is missing from the incoming leg, default value 1 kHz audio is taken.

Forward call indications

• ISUP preference – a rule that governs ISUP preference indicator modification. In a standard situation, these bits should not be changed;
• Interworking indicator – defining whether the interaction indicator should be modified or not (defines whether the interaction with non-ISDN network has occurred);
• Call type indicator – modifying a National/international call indicator parameter in FCI.

Connect type indicators

Satellite indicator – identifies the presence of a satellite channel:

• Change to ‘no satellite’ – changing identifier value to no satellite regardless of the value received from the incoming channel;
• Unchanged – keeping the indicator value unchanged;
• Add one satellite – this setting is used if the signal link operates via satellite channel. In this case, a satellite channel parameter transmitted in the nature of connection indicators will be increased by 1.
• Enable continuity check – enables integrity check support in the SS7 link set. During the outgoing call, the called party establishes a remote loop in the stream. The SMG sends the frequency value to the channel and then detects it on reception after transmission through the channel. If the frequency is detected, the call will be served at this channel; if it is not detected, the similar attempt will be performed at the next channel. After 3 unsuccessful attempts (for three different channels), call serving will stop;
• Continuity check frequency – calls performed via the SS7 link set. For example, value 3 means that each third outgoing call will be performed with the channel integrity check.
  defines the frequency of channel continuity checks during outgoing.

For the gateway, you may assign the correspondence of SS categories to Caller ID categories. For configuration, see section SS7 categories. 

Netlink:

Configuration of the device operating mode when SS-7 streams are stacked, and related parameters.

• Enable;
• Primary SIP interface;
• Secondary SIP interface.

Master mode:

• Local interface – network interface on which the device will receive incoming connection requests.
• Local port – network port on which the device will receive incoming connection requests.

Slave mode:

• Local interface – network interface on which the device will send incoming connection requests.
• local port – network port on which the device will send incoming connection requests.
• Master IP address – address to which connection requests will be sent.
• Master port – port to which connection requests will be sent.

In slave mode, it is possible to establish up to four simultaneous outgoing connections.

Detailed description of the operation and configuration of SS-7 stream stacking is provided in Appendix M. SS7 stack monitoring.

Examples

• SMG connection method example for operation in SS7 quasi-associated mode via signaling transition points (STP):

Figure 46 – SMG connection method for operation in SS7 quasi-associated mode via STP

Objective

It is necessary to provide the SMG connection to the remote signalling point (SP) using two signal links. The first signal link should pass through the signalling transition point STP 1 and the second signal link should pass through the STP 2.

Point code: SMG = 22, STP 1 = 155, STP 2 = 166, SP = 23.

Solution

In addition to the basic settings, set the 'origination code (OPC) = 22 and ISUP destination code (DPC-ISUP) = 23 in 'SS7 link set' menu.

Let us assume that stream 0 is connected to STP1 and stream 1 to STP 2. In the stream settings, one should specify: SS7 'Signalling protocol', configure CIC numbering correctly and select the required E1 stream time slot for signalling D-channel, select the pre-created SS7 link set in 'SS7 link set' settings and define the parameter 'MTP3 destination code (DPC-MTP3)' equal to 155 for stream 0, and 166 for stream 1.

• SMG connection method example for operation in SS7 quasi-associated mode via PBX with STP features:

Figure 47 – SMG connection method for operation in SS7 quasi-associated mode via PBX with STP (LS – SS7 Link Set)

Objective

It is necessary to provide SMG connection to a couple of PBXes with STP features (PBX/STP); when the failure occurs in the main circuit group 1LS between SMG and PBX/STP 1, signalling messages should be sent via 2LS.

Solution

Let us assume that SMG stream 0 is connected to PBX/STP 1 and used for the first SS7 link set configuration, SMG stream 1 is connected to PBX/STP 2 and used for the second SS7 link set configuration. In the stream settings, specify: SS7 'Signalling protocol', configure CIC numbering correctly and select the required E1 stream time slot for signalling D-channel, select the second SS7 link set in the ‘Reserve SS7 Linkset' setting in the first SS7 link set configuration.

• SMG connection method example for operation in combined mode:

Figure 48 – SMG connection method for operation in combined mode

Objective

Only the voice channels exist between SMG and PBX/SP, signalling traffic should be transferred via PBX/STP 1 and PBX/STP 2.

Solution

Let us assume that SMG stream 0 is connected to PBX/STP 1 and used for the first SS7 linkset configuration, SMG stream 1 is connected to PBX/STP 2 and used for the second SS7 linkset configuration, SMG stream 2 is connected to PBX/SP and used for the third SS7 linkset configuration. In the stream settings, you should specify: SS7 'Signalling protocol', configure CIC numbering correctly and for streams 0 and 1 select the required E1 stream time slot for signalling D-channel, select the first SS7 linkset in the 'Primary SS7 Linkset' setting and the second SS7 linkset in the 'Secondary SS7 link set' setting in the third SS7 link set configuration.

SIP/SIP-T/SIP-I, SIP-profiles

Configuration

This section describes configuration of general parameters for SIP stack, custom settings for each direction operating via SIP/SIP-Т/SIP-I protocols, and SIP subscriber profiles.

SIP (Session Initiation Protocol) is a signalling protocol, which used in IP telephony. It facilitates basic call management tasks such as session start and termination.

SIP network addressing is based on the SIP URI scheme:

sip:user@host:port;uri-parameters

user – the number of a SIP subscriber;

@ – a separator located between the number and domain of the SIP subscriber;

host – domain or IP address of the SIP subscriber;

port – the UDP port used for subscriber's SIP service operation;

uri-parameters – additional parameters.

One of the additional SIP URI parameters is user=phone. If this parameter is specified, the syntax of the SIP subscriber number (in the user part) should match the TEL URI syntax described in RFC 3966. In this case, SMG PBX will process requests that contain ‘+’, ‘;’, ‘=’, ‘?’ in the SIP subscriber number, and will automatically add ‘+’ before the called number for international calls using the SIP-T protocol.

Call routing → SIP interfaces → Settings

Common SIP settings

• (x100 ms) T1 timer – timeout for a response to the request, after which the request will be sent again. The maximum retranslation interval for INVITE requests is 64*T1;
• (x100 ms) T2 timer – the maximum retranslation interval for responses to the INVITE request and for all requests except for the INVITE requests;
• (x100 ms) T4 timer – the maximum time for all retranslations of the final response;
• Ringing timeout (sec) – pre-answering state timeout of the call after reception of 18X message, during which the ringback tone or IVR message is played to the subscriber;
• Ignore address from R-URI – when this option is active, address information after the ‘@’ separator in Request-URI is ignored. Otherwise, the gateway checks if the address information matches the device’s IP address and host name; if there is no match, the call is rejected;
• Enable KZ SIP specification
• Save subscribers DB – when this option is active, saving details of registered subscribers to the non-volatile memory of the gateway. The option is required to save the database of registered subscribers in case of device reboot due to power loss or failure. If the gateway is rebooted from WEB or CLI, the current database will be saved to non-volatile memory regardless of this setting;
• Ignore OPTIONS from unknown subscribers – enable or disable the sending of responses to OPTIONS requests from unknown sources.

The SIP protocol defines two types of responses to connection initiating requests (INVITE) — provisional and final. 2хх, 3хх, 4хх, 5хх and 6хх-class responses are final, their transfer is reliable and confirmed by the ACK message. 1хх-class responses, except for the 100 Trying response, are provisional and do not have a confirmation (RFC3261). These responses contain information on the current INVITE request processing step; in SIP-T/SIP-I protocols, SS-7 messages are encapsulated into 1xx class responses, therefore the loss of these responses is unacceptable. Utilisation of reliable provisional responses is also realised in the SIP protocol (RFC3262) and is defined by the 100rel tag in the initiating request. In this case, provisional responses are confirmed by a PRACK message.

Up to 255 interfaces can be created. To create, edit, or remove SIP/SIP-T interfaces, use the Objects – Add Object, Objects – Edit Object, or Objects – Remove Object menus and the following buttons:

– Add interface;

– Edit interface parameters;

– Remove interface;

  – Move interfaces up or down.

The signal processor of the gateway encodes analogue voice traffic and fax/modem data into digital signals and performs its reverse decoding. The gateway supports the following codecs: G.711A, G.711U, G.722, G.723.1, G.726, G.729, Т.38 protocol and CLEARMODE.

G.711 is a PCM codec without compression of voice data. To ensure correct operation, this codec should be supported by all manufacturers of VoIP equipment. G.711A and G.711U codecs differ from each other in encoding law (А-law is a linear encoding and U-law is a non-linear). The U-law encoding is used in North America, and the A-law encoding – in Europe.

G.722 – ITU-T standard broadband voice codec operating at bit rates of 48, 56 and 64 kbps. The codec technology is based on ADPCM.

G.726 is an ITU-T standard for adaptive pulse code modulation — ADPCM and describes voice transmission with a bandwidth of 16, 24, 32, and 40 kilobits/sec. G.726-32 replaces G.721, which describes ADPCM voice transmission with a bandwidth of 32 kilobits/sec.

G.723.1 is a codec with speech information compression, provides two operating modes: 6.3 Kbit/s and 5.3 Kbps. The G.723.1 codec has a speech activity detector and provides generation of comfortable noise at the remote end during the silent period (Annex A).

G.729 is also a voice compression codec and provides a bit rate of 8 Kbps Similar to the G.723.1 codec, the G.729 codec supports speech activity detection and ensures the generation of comfortable noise (Annex B).

T.38 is a standard that describes the transmission of fax messages in real time over IP networks. Signals and data transmitted by a fax machine are encoded into T.38 protocol packets. In generated packets, redundancy can be introduced — data from previous packets, which allows carring out reliable fax transmission over unstable channels.

CLEARMODE is a mode in which signal encoding/decoding is not used. Used for transparent transmission of digital information 64 kbit/s (RFC4040).

‘SIP interface settings’ tab

To create SIP/SIP-T interfaces, use the ‘Objects’ menu – ‘Add object’ or the , button, when pressed, the following menu appears:

Call routing → SIP interfaces → Settings →→ SIP interface settings

• Title – the interface name;
• Mode – selects the interface protocol (SIP/SIP-T/SIP-I/SIP profile);
• Ingress RADIUS profile – selects the RADIUS profile for the SIP profile interface for incoming communication (for other interfaces, the RADIUS profile is assigned in the trunk group);
• Egress RADIUS profile – selects the RADIUS profile for the SIP profile interface for outgoing communication (for other interfaces, the RADIUS profile is assigned in the trunk group);
• Trunk group¹ – name of the trunk group to which the interface belongs;
• Access category – selects an access category;
• Dial plan – defines the dial plan that will be used for dialling from this port (required for coordination of dial plans);
• Hostname/IP-address¹ – IP address or name of the host communicating via the gateway SIP/SIP-T protocol;
• Subnet mask for incoming calls – if the mask is set, SMG will receive calls from the subnet holding the connecting host, specified in the ‘Host name/IP address’ field. Note that when using the masks 0.0.0.0 (/0), 255.255.255.255 (/32) or 255.255.255.254 (/31), SMG will only accept calls from the IP address indicated in the ‘Host name/IP address’ field, rather than from the subnet;
• Remote SIP port¹ – a UDP/TCP port of the communicating gateway that is used to receive SIP/SIP-T signalling;
• Local SIP port¹ – a local UDP/TCP port of the device used to receive SIP/SIP-T signalling from the device communicating via this interface;
• SIP domain – a domain that is placed into the from field when an outgoing call is made through the SIP interface; is used in the SIP interface registration;
• Ignore source port for incoming calls – when this option is checked, the signalling transmission UDP port of the communicating gateway that is specified in the Port for SIP Signalling Reception parameter is not checked; otherwise, the port is checked and the call is cleared back if the INVITE request is received from another port. If the INVITE request is received via TCP, the port is not checked regardless of the parameter value;
• Trusted network – means that the interface is connected to a trusted network. This option defines generation of the INVITE request fields for calls with hidden caller number (presentation restricted). When this option is checked, the caller number information is transmitted in the from and P-Asserted-identity fields together with the information on its hidden state in the Privacy: id field; otherwise, the caller number information is not transmitted in any fields;
• Alarm indication – выбор сетевого интерфейса для приема и передачи сигнальных SIP-сообщений;
• Network interface for RTP – selects a network interface to receive and transmit voice traffic;
• Q.850-cause and SIP-reply mapping table – table of correspondence between Q.850-cause and SIP-reply codes. To configure correspondence tables, use the ‘Internal Resources’ menu;
• SIP-replies list for switching to reserve TG – selects the reply table for SIP 4XX – 6XX classes for transition to a reserve trunk group. The replies list table is configured in Internal resources section; 
• Scheduled routing profile – selects a profile for the Scheduled Routing service configured in the Internal resources section;

• Lines operation mode – setting lines operation mode to limit the number of simultaneous calls via this interface:

  • Common – considering the total number of simultaneous calls (incoming and outgoing) via this interface;

  • Separate – incoming and outgoing calls are counted separately.

• Max active calls – maximum number of simultaneous (incoming and outgoing) connections via this interface. The field is displayed if Common operation mode is selected;

• Ingress lines number – number of simultaneous incoming calls via this SIP interface. The field is displayed if Separate operation mode is selected;

• Egress lines number – number of simultaneous outgoing calls via this SIP interface. The field is displayed if Separate operation mode is selected;

• Transport – selecting a transport level protocol using for reception and transmission of SIP messages:
  
  • TCP-prefer – receiving by UDP and TCP. Sending via TCP. If not connected by TCP, make attempt by UDP;
  • UDP-prefer – receiving by UDP and TCP. Transmitting by TCP whenever packet is greater than 1300 bytes, otherwise by UDP;
  • UDP-only – receiving and transmitting only by UDP;
  • TCP-only – receiving and transmitting only by TCP;
• Max Forwards – selecting the “Max Forwards” header processing mode: 
  • Transit – pass-by-value (default behaviour);
  • Don't change – pass by no change;
  • Value – pass by a given value;
  • Deny – disable the setting of Max Forwards for outgoing connections, or ignore the received parameter for incoming connections.
• Global Callref generation – if there is no GCR in a call, it will be generated locally. If there is GCR in a call, it will be transmitted further without generating a new one. The option is only available for SIP-I;

• Node ID – an identifier used for generating a global Callref. The range of allowed values is [0;255]. The option is only available for SIP-I.
  

STUN server settings and Public IP

STUN network protocol (RFC 5389) allows applications located behind a network address translation server (NAT) to discover their external IP address and port mapped to an internal port. Used when SMG is located behind a NAT. To identify external device address, use STUN or Public IP (used separately).

• Enable – when checked, use STUN server, otherwise use a specified public IP address;
• IP-address – IP address of STUN server;
• Port – server port for request transmission (default value is 3478);
• Requests period – time interval between requests (10–1800 seconds);
• Public IP – sets public (external) address of NAT WAN interface to insert in SIP messages.

Before signalling message transmission, the request (Binding Request) has been sent to the STUN server from the interface; in the response (Binding Response) message, STUN server communicates device IP address and port (udp) that are used by SMG in signalling message generation.

Requests to STUN server has been generated before each SIP signalling message transmission, but not more often than the configured request period time.

Public IP setting is not used in the ‘SIP profile’ interface mode.

‘SIP protocol setting’ tab

Call routing → SIP interfaces → Settings → → SIP protocol settings

Setting options for SIP/SIP-T/SIP-I protocols

• Keep-alive control – a function that controls direction availability by sending OPTIONS requ-ests; when a direction is not available, the redundant trunk group is used for the call. This function also analyses the received OPTIONS response that allows avoiding the use of the 100rel, replaces, and timer features configured in this direction, unless the opposite party supports them. The parameter defines the request transmission period and may take values in the range of 30–3600 seconds;
• Keep alive mode:
  • SIP-OPTIONS – at specified opposite party control intervals, the device will send the OPTIONS control message. This message should receive a response from the opposite party; if no response is received, the direction is considered unavailable, and the failure status is registered in the device;
  • SIP-NOTIFY – the device will send the NOTIFY control message at specified oppo-site party control intervals. This message should receive a response from the opposite party; if no response is received, the direction is considered unavailable, and the failure status is registered in the device;

  • UDP-CRLF – device will send an empty UDP packet at specified opposite party control intervals; the opposite party response to an empty UDP packet is not applicable; consequently, the failure status will not be initiated on the device.

    These methodds are also used to maintain the NAT connection.

• Always transmit SDP in provisional responses – allows early forwarding of the voice frequency path. For example, when this option is not checked, SMG sends reply 180 without SDP session description; according to this reply, the outgoing party plays the ringback tone; when this option is checked, SMG sends reply 180 with SDP session description and the ringback is played by the incoming party;
• 'In-band signal' with 183+SDP transmission – issues SIP-reply 183 with SDP session description for voice frequency path forwarding upon receipt of the CALL PROCEEDING or PROGRESS messages from ISDN PRI that contain the progress indicator = 8 (in-band signal);
• Local ringback instead of early-media – when the early media marker is received from the outgoing leg, ringback tone will be played to the caller instead of the inband voice message;
• Enable P-Early-Media (RFC5009) – use the P-Early-Media header described in RFC 5009. With outgoing call, the device will transmit the P-Early-Media: supported header in the INVITE. Upon receving INVITE with P-Early-Media: supported marker, the response 18X messages will contain the P-Early-Media header: sendrecv;
• Display name – сonfiguring the Display-Name field:
  • Don't change – when a call is received with an existing Display-Name, pass it through unchanged;
  • Fill empty display name – when receiving a call with a missing Display-Name, fill it with the user name (or number) taken from the URI;
  • Replace with number – when receiving a call with an existing or missing Display Name, replace it with the phone number or fill it in;
  • Don't send – when receiving a call with an existing Display-Name, do not relay it.
• Send DisplayName in Remote-Party-ID header – enables/disables substitution of DisplayName in Remote-Party-ID;
• Allow to send 'Reason: preemption' – option enables or disables the insertion of the additional header «Reason: preemption; cause=1; text="Transit release from SIP-peer"» in release messages;
• Ignore RURI and To difference – disables issuing the Redirecting and Original Called numbers in SS7 calls when the values in SIP RURI and To fields are different;
• Do not use plus sign in CdPN and Diversion – disables addition of ‘+’ to a number, for International number type;
• Diversion header with SIP URI – uses SIP URI in the Diversion header instead of TEL URI;
• Enable CCI – for SIP-I/T, enable transmission of IAM with a Continuity check indication value of 2. The option is available only for SIP-T and SIP-I protocols;
• Enable redirection (302) processing – when this option is checked, the gateway is allowed to perform forwarding upon receipt of reply 302 from this interface. When unchecked and reply 302 is received, the gateway will reject the call and perform forwarding;
• Redirection server direction – option is available when 302 response handling is enabled (the ‘Allow redirection (302)’ parameter). It allows a call sent to a public address to be redirected to the subscriber’s private address specified in the 302 response, without using routing via the numbering plan. Routing is performed directly to the address from the contact header of the 302 response received from the redirection server;
• Enable REFER processing – a REFER request is sent by the communicating gateway to enable the Call Transfer service. When this option is checked, the gateway is allowed to process REFER requests received from this interface. When unchecked, the gateway clears back the call upon receipt of a REFER request and does not provide the Call Transfer service;
• Enable Re-INVITE with a=sendonly processing – when this option is checked, it allows a call to be put on hold when the Re-INVITE message is received with a=sendonly marker in SDP;
• Send calling category – selection of the method for transmitting the calling party’s category via the SIP protocol (in the From and P-Asserted-Identity headers). The following methods are implemented:
  • off – sending and receiving of Caller ID category are disabled;
  • category – the caller category is sent/received in a separate category field in the INVITE message; in this case, the SS7 category with values 0 – 255 is sent;
  • cpc – the caller category is sent/received via the “cpc=” tag transmitted in the from field, in this case, the Caller ID category with values 1–10 is sent;
  • cpc-rus – the caller category is sent/received via the “cpc-rus=” tag transmitted in the from field; in this case, the Caller ID category with values 1–10 is sent.
• Reliable provisional responses (1xx) – when this option is checked, the INVITE request and 1хх class provisional responses will contain the require: 100rel option, which requires assured confirmation of provisional responses:
  • off – reliable delivery of provisional responses is disabled;
  • support – the INVITE request and 1хх class provisional responses will contain the support: 100rel option;
  • support+ – duplicate SDP in 200 OK message when using support: 100rel;
  • require – the INVITE request and 1хх class provisional responses will contain the require: 100rel option, which requires assured confirmation of provisional responses;
  • require+ – duplicating SDP in 200 OK message when using require: 100rel.

• DSCP for Signaling – a service type (DSCP) for SIP signalling traffic;

  The DSCP value for Signalling must be the same for all directions with the same transport parameter (network interface + signalling receive port). DSCP for RTP and DSCP for SIP settings will be ignored when using VLAN for RTP transmission and signaling. To prioritize traffic in this case there will be used Class of Service VLAN.

  
  

• Transit SIP header – enables transit of the received SIP headers into the outbound leg.
• Send Remote-Party-ID – transmit the Remote-Party-ID header; if it is missing from the incoming call, generate it based on the Generic Number;
• Routing by To header – option enables call routing using the CdPN value from the To header (by default, the CdPN value from the RURI is used). 

SIP-session timers (RFC 4028)

• Enable – when this option is checked, support of SIP session timers (RFC 4028) is enabled. A session is renewed by re-INVITE requests sent during the session;
• Session Expires – a period of time in seconds before a forced session termination if the session is not renewed in time (from 90 to 64,800 seconds; 1,800 seconds is recommended);
• Min SE – the minimal time interval for connection health checks (from 90 to 32,000 seconds). This value should not exceed the Sessions Expires forced termination timeout;

• Refresher side – defines the party to renew the session (client (uac) – client (calling) party, server (uas) – server (called) party).

Registration settings¹:

• Upper registration – the selected type of registration on an upstream server:
  
  • No registration – do not perform registration on the upstream server;
  • Trunk registration – registration on the upstream server using parameters specified in this section;
  • User registration – registration on the upstream server using parameters specified on the 'registration' tab. This registration type allows to define the list of subscribers with enabled access via this interface;
  • Upper registration – transit registration of device subscribers on the upstream server; when this option is selected, SMG will transfer subscribers' SIP messages via this SIP interface. When transit registration is selected, you should specify this SIP interface in the settings of SIP profile that requires transit registration.
• Login – the name used for authentication;
• Password – the password used for authentication;
• Username/Number – the user number which is used as a caller number for outgoing trunk calls;
• Default CdPN – the default CdPN number that will be used for all calls via this SIP interface;
• Replace CgPN on egress call – when this option is checked, the caller number (CgPN) is taken from the Username/Number parameter; otherwise, the CgPN number received in the incoming call is used;
• Hang up code when the upstream server is unavailable – Q.850 hang-up code in the event that the upstream trunk registration server is unavailable. If the value in the field is 0, the default hang-up code will be 27;
• Registration period (sec) – the time interval for registration renewal;
• Registration requests interval (ms) – the minimum interval between the Register messages that is used to protect from high traffic caused by simultaneous registration of a large number of subscribers.

Setting options for SIP profile

Call routing → SIP interfaces → Settings → SIP interface #1 → SIP protocol settings

• Keep-alive control – a function that controls the direction availability (NAT keep-alive) using the SIP-OPTIONS, SIP-NOTIFY or empty UDP method. The parameter determines the request transmission period and takes values from the range 30–3600 s;
• Keep-alive mode:
  
  • SIP-OPTIONS – at specified opposite party control intervals, the device will send the OPTIONS control message. This message should receive a response from the opposite party; if no response is received, the direction is considered unavailable, and the failure status is registered in the device;
  • SIP-NOTIFY – the device will send the NOTIFY control message at specified oppo-site party control intervals. This message should receive a response from the opposite party; if no response is received, the direction is considered unavailable, and the failure status is registered in the device;

  • UDP-CRLF – device will send an empty UDP packet at specified opposite party control intervals; the opposite party response to an empty UDP packet is not applicable; consequently, the failure status will not be initiated on the device.

    These methods are also used to maintain the NAT connection.

• Always transmit SDP in provisional responses – allows for early connection of the voice path. For example, if the flag is unchecked, then SMG sends a 180 response without SDP session description, based on this response, the outgoing party plays a ringback, when the flag is checked, SMG sends a 180 response with SDP session description, and the ringback is played by the incoming party;
• 'In-band signal' with 183+SDP transmission – issues SIP-reply 183 with SDP session description for voice path forwarding upon receipt of the CALL PROCEEDING or PROGRESS messages from ISDN PRI that contain the progress indicator = 8 (in-band signal);
• Local ringback instead of early-media – when the early media marker is received from the outgoing leg, ringback tone will be played to the caller instead of the inband voice message;
• Enable P-Early-Media (RFC5009) – use the P-Early-Media header described in RFC 5009. With outgoing call, the device will transmit the P-Early-Media: supported header in the INVITE. Upon receving INVITE with P-Early-Media: supported marker, the response 18X messages will contain the P-Early-Media header: sendrecv;
• Display name – сonfiguring the Display-Name field:
  • Don't change – when a call is received with an existing Display-Name, pass it through unchanged;
  • Fill empty display name – when receiving a call with a missing Display-Name, fill it with the user name (or number) taken from the URI;
  • Replace with number – when receiving a call with an existing or missing Display Name, replace it with the phone number or fill it in;
  • Don't send – when receiving a call with an existing Display-Name, do not relay it.
• Send DisplayName in Remote-Party-ID header – enables/disables substitution of DisplayName in Remote-Party-ID;
• Allow to send 'Reason: preemption' – option enables or disables the insertion of the additional header «Reason: preemption; cause=1; text="Transit release from SIP-peer"» in release messages;
• Ignore RURI and To difference– disables issuing the Redirecting and Original Called numbers in SS7 calls when the values in SIP RURI and To fields are different;
• Do not use plus sign in CdPN and Diversion – disables addition of ‘+’ to a number, for International number type;
• Diversion header with SIP URI – uses SIP URI in the Diversion header instead of TEL URI;
• Enable redirection (302) processing – when this option is checked, the gateway is allowed to perform forwarding upon receipt of reply 302 from this interface. When unchecked and reply 302 is received, the gateway will reject the call and perform forwarding;
• Enable REFER processing – a REFER request is sent by the communicating gateway to enable the Call Transfer When this option is checked, the gateway is allowed to process REFER requests received from this interface. When unchecked, the gateway clears back the call upon receipt of a REFER request and does not provide the Call Transfer service;
• Enable Re-INVITE with a=sendonly processing – when this option is checked, it allows a call to be put on hold when the Re-INVITE message is received with a=sendonly marker in SDP;
• Reliable provisional responses (1xx)– when this option is checked, the INVITE request and 1хх class provisional responses will contain the require: 100rel option, which requires assured confirmation of provisional responses:
  
  • off – reliable delivery of provisional responses is disabled;
  • support – the INVITE request and 1хх class provisional responses will contain the support: 100rel option;
  • require – the INVITE request and 1хх class provisional responses will contain the require: 100rel option, which requires assured confirmation of provisional responses.
• DSCP for Signaling – a service type (DSCP) for SIP signalling traffic.

• Send Remote-Party-ID – enables/disables substitution of DisplayName in Remote-Party-ID;
• Transit SIP header – enables transit of the received SIP headers into the outbound leg;
• Max forwarding count between subscribers — maximum possible number of consecutive forwardings between subscribers, default value is 5.

• Routing by To header – option enables call routing using the CdPN value from the To header (by default, the CdPN value from the RURI is used). 

Expires

• Register – minimum and maximum values for the “expires” registration time;
• Subscribe – minimum and maximum values for the subscription expires time.

NAT settings

• NAT (comedia mode) – option required for correct operation of SIP through NAT (Network Address Translation) when SMG is used in a public network. Verifies source data in the incoming RTP stream and translate the outgoing stream to IP address and UDP port that the media stream is coming from;
• Transmit SDP in 18x messages – translate SDP in 18x provisional replies when NAT option is enabled (comedia mode). Allows performing an early forwarding of voice path (before the subscriber answers) and early source data verification in the incoming RTP stream;

• VIA and IP address match control – NAT traversal support option. When enabled, VIA address and request originator IP address will be analyzed. When they match, SMG will assume that the device is located outside the NAT.

SIP Session Timers (RFC 4028)

• Enable – when this option is checked, enables support of SIP session timers (RFC 4028). A session is renewed by re-INVITE requests sent during the session;
• Session Expires – a period of time in seconds before a forced session termination if the session is not renewed in time (from 90 to 64,800 seconds; 1,800 seconds is recommended);
• Min SE (Minimum session expiration) – the minimal time interval for connection health checks (from 90 to 32,000 seconds). This value should not exceed the Sessions Expires forced termination timeout;
• Refresher side – defines the party to renew the session (client (uac) – client (caller) party, server (uas) – server (callee) party).

Upper registration settings¹:

• Upper registration interface – selecting a SIP interface for transit registration. Up to five transit registration interfaces can be configured for a single SIP profile. If the first transit interface in the list becomes unavailable, the system will automatically switch to the next interface. If a previously unavailable interface becomes available again, there will be no automatic switch back to that interface – the current interface will be used until the registration is updated.  If all interfaces in the list are unavailable, registrations and calls will be processed locally on the SMG.

Setting options for SIP-Q

• Keep-alive control – a function that controls the direction availability (NAT keep-alive) using the SIP-OPTIONS, SIP-NOTIFY or empty UDP method. The parameter determines the request transmission period and takes values from the range 30–3600 s;
• Keep-alive mode:
  
  • SIP-OPTIONS – at specified opposite party control intervals, the device will send the OPTIONS control message. This message should receive a response from the opposite party; if no response is received, the direction is considered unavailable, and the failure status is registered in the device;
  • SIP-NOTIFY – the device will send the NOTIFY control message at specified oppo-site party control intervals. This message should receive a response from the opposite party; if no response is received, the direction is considered unavailable, and the failure status is registered in the device;

  • UDP-CRLF – device will send an empty UDP packet at specified opposite party control intervals; the opposite party response to an empty UDP packet is not applicable; consequently, the failure status will not be initiated on the device.

• DSCP for Signaling – a service type (DSCP) for SIP signalling traffic;

• Transit SIP header – enables transit of the received SIP headers into the outbound leg.
• Send Remote-Party-ID – transmit the Remote-Party-ID header and, if this is missing from the incoming call, generate it based on the Generic Number;
• Routing by To header – enable call routing using the CdPN value from the To header (by default, the CdPN value from the RURI is used).

SIP-session timers (RFC 4028)

• Enable – when this option is checked, enables support of SIP session timers (RFC 4028). A session is renewed by re-INVITE requests sent during the session;
• Session Expires – the time interval in seconds after which the session will be forcibly terminated if it is not refreshed within that time (from 90 to 64,800 seconds; the recommended value is 1,800 seconds);
• Min SE (Minimum session expiration) – the minimal time interval for connection health checks (from 90 to 32,000 seconds). This value should not exceed the Sessions Expires forced termination timeout;
• Refresher side – defines the party to renew the session (client (uac) – client (calling) party, server (uas) – server (called) party).

‘Codecs/RTP settings’ tab

Call routing → SIP interfaces → Settings →→ Codecs/RTP settings

Options

• Voice activity detector/Comfort noise generator (VAD/CNG) – when checked, silence detector and comfort noise generator are enabled. Voice activity detector disables transmission of RTP packets during periods of silence, reducing loads in data networks;
• Source IP: Port verification – when this setting is checked, control of media traffic received from IP address and UDP port specified in SDP communication session description will be enabled; otherwise the traffic from any IP address and UDP port will be accepted;
• Echo-cancellation – echo cancellation mode:
  • voice(default) – echo cancellers are enabled in the voice data transmission mode. 
  • voice nlp-off – echo cancellers are enabled in voice mode, non-linear processor (NLP) is disabled. When signal levels on transmission and reception significantly differ, weak signal may become suppressed by the NLP. Use this echo canceller operation mode to prevent the signal suppression.
  • modem – echo cancellers are enabled in the modem operation mode (direct component filtering is disabled, NLP control is disabled, CNG is disabled).
  • voice nlp-option 1 – echo cancellers are enabled in the voice mode, non linear processor NLP is enabled in the mode of less intensive effect on a signal than by default;
  • voice nlp-option 2 – echo cancellers are enabled in the voice mode, non linear processor NLP is enabled in the mode of more intensive effect on a signal than by default;
  • off – do not use echo cancellation (set by default);

• DSCP for RTP – service type (DSCP) for RTP and UDPTL (T.38) packets;

  The DSCP setting for RTP and DSCP setting for SIP will be ignored while using VLAN for RTP transmission and signalling. Class of Service VLAN is used for prioritization in this case.

• RTP loss timeout – voice frequency path status control function that monitors the presence of RTP traffic from the communicating device. Permitted value range is from 10 to 300sec. When unchecked, RTP control is disabled; when checked, it is enabled. Control is performed as follows: if there are no RTP packets coming from the opposite device for the duration of the timeout and the last packet was not a silence suppression packet, the call will be rejected;
• RTP loss timeout after Silence-Suppression indication – RTP packet timeout for the silence suppression option utilization. Permitted value range is from 1 to 30. Coefficient is a multiplier that applies to the 'RTP packet timeout' value. Control is performed as follows: if there are no RTP packets coming from the opposite device for the duration of the timeout and the last packet was a silence suppression packet, the call will be rejected;
• RTCP period (sec) – time period in seconds (5-65535 s), after which the device send control packets via RTCP protocol. When unchecked, RTCP will not be used;
• RTCP activity control – voice frequency path status control function, may take up values in the range 2–255. Quantity of time periods (RTCP timer) during which the opposite party will wait for RTCP protocol packets. When there are no packets in the specified period of time, established connection will be terminated. At that, cause of disconnection 'cause 3 no route to destination' is assigned to the TDM and IP protocols. Control period value is calculated using the following equation: RTCP timer * RTCP control period seconds. When unchecked, feature will be disabled

Clear Channel – channel established for the transparent digital data transfer; when this channel is established, the device will not attempt to recode it and will transfer it transparently. To establish such a connection, reception of 'Transmission Medium Requirement' field is required with the following values:

• restricted digital info (Q.931 protocol);
• unrestricted dig.info (Q.931 protocol);
• video (Q.931 protocol);
• 64 kbit/s unrestricted (SS7 protocol);
• Clear Channel override – when checked, during 'clear channel' organization, a single codec CLEARMODE will be specified in SDP (if operation via Clear Channel was requested on the first call leg). When unchecked, the complete list of selected codecs will be always transferred to SDP in priority order.
• Clear Channel transit – mode that allows to transfer RTP directly from the incoming connection branch to the outgoing connection branch in SIP – SIP connection skipping internal switch buses of the device and preserving RTP traffic including packetization time.

Video processing:

• Video Offroad — video traffic passes transparently between clients;
• Video Transit — video traffic passes through the gateway.

Digital gain:

• Rx gain (0.1 dB) – volume of a receiving signal, amplification/attenuation of the level of signal received from an interacting gateway;
• Tx gain (0.1 dB) – volume of a transmitting signal, amplification/attenuation of the level of signal transmitted to an interacting gateway.

AGC (Auto Gain Control):

• Compliance with ITU-T G.169 – when the option is enabled, the automatic amplification operates in compliance with ITU-T G.169. The operation mode uses some algorithms different from the recommendations, which provide better background noise suppresion in the absence of speech.

Rx gain settings:

• AGC master enable – enable automatic amplification of receiving signals;
• Limit gain during doubletalk – limit a signal level if subscribers are talking simultaneously;
• Signal reference level, dBm0 – the level of the signal to which amplification will tend;
• Signal maximum gain, dB – the maximum permissible value of the amplification of an original signal;
• Signal minimum gain, dB – the minimum permissible value of the amplification of an original signal.

Tx gain settings:

• AGC master enable – enable automatic amplification of transmitting signals;
• Limit gain during double talk – limit a signal level if subscribers are talking simultaneously;
• Signal reference level, dBm0 – the level of the signal to which amplification will tend;
• Signal maximum gain, dB – the maximum permissible value of the amplification of an original signal;
• Signal minimum gain, dB – the minimum permissible value of the amplification of an original signal.

Dual-Tone Multi-Frequency transmit settings:

• DTMF transport – method of DTMF transmission via IP network:
  • inband – in RTP packets, inband.
  • RFC2833 – in RTP packets according to RFC2833 recommendation.
  • RFC2833/inband – automatic switching of the RFC2833 transmission mode to in-band and back, depending on the information regarding telephon-event support in the SDP from the other party;
  • SIP-INFO – outband, via SIP, INFO messages are used; at that, DTMF signal appearance will depend on the MIME extension type.

  • SIP-NOTIFY – NOTIFY messages are used via SIP protocol and out-of-band. This DTMF transmission is an implementation of the method that is used on Cisco equipment.

    In order to be able to use extension dialing during the call, make sure that the similar DTMF tone transmission method is configured on the opposite gateway.

    
    

• Flash signal processing (RFC2833) – checkbox that governs activation of FLASH signal processing using INFO, RFC2833, and re-invite methods for 'Call transfer' VAS operation;
• HOLD set/remove by¹ – selecting the method to set/remove a call on hold:
  • flash – set/remove a call on hold by pressing the Flash button on the phone handset;
  • flash/* – set/remove a call on hold by pressing the Flash button or the ‘*’ key on the phone handset;
  • flash/# – set/remove a call on hold by pressing Flash or ‘#’ on the phone handset;

  • flash/*/# – set/remove a call on hold by pressing Flash or ‘*’ or ‘#’ on the telephone handset.

    1 This option is available only for SIP profile mode.

• RFC2833 PT – type of payload used to transfer DTMF packets via RFC2833. Permitted values: 96 to 127. RFC2833 recommendation describes the transmission of DTMF via RTP protocol. This parameter should conform to the similar parameter of a communicating gateway (the most frequently used values: 96, 101);
• DTMF MIME Type – specify payload type used for DTMF transmission in SIP protocol INFO packets:
  • application/dtmf-relay – in SIP INFO application/dtmf-relay packets ('*' and '#' are sent as symbols '*' and '#');
  • application/dtmf – in SIP INFO application/dtmf packets ('*' and '#' are sent as digits 10 and 11).

Dual-Tone Multi-Frequency receive settings:

• DTMF receive transport:
  • inband – in RTP packets, in-band;
  • RFC2833 – in RTP packets, in accordance with RFC 2833;
  • SIP-INFO – out-of-band, via the SIP protocol, INFO messages are used; the type of DTMF signals transmitted will depend on the MIME extension type;
  • SIP-NOTIFY – out-of-band, via the SIP protocol, NOTIFY messages are used. This method of DTMF transmission is the implementation used on Cisco equipment.
• No inband DTMF removal – option is available when the DTMF in-band transmission method is enabled. If the option is disabled, and the SMG receives DTMF in two formats—for example, RFC2833 and in-band—the in-band format will be ignored, and only the RFC2833 format will be processed;
• Enable autodetect – when set, the first detected reception method is stored. All subsequent signals are received using only this method; the rest are ignored;
• RFC2833: same PT – when set and the SMG is the party that sent the SDP offer, RFC2833 packets are expected to be received with the PT value sent to us in the SDP answer; otherwise, RFC2833 packets are expected to be received with the PT value that the SMG sent in the SDP offer.

 Jitter buffer parameters:

• Mode – jitter buffer operation mode: static or dynamic;
• Minimum size, ms – size of fixed jitter buffer or lower limit (minimum size) of adaptive jitter buffer. Permitted value range is from 0 to 200 ms;
• Initial size, ms– initial value of adaptive jitter buffer. Permitted value range is from 0 to 200 ms;
• Maximum size, ms – upper limit (maximum size) of adaptive jitter buffer, in milliseconds. Permitted value range is from 'Minimum size' to 200 ms;
• Adaptation period, ms – time of buffer adaptation to the lower limit without faults in packet sequence order;
• Removal mode – buffer adjustment mode. Defines the method of packet deletion during buffer adjustment to lower limit.
  • Soft – device uses intelligent selection pattern for deletion of packets that exceed the threshold;
  • Hard – packets which delay exceeds the threshold will be deleted immediately.
• Removal threshold, ms – threshold for immediate deletion of a packet, in milliseconds. When buffer size grows and packet delay exceeds this threshold, packets will be deleted immediately. Permitted value range is from max size to 500 ms;
• Adjustment mode – select the adaptive jitter buffer adjustment mode for its increase (gradual/instant);
• Size for VBD, ms – size of a fixed jitter buffer used for data transmission in VBD mode (modem communication). Permitted value range is from 0 to 200 ms.

Codecs:

In this section, you may select codecs for an interface and an order of their usage on connection establishment. Codec with the highest priority should be placed in top position.

Click the left mouse button to highlight the row with the selected codec. Use arrow buttons  (up, down) to change the codec priority.

• On — when checked, use a codec specified in the adjacent field;
• Codec — codec, used for voice data transmission. Supported codecs: G.711A, G.711U, G.722, G.729A, G.729B, G.723.1, G.726-32;

  When VAD/CNG are enabled, G.729 codec operates as G.729B, otherwise as G729A, and G.723.1 codec operates with annex А support, otherwise without annex А support.

  
  

• PType — payload type for a codec. Field is available for editing only when G.726 codec is selected (permitted values: from 96 to 127, or 2 for negotiation with devices that does not support dynamic payload type for this codec). For other codecs, it is assigned automatically;
• PTE — packetization time — amount of voice data in milliseconds (ms), transmitted in a single packet.

"Fax/Modem settings" tab 

Call routing → SIP interfaces → Configuration →  → Fax/Modem settings

Data transmission:

• Enable VBD — when checked, create VBD channel according to V.152 recommendation for modem transmission. When CED signal is detected, the device enters Voice band data Deselect the checkbox to disable modem tone detection; at that, modem communication will not be affected (switching to modem codec will not be initiated, but such operation still may be performed by the opposite gateway);
• VCodec for VBD — codec, used for data transmission in VBD mode;
• Payload type for VBD — payload type, used for data transmission in VBD mode:
  
  • Static — use payload type standard values for a codec (for G.711A codec payload type is 8, for G.711U payload type is 0).
  • 96-127 — payload types from the dynamic range.

Fax settings: 

• Fax detector mode — detects transmission direction for fax tone detection and subsequent switching to fax codec:
  
  • no detect fax — disables fax tone detection, but will not affect fax transmission (switching to fax codec will not be initiated, but such operation still may be performed by the opposite gateway).
  • Caller and Callee — tones are detected during both fax transmission and receiving. During fax transmission, CNG FAX signal is detected from the subscriber's line. During fax receiving, V.21 signal is detected from the subscriber's line.
  • Caller — tones are detected only during fax transmission. During fax transmission, CNG FAX signal is detected from the subscriber's line.
  • Callee — tones are detected only during fax reception. During fax receiving, V.21 signal is detected from the subscriber's line.

    V.21 signal may also be detected from fax performing transmission.

• Fax relay mode — select protocol for fax transmission;
• Fax relay max rate (bps) — maximum transfer rate of fax transmitted via Т.38 protocol. This setting affects the ability of a gateway to work with high-speed fax units. If fax units support data transfer at 14400 baud, and the gateway is configured to 9600 baud, the maximum rate of connection between fax units and the gateway will be limited at 9600 baud. And vice versa, if fax units support data transfer at 9600 baud, and the gateway is configured to 14400 baud, this setting will not affect the interaction, maximum rate will be defined by the performance of fax units;
• Fax relay rate management — set the data transfer rate management method:
  • local TCF — method requires that the TCF tuning signal was generated locally by the recipient gateway. In general, used in T.38 transmission via TCP.
  • transferred TCF — method requires that the TCF tuning signal was sent from the sender device to the recipient device. In general, used in T.38 transmission via UDP.
• Т.38 data fill bits removal — padding bit removals and inserts for data that does not relate to ЕСМ (error correction mode);
• Т.38 data redundancy — redundancy amount in Т.38 data packets (amount of previous packets in the following Т.38 packet). Introduction of redundancy allows to restore the transmitted data sequence on reception when packets were lost during transmission;
• Т.38 data packetization — define Т.38 packet generation frequency in milliseconds (ms). This option allows to adjust the size of a transmitted packet. If the communicating gateway is able to receive datagrams with max. size of 72 bytes (maxdatagrammSize: 72), packetization time should be set to a minimum on SMG;
• Т.38 data transit — when the call is performed using two SIP interfaces and T.38 fax transfer protocol is used by both interfaces, this setting allows to transit T.38 packets between interfaces with a minimum delay.

'Service type' (IP DSCP) field value for RTP, T.38 and SIP/SIP-T/SIP-I: 

0 (DSCP 0x00, Diffserv 0x00) – standard forwarding (Best effort) – default value

8 (DSCP 0x08, Diffserv 0x20) – Class 1;

10 (DSCP 0x0A, Diffserv 0x28) – assured forwarding, low drop precedence (Class1, AF11);

12 (DSCP 0x0C, Diffserv 0x30) – assured forwarding, medium drop precedenceа (Class1, AF12);

14 (DSCP 0x0E, Diffserv 0x38) – assured forwarding, high drop precedence (Class1, AF13);

16 (DSCP 0x10, Diffserv 0x40) – Class 2;

18 (DSCP 0x12, Diffserv 0x48) – assured forwarding, low drop precedence (Class2, AF21);

20 (DSCP 0x14, Diffserv 0x50) – assured forwarding, medium drop precedence (Class2, AF22);

22 (DSCP 0x16, Diffserv 0x58) – assured forwarding, high drop precedence (Class2, AF23);

24 (DSCP 0x18, Diffserv 0x60) – Class 3;

26 (DSCP 0x1A, Diffserv 0x68) – assured forwarding, low drop precedence (Class3, AF31);

28 (DSCP 0x1C, Diffserv 0x70) – assured forwarding, medium drop precedence (Class3, AF32);

30 (DSCP 0x1E, Diffserv 0x78) – assured forwarding, high drop precedence (Class3, AF33);

32 (DSCP 0x20, Diffserv 0x80) – Class 4;

34 (DSCP 0x22, Diffserv 0x88) – assured forwarding, low drop precedence (Class4, AF41);

36 (DSCP 0x24, Diffserv 0x90) – assured forwarding, medium drop precedence (Class4, AF42)

38 (DSCP 0x26, Diffserv 0x98) – assured forwarding, high drop precedence (Class4, AF43);

40 (DSCP 0x28, Diffserv 0xA0) – Class 5;

46 (DSCP 0x2E, Diffserv 0xB8) – expedited forwarding (Class5, Expedited Forwarding).

IP Precedence:

0 – IPP0 (Routine);

8 – IPP1 (Priority);

16 – IPP2 (Immediate);

24 – IPP3 (Flash);

32 – IPP4 (Flash Override);

40 – IPP5 (Critical);

48 – IPP6 (Internetwork Control);

56 – IPP7 (Network Control).

Extended SIP settings

In this section, extended SIP settings are configured. These settings allow modifying SIP message fields using defined rules.

Call routing → SIP interfaces → Configuration →   → Extended SIP settings 

Field entry format

[sipheader:HEADER_NAME=operation],[sipheader:...],...

where:

• Operation – disable, insert, transit or modification rule;
• HEADER_NAME – case insensitive parameter, for example Accept = accept = ACCEPT. Other parameters are case sensitive.

Change of SIP headers:

• Disable – allow removing the selected headers from SIP messages.
  

  List of mandatory SIP message headers that must not be ignored or transit: via, from, to, call-id, cseq, contact, content-type, content-length.

  Example:

  Original headings: Accept: application/SDP User-Agent: TAU-8.IP X-UniqueTag: 12345678 90abcdef 12345678 90abcdef Modification rule: [sipheader:Accept=disable],[sipheader:user-agent=disable] Result: All SIP messages sent by the device via this SIP interface will be sent without the Accept and user-agent fields Modification rule: [unique-tag=disable] Result: All SIP messages sent by the device via this SIP interface will be sent without the X-UniqueTag.

  
  

• Insert – allow inserting the specified headers into SIP messages.
  Example:

  Modification rule: [sipheader:insert[LIST_OF_HEADERS]:RemoteIp=+(TEXT)] - Add the field in all queries: [sipheader:insert:RemoteIp=+(example.SMG)] - Only in INVITE: [sipheader:insert,INVITE:RemoteIp=+(example.SMG)] - Only in mentioned queries (f.e. INVITE and ACK): [sipheader:insert,INVITE,ACK:RemoteIp=+(example.SMG)] Result: Mentioned SIP messages sent by the device via this SIP interface will include a new RemoteIp:example.SMG field

  

• Transit – allow a group of headers from the first leg to be transferred without modification.

  To implement changes to SIP headers, enable the ‘SIP Header Transit’ option on the SIP interface from which the headers will be extracted.

  
  

  List of mandatory SIP message headers that must not be ignored or transit: via, from, to, call-id, cseq, contact, content-type, content-length.

  Example:

  Original headers on first leg: P-Asserted-Identity: sip: username@domain P-Called-Party-ID: sip: username@domain Privacy: id Subject: Test call Modification rule: [sipheader:[LIST_OF_MESSAGES]: [HEADER_MASK]=transit] [sipheader:[HEADER_MASK]=transit] - All headers, starting with "P-" [sipheader:P-=transit] Modification rule rule: [sipheader:=transit] will not work, because *characters can replace only the part of the name. Result: The specified headings will appear on the second leg: P-Asserted-Identity: sip:username@domain P-Called-Party-ID: sip:username@domain Modification rule: - In INVITE and 200 messages: [sipheader:INVITE,200:Subject=transit] - In any messages: [sipheader:Subject=transit] Result: The specified heading will appear on the second leg: Subject: Test call

  
  

• Modification rules – set of rules allow to modify selected headers in SIP messages. 
  

  Modification rules are described by the following characters:

  • $ – keep the text that follows;
  • ! – delete the remaining text;
  • +(ABC) – add the text specified;
  • -(ABC) – delete the text specified.
  
  

  For implementation examples of operation rules, see table 21 below.

  Table 21 — Implementation examples of operation rules

  Operation	Initial header	Rule	Result
  Add text at the beginning	Accept: application/SDP	[sipheader:accept=+(application/ISUP,)$]	Accept: application/ISUP,application/SDP
  Add text at the end	Accept: application/SDP	[sipheader:accept=$+(,application/ISUP)]	Accept: application/SDP,application/ISUP
  Delete text	Accept: application/SDP,application/ISUP	[sipheader:accept=-(application/SDP,)$]	Accept: application/ISUP
  Delete beginning from the specific place	Accept: application/SDP,text/plain	[sipheader:accept=-(text)!]	Accept: application/SDP
  Replace text completely	Accept: application/SDP	[sipheader:accept=+(application/ISUP)!]	Accept: application/ISUP
  Replace text	Accept: application/SDP,text/plain	[sipheader:accept=-(SDP)+(ISUP)$]	Accept: application/ISUP,text/plain
  Replace text, discarding data at the end	Accept: application/SDP,text/plain	[sipheader:accept=-(SDP)+(ISUP)!]	Accept: application/ISUP
  Complete the text	To: "Ivanov A.A." <sip:123@eltex>	[sipheader:to=-(eltex)+(eltexdomain.loc)$]	To: "Ivanov A.A." <sip:123@eltexdomain.loc>
  Example of a complex modification	From: <sip:who@host>;tag=aBc	[sipheader:from=+(DISPLAY )-(who)+(12345)-(>)+(;user=phone>)$+(;line=abc)]	From: DISPLAY <sip:12345@host;user=phone>;tag=aBc;line=abc
  Additional rules
  Transmit X-UniqueTag content in another title	X-UniqueTag: 12345678 90abcdef 12345678 90abcdef	[unique-tag=NewHeader-Name]	NewHeader-Name:  12345678 90abcdef 12345678 90abcdef
  Using TO instead of RURI for routing	Get:  Request-Line: INVITE sip:558018@10.22.128.36:5060 SIP/2.0 ... To: <sip:73852245673@10.22.1.50;user=phone>	[siprequest:cdpn=to]	Send: Request-Line: INVITE sip:73852245673@10.22.120.40:5060 SIP/2.0 ... To: <sip:73852245673@10.22.120.40;user=phone>
  Enable sending history-info in the redirected call		[siprequest:history=true]	Diversion will be replaced with History-Info when redirecting

  
  

Obtaining Display Name from a third-party server via LDAP

To set up receiving Display Name from a third-party server, it is necessary to add a setting in the form line in the menu item ‘Extended SIP settings’.

SMG polls the server(s) at a specified interval and stores the current name. When calling, names are requested for the initiator and destination. If there are no current ones in the database, then they are used default configured subscriber names (from SIP subscriber settings).

Configuration string format:

STRING:: ldap:ID:display:INTERVAL:DIRECTION:IP:PORT:LOGIN:PASSWORD:BASE[:ATTRPHONE:ATTRDISPLAY]

• ID – record identifier, for several interfaces there may be the same description, in this case the identifier should also be the same; in particular, it solves the issue duplication of records for sip profiles (when all users of the same profile will have the same record);
• INTERVAL – database update interval (minutes);
• DIRECTION – for which subscriber to use:
            • sip – value for From when calling from the SIP side and To when calling to the SIP side;
            • exchange – value for To when calling from the SIP side and From when calling to the SIP side;
            • *  – both names are requested in one paragraph.
• IP – LDAP server address;
• PORT – LDAP server port:
            • * – for shortness, it can be specified instead of the usual LDAP port 389.
• LOGIN – database username;
• PASSWORD – database user password;
• BASE – path to the server subscriber database;
• ATTRPHONE – attribute describing in the database the number by which the name will be searched. The parameter is optional, may not be specified: default value: telephoneNumber;
• ATTRDISPLAY – attribute describing DisplayName in the database. The parameter is optional, may not be specified, default value: displayName.

Configuration string format:

Full format: [ldap:L1:display:30:sip:192.168.23.187:389:cn=user,dc=smg,dc=com:userpassword:dc=smg,dc=com:telephoneNumber:displayName]
Short format: [ldap:L1:display:30:*:192.168.23.187:*:cn=user,dc=smg,dc=com:userpassword:dc=smg,dc=com]

Using user=phone in RURI

This feature allows setting or change the user= parameter in the RURI.
Setting:

[siprequest:user=phone]
[siprequest:user=ip] (instead of "ip" any value can be used, other than "phone").

Table 22 — Rules for adding the user=phone parameter to the RURI depending on the interface type

Prohibition of call transfer

This feature allow restricting call transfer if the Refer-To and RURI fields match.

Setting:

[siprequest:refer-back=disable]

For example, if the query contains the following headings

REFER sip:521@192.168.113.132:5060 SIP/2.0
...
Refer-To: <sip:521@192.168.113.132>
...

with advanced settings enabled, such a request will result in a 400 Bad Request response.

Advanced setting: ‘Replace 183+SDP with Alerting+inband’

This feature allows the 183 Session Progress response, containing an embedded SDP, to be transferred to the Alerting+inband channel on the PRI link.

Setting:

[sipresponse:183=alert]

H.323 interface

In this section, H.323¹ stack general configuration parameters, custom settings for each direction operating via H.323 protocol.

H.323 protocol is a signaling protocol used in VoIP applications for multimedia data transmission via packed-based data networks. It performs basic call management tasks such as starting and finishing session.

H.323 signaling is a stack of protocols based on the Q.931 recommendation implemented in ISDN. The gateway uses the following recommendations: H.225.0 and H.245.

SMG may operate within a method that may or may not feature the Gatekeeper. The separate license allows using SMG gateway as a gatekeeper and to interact with Directory gatekeeper for defining subscriber location. 

Call routing → H.323 interfaces

Common H323 settings:

• Device ID (H323 Alias) – gateway name during registration at the Gatekeeper;
• Port for signaling – a network interface for H.323 signaling;
• Signaling Receive Port – local TCP port for receiving H.323 signaling messages.

GateKeeper settings:

• GateKeeper – defines the mode of gatekeeper operation. In the ‘remote’ mode, SMG interacts with external gatekeeper. In the ‘local’ mode, SMG operates as a gatekeeper.

Settings for ‘remote’ mode:

Call routing → H.323 interfaces → Remote mode

• Search GateKeeper – when checked, automatic Gatekeeper discovery method will be used in multicast mode using IP address 224.0.1.41 and UDP port 1718, otherwise this method will not be used and the Gatekeeper will have a specific IP address;
• GateKeeper IP – identification of the gatekeeper at the specific IP;
• GateKeeper Port – gatekeeper UDP port (port 1719 is used by the majority of gatekeepers by default);
• Registration time – time period in seconds, for which the device will keep its registration on a gatekeeper;

• Keep-alive timeout – time period in seconds, after which the device will renew its registration on a gatekeeper.

  To reliably re-register a device to Gatekeeper, the re-registration period value ‘Keep Alive Time’ should be set to 2/3 of the ‘Time To Live’ registration period. In this case, it is recommended to configure the ‘Time To Live’ parameter the same as on Gatekeeper, so that the value of the ‘Keep Alive Time’ gateway re-registration period was not greater than or equal to the ‘Time To Live’ value sent in Gatekeeper responses. Otherwise, incorrect configuration may cause Gatekeeper to remove registration from the gateway before the gateway re-registers, which in turn lead to termination of all active connections, established through the gatekeeper.

  
  

  When settings are applied in this section, H.323 will be restarted and all established H.323 voice connections will be forcibly terminated, also H323-MODULE LOST fault may appear shortly

Settings for ‘local’ mode¹:

Call routing → H.323 interfaces → Local mode

• GateKeeper – identifier of local Gatekeeper operating on SMG;
• Default technology prefix – defines the default directions to which the GateKeeper will transmit calls returned from Directory GateKeeper and not intended for SMG SIP subscribers. The direction must be registered on a local GateKepper of SMG;
• DSCP for RAS – type of service (DSCP) for signaling traffic (H.323 RAS);
• Primary Directory Gatekeeper and Secondary Directory Gatekeeper – settings for interaction with a main and redundant Directory GateKeepers;
• H.323 ID – identifier of Directory Gatekeeper;
• IP address – IP-адрес Directory Gatekeeper.

The interaction of local GateKeeper and Directory GateKeeper is performed as follows: While egress call: SMG transmits location request (RAS LRQ) to Directory GateKeeper. Directory GateKeeper defines the subscriber location and transmits its signal address in location confirm message (RAS LCF). If the Directory GateKeeper cannot define the location, the call will be released with the location reject message (RAS LRJ). While ingress call: Directory GateKeeper transmits location request (RAS LRQ) to SMG. If the callee is a subscriber of SMG, SMG transmits its signal address in location confirm message (RAS LCF).  If the callee is not a subscriber of SMG, but has a registered technology prefix, SMG transmits a signal address of a device which registered this prefix in location confirm (RAS LCF). If there is no registered prefix, SMG releases the call with location reject message (RAS LRJ).

'H.323 interface settings' tab

Call routing → H.323 interfaces → → H.323 interface settings

• Name — interface name;
• TrunkGroup — select a trunk group, that the interface belongs to;
• Access category — select access category;
• Dial plan — define dial plan that will be used for dialing from this interface (necessary for dial plan negotiation);
• Use GateKeeper — when checked, the current interface will interact with the GateKeeper which settings are specified in Common H323 settings;

• Search for alternative routes — allows calls to be routed between different VLANs (mutually exclusive with the ‘Cisco 1700 Adaptation’ option);

• Host name/IP-address — IP address or name of the host communicating via gateway H.323 protocol;
• Port for signaling — signaling TCP port of the communicating gateway used for H323 signaling reception;
• Network interface for RTP — select network interface for voice traffic transmission and reception;
• Scheduled routing profile — select 'Scheduled routing' service profile, configured in the Internal resources section;
• Max active calls — maximum number of simultaneous (incoming and outgoing) connection through the interface specified.

'H.323 protocol settings' tab

Call routing → H.323 interfaces → → H323 protocol settings

• Device ID (H323 alias) — gateway name during registration at the Gatekeeper;
• Fast start — when checked, fast start function is enabled, otherwise it is disabled. When option is used, session description for media channel establishing is performed via Н.225 protocol, otherwise via H.245 protocol;
• H245-tunnel — when checked, H.245 signaling tunneling is enabled through the Q.931 signal channel, otherwise it is disabled;
• Name coding:
  
  • Transit — no recoding is performed (by default, the name is assumed to be accepted in UTF-8);
  • CP 1251 — coding of Windows-1251;
  • Siemens adaptation — coding of АТС Siemens;
  • AVAYA adaptation — coding of PBX AVAYA;
  • Latin transliteration — Russian names will be transliterated into Latin letters.
• Name transmission:
  
  • Q931 DISPLAY — transmission in Q.931 Display element with Codeset 5;
  • AVAYA DISPLAY – transmission in Q.931 Display element with Codeset 6;
  • QSIG-NA — transmission via QSIG-NA protocol (ECMA-164).

• CISCO 1700 adaptation:

  • Bandwidth for Adminssion Request is 64000;

  • For outgoing calls:

    • remote alias with CgPN value;
    • local alias with H.323 ID Primary Directory Gatekeeper value;
    • In addition to the above, a local alias with the value ‘Device ID (Alias)’ from the general H.323 configuration is added.
  • No search for an alternative H.323 interface takes place during an incoming call.

• DSCP for Signaling – service class (DSCP) for signalling traffic (H.323);

  The DSCP value for Signalling must be the same for all directions with the same transport parameter (network interface + signalling receive port). The DSCP for RTP and DSCP for SIP settings will be ignored while using VLAN for RTP transmission and signalling. Class of Service VLAN will be used for traffic prioritization in this case.

  
  

• Number prefixes (Prefix 1, Prefix 2, Prefix 3) – numbers, which SMG register on a Gatekeeper according to settings – local or remote. The table is filled with the numbers or initial digits of numbers of SIP subscribers registered on SMG in order to gatekeeper could forward calls to SMG (for example, it is sufficient to write the same prefix 10010 for subscribers with numbers 100101 and 100102).

'RTP/codec configuration' tab

Call routing → H.232 interfaces → → Codecs/RTP settings

Options:

• Voice activity detector / Comfort noise generator (VAD/CNG) — when checked, silence detector and comfort noise generator are enabled. Voice activity detector disables transmission of RTP packets during periods of silence, reducing loads in data networks.
• Source IP: Port verification — when this setting is checked, control of media traffic received from IP address and UDP port specified in SDP communication session description will be enabled; otherwise the traffic from any IP address and UDP port will be accepted;
• Echo cancellation — echo cancellation mode:
  • voice (default) — echo cancellers are enabled in the voice data transmission mode;
  • voice nlp-off — echo cancellers are enabled in voice mode, non-linear processor (NLP) is disabled. When signal levels on transmission and reception significantly differ, weak signal may become suppressed by the NLP. Use this echo canceller operation mode to prevent the signal suppression;
  • modem — echo cancellers are enabled in the modem operation mode (direct component filtering is disabled, NLP control is disabled, CNG is disabled);
  • voice nlp-option 1 – echo cancellers are enabled in the voice mode, non linear processor NLP is enabled in the mode of less intensive effect on a signal than by default;
  • voice nlp-option 2 – echo cancellers are enabled in the voice mode, non linear processor NLP is enabled in the mode of more intensive effect on a signal than by default;
  • off — do not use echo cancellation (this mode is set by default).
• Rx gain (0.1 dB) — volume of signal received, gain of the signal received from the communicating gateway;
• Tx gain (0.1 dB) — volume of signal transmitted, gain of the signal transmitted to the communicating gateway direction;
• DSCP for RTP — service type (DSCP) for RTP and UDPTL (T.38) packets.

• The DSCP for RTP and DSCP for SIP settings will be ignored while using VLAN for RTP transmission and signalling. Class of Service VLAN is used for traffic prioritization in this case.

• RTP-loss timeout – the function that controls the presence of RTP traffic from interacting device on a voice-frequency path. The permissible values are from 10 to 300 seconds. When unchecked, RTP control is disabled, when checked – enabled. The control is implemented as follows: if during the set timeout there is no RTP packets received and the last packet was not the packet of pause suppression, the call will be released;
• RTP-loss timeout after Silence-Suppression indication (coefficient) – timeout for RTP packets when using the option of pause suppression. The permissible values are from 1 to 30. The coefficient defines how many times this value greater than RTP-loss timeout. The control is implemented as follows: if there is no RTP packets received and the last packet was the packet of pause suppression, the call will be released;
• RTCP period (sec) — time period in seconds (5–65535), after which the device sends control packets via RTCP protocol. When unchecked, RTCP will not be used;
• RTCP activity control — voice frequency path status control function, may take up values in the range 2–255 seconds. Quantity of time periods (RTCP timer) during which the opposite party will wait for RTCP protocol packets. When there are no packets in the specified period of time, established connection will be terminated. At that, cause of disconnection 'cause 3 no route to destination' is assigned to the TDM and IP protocols. Control period value is calculated using the following equation: RTCP timer* RTCP control period When unchecked, feature will be disabled.

Dual-Tone Multi-Frequency (DTMF) signaling setting:

• DTMF transport — a method of DTMF transmission via IP network;
  • inband — inband, in RTP voice packets;
  • RFC2833 — according to RFC2833 recommendation, as a dedicated payload in RTP voice packets;
  • 245-ALPHANUM — outband; in Н.245 userInput messages, basicstring compatibility is used for DTMF transmission;
  • 245-SIGNAL — outband; in Н.245 userInput messages, dtmf compatibility is used for DTMF transmission;
  • Q931-KEYPAD— outband; Keypad information element is used for DTMF transmission in Q.931 INFORMATION message.

    In order to be able to use extension dialing during the call, make sure that the similar DTMF tone transmission method is configured on the opposite gateway.

• RFC2833 PT — type of payload used to transfer DTMF packets via RFC2833. Permitted values: 96 to 127. RFC2833 recommendation describes the transmission of DTMF via RTP protocol. This parameter should conform to the similar parameter of a communicating gateway (the most frequently used values: 96, 101).
• RFC2833: same PT – when checked, if SMG is an initiating side of connection, RFC2833 packets with PT value which has been transmitted by OpenLogicalChannelAck, are expected to be received. Otherwise, the RFC2833 with the PT value, which has been transmitted in OpenLogicalChannelAck request by SMG, are expected to be received.

Jitter buffer settings:

• Mode — jitter buffer operation mode: static or dynamic;
• Minimum size, ms — size of fixed jitter buffer or lower limit (minimum size) of adaptive jitter buffer. Permitted value range is from 0 to 200 ms;
• Initial size, ms — initial value of adaptive jitter buffer. Permitted value range is from 0 to 200ms.
• Maximum size, ms — upper limit (maximum size) of adaptive jitter buffer, in milliseconds. Permitted value range is from 'Min size' to 200 ms;
• Adaptation period, ms — time of buffer adaptation to the lower limit without faults in packet sequence order;
• Removal mode — buffer adjustment mode. Defines the method of packet deletion during buffer adjustment to lower limit:
  • Soft — device uses intelligent selection pattern for deletion of packets that exceed the threshold;
  • Hard — packets which delay exceeds the threshold will be deleted immediately.
• Removal threshold, ms — threshold for immediate deletion of a packet, in milliseconds. When buffer size grows and packet delay exceeds this threshold, packets will be deleted immediately. Permitted value range is from 'Max size' to 500 ms.
• Adjustment mode — select the adaptive jitter buffer adjustment mode for its increase (gradual/instant);
• Size for VBD, ms — size of a fixed jitter buffer used for data transmission in VBD mode (modem communication). Permitted value range is from 0 to 200 ms. 

Codecs:

In this section, codecs for an interface and an order of their usage on connection establishment can be selected. Codec with the highest priority should be placed in top position.

Click the left mouse button to highlight the row with the selected codec. Use arrow buttons(up, down) to change the codec priority.

• 
  

  On — when checked, use a codec specified in the adjacent field;

• Codec — codec, used for voice data transmission. Supported codecs: G.711A, G.711U, G.729A, G.729B, G.723.1.

  When VAD/CNG are enabled, G.729 codec operates as G.729B, otherwise as G729A, and G.723.1 codec operates with annex А support, otherwise without annex А support.

• PType — payload type for a codec. Field is available for editing only when G.726 codec is selected (permitted values: from 96 to 127, or 2 for negotiation with devices that does not support dynamic payload type for this codec). For other codecs, it is assigned automatically;
• PTE — packetization time — amount of voice data in milliseconds (ms), transmitted in a single packet.

'Fax/Modem settings' tab

Call routing → H.232 interfaces → → Fax/Modem settings

Modem settings:

• Enable VBD — when checked, create VBD channel according to V.152 recommendation for modem transmission. When CED signal is detected, the device enters Voice band data Deselect the checkbox to disable modem tone detection; at that, modem communication will not be affected (switching to modem codec will not be initiated, but such operation still may be performed by the opposite gateway);
• Codec for VBD — codec, used for data transmission in VBD mode;
• Payload type for VBD — payload type, used for data transmission in VBD mode;
  • Static — use payload type standard values for a codec (for G.711A codec payload type is 8, for G.711U payload type is 0);
  • 96-127 — payload types from the dynamic range.

Fax settings:

• Fax detector mode — detects transmission direction for fax tone detection and subsequent switching to fax codec:
  • no detect fax — disables fax tone detection, but will not affect fax transmission (switching to fax codec will not be initiated, but such operation still may be performed by the opposite gateway);
  • Caller and Callee — tones are detected during both fax transmission and receiving. During fax transmission, CNG FAX signal is detected from the subscriber's line. During fax receiving, V.21 signal is detected from the subscriber's line;
  • Caller — tones are detected only during fax transmission. During fax transmission, CNG FAX signal is detected from the subscriber's line;
  • Callee — tones are detected only during fax reception. During fax receiving, V.21 signal is detected from the subscriber's line.

    V.21 signal may also be detected from fax performing transmission.

    
    

• Fax relay mode — select protocol for fax transmission;
• Fax relay max rate (bps) — maximum transfer rate of fax transmitted via Т.38 protocol. This setting affects the ability of a gateway to work with high-speed fax units. If fax units support data transfer at 14400 baud, and the gateway is configured to 9600 baud, the maximum speed of connection between fax units and the gateway will be limited at 9600 baud. And vice versa, if fax units support data transfer at 9600 baud, and the gateway is configured to 14400 baud, this setting will not affect the interaction, maximum speed will be defined by the performance of fax units;
• Fax relay rate management — set the data transfer speed management method:
  • local TCF — method requires that the TCF tuning signal was generated locally by the recipient gateway. In general, used in T.38 transmission via TCP;
  • transferred TCF — method requires that the TCF tuning signal was sent from the sender device to the recipient device. In general, used in T.38 transmission via UDP.
• 38 data fill bits removal and insertion — padding bit removals and inserts for data that does not relate to ЕСМ (error correction mode);
• 38 data redundancy — redundancy amount in Т.38 data packets (amount of previous packets in the following Т.38 packet). Introduction of redundancy allows to restore the transmitted data sequence on reception when packets were lost during transmission;
• 38 data packetization — define Т.38 packet generation frequency in milliseconds (ms). This option allows to adjust the size of a transmitted packet. If the communicating gateway is able to receive datagrams with max. size of 72 bytes (maxdatagrammSize: 72), packetization time should be set to a minimum on SMG;
• Т.38 data transit — when the call is performed using two VoIP interfaces and T.38 fax transfer protocol is used by both interfaces, this setting allows to transit T.38 packets between interfaces with a minimum delay.

Trunk directions

Trunk direction is a set of trunk groups. For a call to a trunk direction, you may specify the selection order for trunk groups comprising this direction.

Call routing → Trunk Directions

To create, edit or remove trunk directions, use 'Objects’ —  'Add object', 'Objects’ —  'Edit object' and 'Objects’ —  'Remove object' menus and the following buttons:

– 'Add direction'

– 'Edit direction parameters'

– 'Remove direction'

Call routing → Trunk Directions →  

• Name — trunk direction name;
• TrunkGroup select mode — trunk group selection order in the direction:

• • Successive forward — all trunk groups comprising the direction are selected in turns beginning from the first in the list;
  • Successive backward — all trunk groups comprising the direction are selected in turns beginning from the last in the list;
  • Starting from first forward — the first free trunk group comprising the direction is selected beginning from the first in the list;
  • Starting from last backward — the first free trunk group comprising the direction is selected beginning from the last in the list.
• Local direction — when checked, subscribers of this direction are considered as local. Subscribers in this direction are placed under SORM control with the type and sign of the number ‘subscriber of this station’.

List of trunk groups in direction:

Call routing → Trunk Directions → → Trunk Direction settings # 0 →

To add or remove trunk groups, use the following buttons:

– 'Add'

– 'Remove'

Use arrow buttons (up, down) to change the trunk group order in the list.

V5.2 interfaces

The menu is dedicated to V5.2 interface parameters configuration. To add a new interface into the configuration, clickin the left screen part with highlighted ‘V5.2 Interfaces’. The quantity of created interfaces should be equal to the quantity of outstations.

Interface settings

Call routing → V5.2 interface → Interface selection → Interface settings

• Name — displayed interface name;
• Primary E1 stream — primary stream for V5.2 interface;
• Secondary E1 stream — secondary stream for V5.2 interface;
• Interface ID — interface identifier;
• Variant ID — provision option in the initial configuration;
• C-chan ID — logical C-channel identifier;
• PSTN link — stream number to which the PSTN protocol will be assigned;
• PSTN ts — CI number to which the PSTN protocol will be assigned;
• Link identification — checking the compliance of E1 ID paths on the LE and AN sides during launching the interface;
• Accelerated port alignment — using an accelerated port unlocking mechanism (Accelerated Port Alignment) during interface startup. Possible accelerated port alignment parameters:
  • PSTN&ISDN – PSTN and ISDN port alignment;
  • PSTN — PSTN port alignment.
• Alarms — when checked, the alarm message will be displayed;
• RADIUS profile — selection of RADIUS prodile for the interface.

– ‘Add E1 stream’

When adding a new E1 stream it is necessary to specify its LinkID in the filed opposite the drop-down streams list.

To change the order of E1 streams in the list, use thearrows (down, up).

Subscribers list

This section is intended for binding created V5.2 subscribers to this V5.2 interface. Each cell for a subscriber contains a “Layer 3 address”, which is unique within one interface.

Routing → V5.2 interfaces → Object → Subscribers list

• № — subscriber sequence number;
• L3 Address — Layer 3 address required for subscriber identification within V5.2 interface;
• Suscriber ID — unique subscriber identifier;
• Subscriber name — subscriber name;
• Subscriber number — subscriber phone number.

 To edit the list, use the buttons:

• Add — add V5.2 subscriber;
• Swap selected — swap places of two selected subscribers;
• Clear selected — delete the nuber binding (cell content);
• Delete selected — delete the subscriber (the cell completely).

SIP-Trunk Registrations

Settings

SIP-Trunk Registrations → Registrations → Settings

Configuring subscriber registration and authentication parameters for interfaces with subscriber registration type.

Registration settings:

• Login — name used for authentication;
• Password — password for authentication;
• Username/Number — user number registered in the SIP domain;
• SIP-domain — the domain in which the subscriber is registered on the upstream server.

In the list of SIP interfaces, registration binding to a specific SIP interface is assigned/removed. This allows one to define a list of subscribers who are allowed to make calls via this interface.

Monitoring

When selecting the ‘Monitoring’ item in the drop-down list, a table for monitoring is displayed subscriber registration on the upstream server.

SIP-Trunk Registrations → Registrations → Monitoring

• Login — name used for authentication;
• Username/Number — user number registered on the upstream server;
• SIP interface list — list of interfaces through which the subscriber access is allowed to;
• Status — subscriber registration status (registered, not registered, registration expired);
• Reason — possible reason for lack of registration;
• Expire in — time remaining until registration expires.

Subscribers

The menu is intended to configure the parameters of PRI, V5.2, SIP subscribers¹, as well as for PBX and PRI profiles.

SIP Subscribers

Configuration of SIP subscribers

Subscribers → SIP Subscribers → Configuration

Search subscriber — checking the presence of a subscriber in the database of configured SIP subscribers, it is possible to check by name, number, callerID, IP address: Port, SIP domain, SIP profile, PBX profile and dial plans;

Edit selected — pressing the button takes you to the group menu editing parameters of selected subscribers (opposite to which the flag is set ‘Select’). To be able to edit, set the ‘Edit’ flag opposite the required parameter. A description of the parameters for configuration is given below;

Remove selected — pressing the button allows to delete a group of selected subscribers.

Import – import a subscriber database from a .json file. More details in Appendix P. Example of SMG configuration;

Export – save the subscriber database to a .json file. More details in Appendix P. Example of SMG configuration.

To create, edit and delete an individual subscriber record, use the ‘Objects’ menu – ‘Add object’, ‘Objects’ – ‘Edit object’ and ‘Objects’ – ‘Delete object’, and also buttons:

– 'Add subscriber’

– 'Edit subscriber parameters'

– 'Remove subscriber’

'SIP Subscribers' tab

Subscribers → SIP subscribers → Configuration →

• Subscribers count – number of subscribers;
• Starting description – free text description of subscribers;
• Starting number – subscriber number for a group of subscribers, each subsequent subscriber will be assigned a number increased by one;
• Starting CallerID number – subscriber’s Caller ID number, for a group of subscribers to each subsequent number increased by one will be assigned;
• Use CallerID number for redirection – when using redirections, the caller ID number will be substituted into the Diversion or Redirecting number fields instead of the subscriber number;
• Calling party number type – subscriber number type;
• Calling party category – CallerID category;
• Lines operation mode – operating mode limiting the number of simultaneous calls. It can take two values: ‘Combined’ and ‘Separate’. In the first mode, the total number of simultaneous calls involving the subscriber, in the second mode incoming and outgoing calls are counted separately;
• Lines number – number of simultaneous calls involving the subscriber. This field is displayed if the ‘Combined’ line operation mode is selected. Acceptable values [1;255] or 0 – no restrictions;
• Ingress lines number¹ – number of simultaneous incoming calls to a subscriber. The field is displayed if the ‘Separate’ line operation mode is selected. Acceptable range [1;255] or 0 – no restrictions;
• Egress lines number¹ – number of simultaneous outgoing calls from a subscriber. The field is displayed if the ‘Separate’ line operation mode is selected. Acceptable range [1;255] or 0 – no restrictions;
• IP-address: port – IP address and port of the subscriber. When setting the value 0.0.0.0, the subscriber is allowed to register from any IP address. Setting the port to zero ignores the port with which registration comes;
• Allow unregistered calls – option becomes active only if in the ‘IP-address:Port’ both the address and port of the subscriber are specified. When the flag is set, the subscriber will be able to do calls without prior registration from the specified IP and port. This option does not work if sip profile ‘Any’ is selected;
• SIP domain – determines whether a subscriber belongs to a specific domain. Sent by the gateway subscriber in the ‘host’ parameter of the SIP URI scheme of the from and to fields;
• SIP profile – selecting a SIP profile. The SIP profile determines most of the subscriber's settings. If you select the ‘Any’ profile, this will make it possible to register a SIP subscriber to any from the available SIP profiles in the system (see SIP/SIP-T/SIP-I, SIP profiles section);
• РВХ profile – PBX profile selection (see PBX profiles);
• Dial plan – dial plan in which the subscriber will be located;
• Authorization – authentication mode for the device:
  
  • Not set – authentication is disabled;
  • With REGISTER – authentication is carried out only during registration — upon REGISTER request;
  • With REGISTER and INVITE – authentication is carried out both during registration and during making outgoing calls – based on REGISTER and INVITE requests. In this mode, the subscriber will be able to make a call using an IP address and port different from those used for registration.
• Login – username for authentication;
• Password – password for authentication. When creating a subscriber group, you can specify an algorithm for generating unique passwords using the following commands:
  • %CMD+N% – sequence with a password value that increases by N.
  • %CMD-N% – sequence with a password value that decreases by N.
  • %CMDRAND{N}[abc]% – sequence of N random characters from the alphabet abc, as specified by the administrator. If the value of N is not specified, a random sequence of 8 characters from the specified alphabet will be generated. If neither N nor the alphabet is specified, a sequence of 8 random characters will be generated. 

Example:

%CMD+N%

qwe6453%CMD+1%

qwe6453
qwe6454
qwe6455
qwe6456

0secret
1secret
2secret
3secret

%CMD-N%

pass999%CMD-1%

pass999
pass998
pass997
pass996

cred0
cred-1
cred-2
cred-3

1a2s3d%CMDRAND3%

%CMDRAND4[*#951753]%

9#55
59#*
*319
5153

%CMDRAND%

7JRzedHg
D0594e&Y
*R$&GX(-
hK4KBj3g

• Ignore source port after registration – after registration, the messages from subscribers can come from any port of the registered address;
• Subscriber service mode – sets a limit on incoming and outgoing communications to the subscriber:
  • off: The subscriber number will be present in the dial plan, but the subscriber's terminal will not be able to register. Respectively incoming calls will be rejected with the ‘out of order’ reason, outgoing calls will not be able to be initiated;
  • on: disabled, all types of communication are available;
  • off 1: there is incoming communication, outgoing communication is only to special services;
  • off 2: there is no incoming communication, outgoing communication is only to special services;
  • denied 1: complete deny on incoming and outgoing traffic. Calls will be routed via dial plan, but will be rejected;
  • denied 2: complete deny on incoming and outgoing traffic except for special services;
  • denied 3: incoming calls are prohibited, outgoing calls are allowed;
  • denied 4: incoming calls are prohibited, outgoing calls are allowed only within local and

  • departmen networks;

  • denied 5: incoming calls are allowed, outgoing calls are completely prohibited;

  • denied 6: incoming calls are allowed, outgoing calls are allowed only to special services;

  • denied 7: incoming calls are allowed, outgoing calls are allowed for local and department networks;

  • denied 8: incoming calls are allowed, outgoing calls are allowed only within the local, department and zone networks;

  • denied 9: incoming calls are allowed, outgoing calls are allowed only within the local, department, zone and long-distance networks;
  • ignore: excluded from numbering. The number is completely excluded from subscriber numbers of a dial plan. When calling this number, the call will be rejected due to no route to destination or will go to a suitable prefix in the dial plan.
• Display name – name that will be passed to display-name. The parameter also affects using display-name as the Connected Name in responses to calls to the subscriber;
• Use display name – mode of using display-name (SIP display-name). It can take the values:
  • Received only – ‘Display name’ setting will not be used, display-name will always take the value that was in the initiating INVITE;
  • Received prefer – if a subscriber receives a call initiation request without a display-name, then the display-name will be filled in with what is configured for SMG. Otherwise, the received display-name will be used;
  • Configured only – regardless of what came in the subscriber's request, display-name set to SMG will be used.
• Authentication with qop – add a quality of protection to digest authentication.

Multiple registration (SIP-forking)

Multiple registration of up to five clients on one account is allowed. Registration is possible as on the same or on different network interfaces. The call goes to all registered contacts simultaneously. Working with priorities (q-parameter) will be implemented in subsequent versions.

SIP-forking – enabling multiple registration on a subscriber;

Max registered contacts number – allowed valid registration range on one subscriber (Range of acceptable values [2; 5]).

Subscriptions:

Subscribe to BLF events – разрешает абоненту подписываться на события BLF других абонентов;

Subscribe to Presence events – разрешает абоненту подписываться на статус регистрации (доступности) других абонентов.

Intercom call settings:

• Intercom call type – type of incoming intercom call (auto answer call of subscriber B):
  • One-way – with an incoming intercom call, subscriber B will hear subscriber A, but subscriber A will not hear subscriber B (one-way notification);
  • Two-way – with an incoming intercom call, both subscribers will hear each other;
  • Ordinary call – incoming intercom call will be made as normal without auto answer of B side;
  • Ignore – incoming intercom call will be rejected.
• Intercom call priority – priority of incoming intercom call over others calls. 
  • ordinary call – priority 1;
  • Intercom call can be defined with the priority of 1–5, by default – 3;
  • notification – 7.

Examples:

• If subscriber A with priority 1 calls an already busy subscriber B (with one line and any priority), then subscriber A will hang up;

• If subscriber A with priority 2 calls already busy subscriber B (with one line and any priority), then 1 more extension line will be allocated for subscriber B and subscriber B will receive a call notification from subscriber A; В случае если на ТА поддержан и настроен приоритет интерком-вызова, вызов абонента А будет автоматически принят, а уже существующий поставлен на удержание;

• If subscriber A with priority 2 calls already busy subscriber B (with one line and any priority), but subscriber B is already busy with subscriber C with priority 3, then subscriber A will hang up;

• Subscriber A should be notified in any case, because subscriber A has more high priority 7.

• Intercom SIP header – Intercom SIP-header, that will be transmitted to the subscriber in the INVITE message during intercom/paging call:

Answer-Mode: Auto;

Alert-Info: Auto Answer;

Alert-Info: info=alert-autoanswer;

Alert-Info: Ring Answer;

Alert-Info: info=RingAnswer;

Alert-Info: Intercom;

Alert-Info: info=intercom;

Call-Info: =\;answer-after=0;

Call-Info: \\;answer-after=0;

Call-Info: ;answer-after=0;

• Pause before answer, sec – transmission of pause time before answering intercom/pagingcall in the “answer-after” parameter.

VAS settings:

CLIRO – service to overcome the ban on issuing a caller's number;

Enable VAS¹ – connection of VAS services for the subscriber. Upon selecting this item, the table ‘VAS activation’ will become available.

RingBack settings:

Allows configuring audio file playback individually. 

Modes:

Default – refer to the settings in the system parameters;

RingBack – play the default Ringback, ignoring the settings in the system parameters;

Audio file – replacing the standard Ringback sound with a sound of your choice, which was uploaded during the Ringback configuration stage under the ‘System Settings’ menu. 

Voice Notification:

Launching a notification – enable subscribers to run VNS tasks;

Confirmation code – set a personal code to confirm receipt of an alert;

Operator access category – select a category that allow/deny subscriber access to the relevant tasks and VNS reports.

VAS activation

Subscribers → SIP Subscribers → Configuration → → Enable VAS

• Call Forward (Unconditional) – enables the Call Forwarding Unconditional (CF Unconditional) service;
• Call Forward (Busy) – enables the Call Forwarding Busy (CF Busy) service;
• Call Forward (No Reply) – enables the Call Forwarding No Reply (CF No Reply) service;
• Call Forward (Out of Service) – enables the Call Forwarding Out of Service (CF Out of Service);
• Call Forward (Time) – enables the Call Forwarding by time;
• Call hold;
• Call transfer – enables the Call Transfer service;
• 3WAY conference;
• Call pickup;
• Conference;
• Disconnect conference by initiator – when checked, the conference will be over when the initiator leaves the conference. Otherwise, the conference will be saved after the initiator is hung up and will be over only when the last participant leaves the conference;
• Intercom/Paging – activation of access to the outgoing intercom or paging call service (call with autoreply of party B);
• Change password – changing the password to restrict outgoing communications;
• Outgroing calls restriction – use the service ‘restricting outgoing communications by password’;
• Restricted by password – allows the subscriber to make a one-time call without restrictions communication by entering the VAS password;
• Password activation – allows the subscriber to enter a password once to remove the outgoing communication restriction. Re-entering the password again sets the restrictions;
• DND – allows the subscriber to set ‘Do Not Disturb’ mode and specify a few numbers on a whitelist that will still be able to call them¹;
• Black list – allows subscribers to block numbers so that they cannot call them¹;
• Call park to – allows the subscriber to use the call hold service;
• Slot setting – allows a subscriber with the “Call Park” service activated to place subscribers in a slot;
• Extraction from slot – allows subscribers with the ‘Call Park’ service activated to pick up subscribers from slots;;
• Voice mail;
• One Touch Record;
• Anonymous call – allows making anonymous calls without revealing phone number or caller ID;
• Reject anonymous calls – allows rejecting anonymous calls (calls without a phone number or caller ID);
• Reminder – allows receiving an incoming call on your phone at a specified time. The subscriber activates the service and sets the time for it to be triggered. At the appointed time, the system places a call to the subscriber. When the subscriber picks up the phone, an alarm tone is played;
• Call waiting;
• Do not disturb in call group;
• Autoredial – allows to automatically redial a subscriber who is currently busy or unavailable;
• Autoredial with callback – allows automatic redialling of the subscriber, followed by a call back to the subscriber who initiated the call;
• Reset all services – allows cancelling all configured numbers for call forwarding by dialling the service prefix configured in the numbering plan.

Additional numbers

A subscriber may have different numbers in different dial plans, and when a call is passing through the dial plan change prefix, the subscriber's CgPN number is automatically replaced with the number in the appropriate dial plan, for example:

The subscriber has an internal short numbering; accordingly, it is registered at the gateway under a short number, when accessing an external network, each such subscriber should enter its own number in international format as a CgPN. Access to the external network is via prefix 9.

To solve this problem, it is necessary to activate 2 dial plans in the ‘System parameters’ section, create a list of subscribers with short numbering on the gateway, indicate its external number in the ‘Additional numbers’ in the ‘Dial plan #1’ field for each subscriber. In the dial plan #1, a prefix for accessing the external network should be created; in the ‘dial plan # 0’ a prefix ‘(9x.)’ with ‘change dial plan’ type should be created, which will switch to the dial plan #1. When a subscriber dials a full number with 9 at the beginning, the call will go through the “Change a dial plan” prefix type, and in the dial plan 1 the CgPN number will be automatically replaced with its external number.

Subscribers → SIP Subscribers → Configuration → Configulation selection → Additional numbers

Dial plan #0-16 – additional subscriber number in the corresponding dial plan.

VAS management

In this section, VAS settings for subscribers are configured.

Each subscriber is provided with VAS services, but to use a specific service it is necessary to active it with an operator. The operator can create a service plan from several VAS functions, to do this, in the Configuration of SIP subscribers section, set the ‘Enable VAS’ flag and the flags opposite the necessary functions of the VAS.

The subscriber can manage the status of services from the phone. The following functions are available:

• Service activation – activation and entry of additional data;
• Service verification;
• Cancel service.

After entering the activation code or canceling the service, the subscriber can hear either a ‘Confirmation’ signal (3 short signals), or ‘Busy’ signal (periodic signal with a duration signal/pause – 0.35/0.35 s). The ‘Confirmation’ signal indicates that the service has been successfully activated or cancelled, the ‘Busy’ signal indicates that the subscriber is not connected to this service.

After entering the service verification code, the subscriber can hear either the ‘Station Answer’ signal (continuous signal) or ‘Busy’ signal. The ‘Station Answer’ signal indicates that the service is enabled and activated for the subscriber, the ‘Busy’” signal indicates that either the service is disabled oк the subscriber is not connected to this service.

The menu displays only those numbers for which the ‘Enable VAS’ flag is set in the configuration menu (Configuration of SIP subscribers section).

Subscribers → SIP subscribers → VAS management

Subscribers → SIP subscribers → VAS management → Object

• Number for call forward (unconditional) – phone number for unconditional forwarding;
• Number for call forward (busy) – phone number for call forwarding by busy;
• Number for call forward (no-reply) – phone number for call forwarding by no reply;
• Number for call forward (out of service) – phone number for call forwarding by out of service;
• Restrict local redirect (Unconditional);
• Restrict local redirect (Busy);
• Restrict local redirect (No-reply);
• Restrict local redirect (Out of service);
• Restrict local redirect (Time);
• password – пароль длиной от 4 до 8 цифр для доступа к услуге ограничения связи по паролю;assword activation – при установленном флаге пароль активирован и ограничения исходящей связи сняты;
• restrict out – задаёт запрет исходящей связи на определённые виды направлений при неактивном пароле:
  • all allowed;
  • Denied 6;
  • Denied 7;
  • Denied 8;
  • Denied 9.
• 'Anonymous call' service activation;
• 'Reject Anonymous calls' service activation;
• Follow me:
• Follow me activation – активация услуги;
• Follow me pin – активация управления услугой с пин-кодом;
• Follow me number – активация номера для переадресации вызовов;
• Follow me pin – пин-код;
• Follow me number – номер для переадресации.
• Follow me (no response):
• Follow me activation;
• Follow me pin;
• Follow me number;
• Follow me (no response) pin;
• Follow me (no response);

Call forward (Time):

• Schedule selection. 

Voice mail:

• Voice mail activation:
  • Not set;
  • Unconditional;
  • No-reply;
  • Busy;
  • Out of service; 
  • DND.
• Password.

Autoredial:

• Responce Waiting Timer;
• Number of call retries;
• Time between call retries.

Autoredial with callback:

• Caller answer qaiting timer;
• Call waiting timer;
• Number of call retries;
• Time between call retries.

‘White List’ tab – on this tab, one can activate ‘Do Not Disturb’ service and set white list of numbers that can call a subscriber, despite the ban.

‘Black List’ tab – on this tab, one can activate the ‘Black List’ service and set a black list of numbers that cannot call a subscriber.

A detailed description of the operation and configuration of VAS services is given in Appendix H. Working with VAS services.

SIP Subscribers monioring

Upon selecting the ‘Monitoring’ item in the drop-down list, a table of subscriber states is displayed

Subscribers → SIP Subscribers → Monitoring

• Search subscriber – checking the database of configured SIP subscribers, one can check by name, number, status, SIP domain, IP address:Port;
• State – subscriber registration status (registration is aсtive, not registered, registration expired);
• Title – arbitrary text description of a subscriber;
• Number – subscriber number;
• SIP domain – domain to which the subscriber belongs;
• IP/Port – IP address and port of the subscriber;
• Last registration – time of the last registration;
• Expire in – time remaining before the registration expiration.

Click the ‘Stop registration’ button to forcibly reset the registration for selected subscribers.

PBX profiles

RBX profiles are used to assign additional settings to SIP subscribers.

Subscribers → PBX profiles

To create, edit, or remove profile, use the Objects – Add Object, Objects – Edit Object or Objects – Remove Object menus and the following buttons:

– Add;

– Edit;

– Remove. 

PBX profiles:

Subscribers → PBX profiles → Oblect

Description – profile name;

Station prefix – prefix added to the beginning of a SIP CgPN;

Direct routing prefix – routing to the prefix without analysing the calling or called subscriber’s number. Designed to route all calls from a SIP subscriber to the trunk group configured in the direct prefix, regardless of the dialled number (without creating masks in the prefixes);

Scheduled routing profile – selection of the ‘scheduled routing’ service profile, which is configured in the ‘Internal Resources’ section.

Ingress calls:

Use voice messages – whe checked, pre-recorded voice messages stored on the device are played when a specific event occurs; see Appendix G. Voice messages and music on hold (MOH);

No Connected number transit – deny Connected number;

Copy CgPN into Redlirecting number – when checked, if there is no Redirecting number in the incoming call, it will be generated from the CgPN number;

Use Redirecting number for routing – when checked, the Redirecting number field is used when employing SS7 or Q.931 signalling protocols, or the SIP protocol’s diversion field, to route an incoming call according to the CgPN number masking scheme;

CdPN modifiers – intended for modifications based on the analysis of the called party’s number received from the subscriber’s equipment;

CgPN modifiers – intended for modifications based on the analysis of the calling party’s number received from the subscriber’s equipment.

Egress calls:

CdPN modifiers – intended for modifications based on the analysis of the called subscriber’s number, prior to sending the call to the subscriber’s equipment;

CgPN modifiers – intended for modifications based on the analysis of the calling subscriber’s number, prior to sending the call to the subscriber’s equipment.

RingBack settings:

Allows configuring the ringback of an audio file for a group of subscribers who belong to a specific PBX profile.

Modes:

Default – refer to the settings in the system parameters.х;

RingBack – play the default Ringback, ignoring the settings in the system parameters;

Audio file – replacing the standard Ringback sound with a sound of your choice, which was uploaded during the Ringback configuration stage under the ‘System Settings’ menu. 

Timeouts:

First digit timeout, sec – таймаут ожидания первой цифры, после нажатия абонентом клавиши FLASH при использовании услуги «Передача вызова». По истечении данного таймаута абоненту будет выдаваться сигнал «занято», диапазон 5–20 секунд;

Next digit timeout, sec – таймаут ожидания следующей за первой цифры набора номера при использовании услуги «Передача вызова». По истечении данного таймаута будет определен конец набора номера, и вызов будет смаршрутизирован, диапазон 5-20 секунд;

Busy-tone timeout, sec – таймаут выдачи сигнала «занято» в случае неуспешного набора номера абонента при использовании услуги «передачи вызова». По истечении данного таймаута произойдет переключение вызова на абонента, который находится на удержании;

Timeout for call answer, sec (for V5.2 abonents) – таймаут ответа абонента при входящем звонке, по его истечению вызывающему абоненту отправиться разъединение;

Timeout for call hold, sec (for V5.2 abonents) – таймаут нахождения абонента в состоянии удержания.

VAS timeouts:

CFNR timeouts, sec – таймаут, по истечении которого у абонента после поступление на него звонка сработает услуга ДВО «переадресация по неответу», диапазон 5–60 секунд;

Timeout for call park, sec – таймаут, по истечении которого у абонента после установки его в слот на парковку сработает обратный вызов на инициатора установки в слот.

Flash signal settings:

Flash mode (for V5.2 abonents):

Treat as on-hook – сигнал flash будет восприниматься как разъединение;

flash1,2,3 – выбор блока параметров сигнала flash. Сам блок параметров настраивается на AN.

PRI profiles

Subscribers → SIP Subscribers → Object

PRI profiles are used to configure PRI subscribers:

Description – PRI profile menu;

Work mode – determines the order in which channels are occupied:

Start first forward;

Start last backward.

Egress calls modifiers:

CdPN – intended for modifications based on the analysis of the called subscriber number transmitted to the outgoing channel;

CgPN – intended for modifications based on the analysis of the calling subscriber number transmitted to the outgoing channel;

Original CdPN – intended for modifications based on the analysis of the original called party number transmitted to the outgoing channel;

RedirPN – intended for modifications based on the analysis of the redirecting number transmitted to the outgoing channel.

Q.931 Streams – streams are selected that will be associated with PRI subscribers.

Ingress calls/egress calls modifiers for PRI subscribers work as follows.

For example, on the trunk group of the E1 stream, to which PRI subscribers are attached, for ingress calls the CgPN (Table1) and CdPN (Table0) modifiers are set on the PBX profile to which PRI subscribers are attached, the CgPN (Table3) and CdPN (Table2) modifiers are also set for ingress calls. In all tables the selection mask is set as (x.)

A call comes from E1 stream:

1. The rule for CgPN from the modifier Table1 is applied.

2. The CgPN number ia checked for a PRI subscriber.

3a. If the call is not from a PRI subscriber, the call is processed as from a regular trunk; the remaining modifiers tied to the trunk group on the incoming connection will be applied.

3b. If the call is from a PRI subscriber, the remaining modifiers tied to the trunk group and the PBX profile are applied. The order of modifiers is as follows:

– The CgPN rule from Table3 is applied

– The CdPN rule from Table1 is applied

– The CdPN rule from Table3 is applied

– The CgPN rule from Table0 is applied

– The CgPN rule from Table2 is applied

– The CdPN rule from Table0 is applied

– The CdPN rule from Table2 is applied

The egress calls modifiers on a PRI profile are triggered if the call is routed to a PRI subscriber associated with this profile.

Dynamic subscribers groups

Configuration of dynamic subscribers group

In this section, the dynamic subscriber groups can be configured.

Dynamic registration uses digest authentication on a RADIUS server (RFC 5090, RFC-no-challenge, draft-sterman) for subscribers.

Subscribers → Dynamic subscribers groups → Configuration

To create, edit, or remove an entry, use the Objects – Add Object, Objects – Edit Object or Objects – Remove Object menus and the following buttons:

– Add subscribers;

– Edit subscriber parameters;

– Remove subscriber. 

It is possible to create up to 64 groups of dynamic subscribers.

Subscribers → Dynamic subscribes groups → Configuration → Object

Dynamic Subscribers Group:

Subscribers number – the number of subscribers in the group;

Description – name of the dynamic subscriber group;

Calling party number type – type of the subscriber number;

Calling party category (RUS) – subscriber's Caller ID category;

Lines operation mode – setting limits on the number of simultaneous calls. Can take two values: Common and Separate. The Common mode takes into account the total number of simultaneous calls in which the subscriber can take part; in the Separate mode, incoming and outgoing calls are counted separately;

Lines number – the number of simultaneous calls in which the subscriber can take part. The field appears if the line mode is set to Common. The range of possible values is [1;255] or 0 – no limits;

Ingress lines number¹ – the number of simultaneous incoming calls to the subscriber. The field appears if the line mode is set to Separate. The range of possible values is [1;255] or 0 – no limits;

Egress lines number¹ – the number of simultaneous outgoing calls from the subscriber. The field appears if the line mode is set to Separate. The range of possible values is [1;255] or 0 – no limits;

SIP domain – identifies the domain to which the subscriber belongs. It is sent by the subscriber gateway as the “host” parameter in the SIP URI of the from and to fields (see section Timer operation examples);

SIP profile – select the SIP profile. The SIP profile defines the most of the subscriber settings. Selecting “Any” profile makes it possible to register a sip subscriber on any of the available sip profiles in the system (see section SIP/SIP-T/SIP-I, SIP profiles section);

РВХ profile – select the РВХ profile (see section PBX profiles);

Access category – select an access category;

Dial plan – define the dial plan for the subscriber;

Ignore source port after registration – after registration, messages from subscribers can arrive from any port;

Subscriber service mode – set a limit on the incoming and outgoing communication for the subscriber:

off: the port is out of service. The subscriber number is present in the dial plan, but the subscriber terminal cannot be registered. Therefore, incoming calls will be rejected with the out of order cause; outgoing calls cannot be initiated;

on: all types of communication are available;

off 1: ;incoming communication is enabled; outgoing communication is to special services only;

off 2: incoming communication is disabled; outgoing communication is to special services only;

denied 1: full prohibition for incoming and outgoing calls. Calls will be routed according to the dial plan, but be rejected;

denied 2: full prohibition for incoming and outgoing calls, except for special services;

denied 3: incoming calls are prohibited, outgoing calls are allowed;

denied 4: incoming calls are prohibited, outgoing calls are allowed only for local and department communication;

denied 5: incoming calls are allowed, outgoing calls are fully prohibited;

denied 6: incoming calls are allowed, outgoing calls are allowed only for special services;

denied 7: incoming calls are allowed, outgoing calls are allowed only for local and private communication;

denied 8: incoming calls are allowed, outgoing calls are allowed only for local and private and zone communication;

denied 9: incoming calls are allowed, outgoing calls are allowed only for local, private, zone and national communication;

ignore: the number is excluded from the dial plan. The number is completely excluded from the subscriber number list of the dial plan. If this number is called, the call will be rejected with the no route to destination cause, or it will be routed to the appropriate prefix in the dial plan.

Intercom call settings

• Intercom call type – the incoming intercom call type (a call with an automatic answer of subscriber B):

• • One-way– with an incoming intercom call, subscriber B will hear subscriber A, but subscriber A will not hear subscriber B (one-way notification);
  • Two-way– with an incoming intercom call, both subscribers will hear each other;
  • Ordinary call– an incoming intercom call is made as a normal call, without an automatic answer of subscriber B;
  • Ignore – an incoming intercom call will be rejected.

• Intercom call priority – the priority of an incoming intercom call over other calls:

• • Ordinary call — priority 1;
  • Intercom call can be defined with the priority of 1–5, by default: 3;
  • Notification — 7.

Examples:

• If subscriber A with priority 1 calls an already busy subscriber B (with one line and any priority), then subscriber A will hang up;

• If subscriber A with priority 2 calls already busy subscriber B (with one line and any priority), then 1 more extension line will be allocated for subscriber B and subscriber B will receive a call notification from subscriber A; В случае если на ТА поддержан и настроен приоритет интерком-вызова, вызов абонента А будет автоматически принят, а уже существующий поставлен на удержание;

• If subscriber A with priority 2 calls already busy subscriber B (with one line and any priority), but subscriber B is already busy with subscriber C with priority 3, then subscriber A will hang up;
• Subscriber A should be notified in any case, because subscriber A has more high priority 7.

Intercom SIP header – select a SIP header to be sent to the callee in the INVITE message during an intercom/paging call:

Answer-Mode: Auto;

Alert-Info: Auto Answer;

Alert-Info: info=alert-autoanswer;

Alert-Info: Ring Answer;

Alert-Info: info=RingAnswer;

Alert-Info: Intercom;

Alert-Info: info=intercom;

Call-Info: =\;answer-after=0;

Call-Info: \\;answer-after=0;

Call-Info: ;answer-after=0.

Pause before answer, sec – the pause duration before answering an intercom/paging call, which can be transmitted in the ‘answer-after’ header.

VAS settings:

CLIRO – a service for overriding the prohibition on caller number identification;

VAS management – selects how VAS services will be activated for dynamic subscribers:

Not used – do not enable VAS services for dynamic subscribers;

Individual – VAS services can be configured for each subscriber individually via the gateway configurator. If this option is selected, the VAS Activation table will become available (see section SIP Subsribers);

From RADIUS – for dynamic subscribers, VAS settings will be sent in the RADIUS server responses. For details, see Appendix D. Transmission of VAS settings from RADIUS server for dynamic subscribers.

RingBack settings:

Allows configuring audio file playback individually.

Modes:

Default – refer to the settings in the system parameters.х;

RingBack – play the default Ringback, ignoring the settings in the system parameters;

Audio file – replacing the standard Ringback sound with a sound of your choice, which was uploaded during the Ringback configuration stage under the ‘System Settings’ menu. 

Voice Notification:

Launching a notification – enable subscribers to run VNS tasks;

Confirmation code – set a personal code to confirm receipt of an alert;

Operator access category – select a category that allow/deny subscriber access to the relevant tasks and VNS reports.

Monitoring of dynamic subscribers group

Subscribers → Dynamic subcribers groups → Monitoring

Click the ‘Search’ button to search entries for the subscriber with the specified number.

State – subscriber registration status (registered, not registered, registration expired);

Group Description – arbitrary text description of the group;

Number – the subscriber number;

SIP domain – the domain to which the subscriber belongs;

IP/Port – IP address and port of the subscriber;

Last registration – the time of the last registration;

Expire in – the time remaining before the registration expiration;

Select – when this option is checked, this entry in the table will be processed when you click the Reset registration button;

Stop registration – forcibly reset the registration for a selected subscriber.

Click the ‘Stop registration’ button to reset the registration of all subscribers in the specified group. You can select a group from the drop-down list.

VAS management of dynamic subscriber groups

Subscribers → Dynamic subcribers groups → VAS management

Click the ‘Search’ button to search entries for the subscriber with the specified number.

Group name – arbitrary text description of the group;

Number – the subscriber number;

Parameters – subscriber VAS parameters;

Select – when this option is checked, this entry in the table will be processed when you click the ‘Reset VAS’

Click the ‘Reset VAS’ button to forcibly reset the VAS settings for selected subscribers.

V5.2 subscribers

Subscribers → V5.2 Subscribers → Configuration

Search subscriber – checking the presence of a subscriber in the database of configured V5.2 subscribers. Can be checked by name, number, caller ID number, PBX profile, dial plans, V5.2 interface;

Edit selected – pressing the button allows going to the group editing menu of selected subscribers (opposite to which the flag is set ‘Select’). To be able to edit, set the ‘Edit’ flag opposite the required parameter. A description of the parameters for configuration is given below;

Remove selected – pressing the button allows group deletion of selected subscribers.

Import – import a subscriber database from a .json file. More details in Appendix P. Example of SMG configuration;

Export – save the subscriber database to a .json file. More details in Appendix P. Example of SMG configuration.

To create, edit, or remove an entry or particular user, use the Objects – Add Object, Objects – Edit Object or Objects – Remove Object menus and the following buttons:

– Add subscriber;
– Edit subscriber parameters;
– Remove subscriber.

Attach selected items – add selected subscribers to the V5.2 interface.

Subscribers → V5.2 Subscribers → Configuration → Object

Subscriber parameters:

Subscribers count – unique subscriber identifier;

Starting description – arbitrary text description of the subscriber;

Starting number – subscriber number, for a group of subscribers each subsequent one will be assigned a number increased by one;

Hotline (incoming) – hotline number is set. If the number is specified, then the service is activated automatically;

Hotline delay (incoming), sec – allows to set a hotline activation delay. Valid range [0;10];

Starting CallerID number – caller ID number of the subscriber, for a group of subscribers each subsequent one will be assigned a number increased by one;

Use CallerID number for redirection – use the number specified in the ‘Starting CallerID number’ field when performing call forwarding service;

Calling party number type – subscriber number type;

Calling party category – CallerID category;

PBX profile – PBX profile selection (see section PBX profiles);

Access category – access category selection (see PBX profiles);

Dial plan – dial plan in which the subscriber will be located;

CallerID generation – format for CallerID generation;

Off;

Caller ID (RUS);

DTMF;

FSK BELL202;

FSK V.23.

Subscriber service mode – sets a limit on incoming and outgoing communications to the subscriber:

off: the port is out of service. The subscriber number is present in the dial plan, but the subscriber terminal cannot be registered. Therefore, incoming calls will be rejected with the out of order cause; outgoing calls cannot be initiated;

on: all types of communication are available;

off 1: ;incoming communication is enabled; outgoing communication is to special services only;

off 2: incoming communication is disabled; outgoing communication is to special services only;

denied 1: full prohibition for incoming and outgoing calls. Calls will be routed according to the dial plan, but be rejected;

denied 2: full prohibition for incoming and outgoing calls, except for special services;

denied 3: incoming calls are prohibited, outgoing calls are allowed;

denied 4: incoming calls are prohibited, outgoing calls are allowed only for local and department communication;

denied 5: incoming calls are allowed, outgoing calls are fully prohibited;

denied 6: incoming calls are allowed, outgoing calls are allowed only for special services;

denied 7: incoming calls are allowed, outgoing calls are allowed only for local and private communication;

denied 8: incoming calls are allowed, outgoing calls are allowed only for local and private and zone communication;

denied 9: incoming calls are allowed, outgoing calls are allowed only for local, private, zone and national communication;

ignore: the number is excluded from the dial plan. The number is completely excluded from the subscriber number list of the dial plan. If this number is called, the call will be rejected with the no route to destination cause, or it will be routed to the appropriate prefix in the dial plan.

VAS settings:

CLIRO – a service to overcome the ban on issuing a caller’s number;

Enable VAS¹ – connection of VAS services for subscriber. When checked, the ‘VAS activation’ table is available.

RingBack settings:

Allows configuring audio file playback individually.

Modes:

Default – refer to the settings in the system parameters.х;

RingBack – play the default Ringback, ignoring the settings in the system parameters;

Audio file – replacing the standard Ringback sound with a sound of your choice, which was uploaded during the Ringback configuration stage under the ‘System Settings’ menu. 

Voice Notification²: 

VAS activation

Subscribers → V5.2 Subscribers → Configuration → Object → Enable VAS

Call Forward (Unconditional) – enables the Call Forwarding Unconditional (CF Unconditional) service;

Call Forward (Busy) – enables the Call Forwarding Busy (CF Busy) service;

Call Forward (No Reply) – enables the Call Forwarding No Reply (CF No Reply) service;

Call Forward (Out of Service) – enables the Call Forwarding Out of Service (CF Out of Service);

Call Forward (Time) – enables the Call Forwarding by Time service (CT Time);

Call hold;

Call transfer – enables the Call Transfer service;

3WAY conference;

Call pickup;

Conference;

Disconnect conference by initiator – when checked, the conference will be over when the initiator leaves the conference. Otherwise, the conference will be saved after the initiator is hung up and will be over only when the last participant leaves the conference;

Change password – changing the password to restrict outgoing communications;

Outgroing calls restriction – use the service ‘restricting outgoing communications by password’;

Restricted by password – allows the subscriber to make a one-time call without restrictions communication by entering the VAS password;

Password activation – allows the subscriber to enter a password once to remove the outgoing communication restriction. Re-entering the password again sets the restrictions;

DND – allows the subscriber to set the ‘Do not disturb’ mode and set several numbers from the white list who will still be able to call the subscriber;

Blacklist – allows the subscriber to blacklist numbers so that they cannot call the subscriber;

Follow me – allows one to forward all calls one’s phone to a remote phone using the remote phone;

Follow me (no response) – allows one to forward all calls coming to ‘local’ number, to the ‘remote’ number in case the local number did not receive a call within the specified time interval;

Call Park To – allows the subscriber to use the call parking service;

Slot setting (within call parking service);

Extraction from slot (within call parking service);

Voice mail – activation of voice mail service;

Reset all services – function required to cancel all configured numbers for forwarding by pressing the service prefix configured in the numbering plan; 

VAS management

Subscribers → V5.2 Subscribers → VAS Management

In this section, VAS settings for subscribers are configured.

Each subscriber is provided with VAS services, but to use a specific service it is necessary to activate it with an operator. The operator can create a service plan from several VAS functions, to do this, set the ‘Enable VAS’ flag and the flags opposite the necessary functions of the VAS in the SIP susbcribers configuration tab (see SIP Subsribers section). 

The subscriber can manage the status of services from the phone. The following functions are available:

• Service activation – activation and entry of additional data;
• Service verification;
• Cancel service.

After entering the activation code or canceling the service, the subscriber can hear either a ‘Confirmation’signal (3 short signals), or ‘Busy’ signal (periodic signal with a duration signal/pause – 0.35/0.35 s). The ‘Confirmation’ signal indicates that the service has been successfully activated or cancelled, the ‘Busy’ signal indicates that the subscriber is not connected to this service.

After entering the service verification code, the subscriber can hear either the ‘Station Answer’ signal (continuous signal) or ‘Busy’ signal. The ‘Station Answer’ signal indicates that the service is enabled and activated for the subscriber, the ‘Busy’ signal indicates that either the service is disabled or the subscriber is not connected to this service.

The menu displays only those numbers for which the ‘Enable VAS’ flag is set in the configuration menu (see SIP Subsribers section).

Subscribers → V5.2 Subscribers → VAS Management →

Number for call forward (unconditional) – phone number for unconditional forwarding;

Number for call forward (busy) – phone number for call forwarding by busy;

Number for call forward (no-reply) – phone number for call forwarding by no reply;

Number for call forward (out of service) – phone number for call forwarding by out of service;

Password – a password of 4 to 8 digits in length to access the ‘outgoing calls restriction’ service;

Password activation – when the flag is set, the password is activated and the restrictions on outgoing calls have been removed;

Restrict out – sets a ban on outgoing calls for certain types of directions with inactive password:

All allowed – restriction on outgoing calls is not in effect, restriction code is 0;

Only to emergency – outgoing communication is limited to calls to emergency, restriction code is 1;

Only local and department network – outgoing communication is limited to calls within local and department networks, restriction code is 2;

Only local, department and zone network – outgoing communication is limited to calls within local, department and zone networks, restriction code is 3.

Only local, department, zone and national network – outgoing communication is limited to calls within local, department, zone and national networks, restriction code is 3.

‘White List’ tab – on this tab, one can activate ‘Do Not Disturb’ service and set white list of numbers that can call a subscriber, despite the ban.

‘Black List’ tab – on this tab, one can activate the ‘Black List’ service and set a black list of numbers that cannot call a subscriber.

A detailed description of the operation and configuration of VAS services is given in Appendix H. Working with VAS services. 

PRI subscribers

PRI subscribers are numbers that are located behind a PRI trunk (E1 streams with Q.931 signaling) and are perceived by SMG as local subscribers with the provision of some subscriber services.

Routing to such subscribers is carried out without creating additional rules in the dial plan.

Checking whether the calling subscriber is a PRI subscriber is carried out by matching the E1 stream Q.931, from which the call and A-numbers came.

Subscribers → PRI Subscribers → Configuration

Search subscriber – checking the presence of a subscriber in the database of configured PRI subscribers, possible checking by name, number, PRI profile, PBX profile, dial plans.

Редактировать выделенных – по нажатию на кнопку осуществляется переход в меню группового редактирования параметров выделенных абонентов (напротив которых установлен флаг «Выделить»). Для возможности редактирования необходимо установить флаг «Изменить» напротив требуемого параметра. Описание параметров для конфигурирования приведено ниже;

Удалить выделенных – по нажатию на кнопку осуществляется групповое удаление выделенных абонентов;

Импорт – позволяет импортировать базу данных абонентов из .json файла. Подробнее в Приложении Р. Расшифровка структуры файла импорта/экспорта абонентов;

Экспорт – позволяет сохранить базу данных абонентов в .json файл. Подробнее в Приложении Р. Расшифровка структуры файла импорта/экспорта абонентов.

To create, edit, or remove an entry of particular user, use the Objects – Add Object, Objects – Edit Object or Objects – Remove Object menus and the following buttons:

– Add;
– Edit;
– Remove.

PRI subscriber parameters:

Subscribers → PRI Subscribers → Configuration → Object

Subscribers count – number of subscribers;

Starting description – arbitrary text description of the subscriber;

Starting number – subscriber number, for a group of subscribers each subsequent one will be assigned a number increased by one;

PRI profile – PRI profile selection;

PBX profile – PBX profile selection (see PBX profiles section);

Calling party category (RUS) – CallerID category;

Lines number – number of simultaneous calls involving the subscriber. Field is displayed if the Common line operation mode is selected. Acceptable range values [1;255] or 0 – no limits;

Ingress lines number – the number of simultaneous incoming calls to the subscriber. The field appears if the line mode is set to Separate. The range of possible values is [1;255] or 0 – no limits;

Egress lines number – the number of simultaneous outgoing calls from the subscriber. The field appears if the line mode is set to Separate. The range of possible values is [1;255] or 0 – no limits;

Access category – select an access category;

Dial plan – define the dial plan for the subscriber;

Subscriber service mode – set a limit on the incoming and outgoing communication for the subscriber:

off: the port is out of service. The subscriber number is present in the dial plan, but the subscriber terminal cannot be registered. Therefore, incoming calls will be rejected with the out of order cause; outgoing calls cannot be initiated;

on: all types of communication are available;

off 1: ;incoming communication is enabled; outgoing communication is to special services only;

off 2: incoming communication is disabled; outgoing communication is to special services only;

denied 1: full prohibition for incoming and outgoing calls. Calls will be routed according to the dial plan, but be rejected;

denied 2: full prohibition for incoming and outgoing calls, except for special services;

denied 3: incoming calls are prohibited, outgoing calls are allowed;

denied 4: incoming calls are prohibited, outgoing calls are allowed only for local and department communication;

denied 5: incoming calls are allowed, outgoing calls are fully prohibited;

denied 6: incoming calls are allowed, outgoing calls are allowed only for special services;

denied 7: incoming calls are allowed, outgoing calls are allowed only for local and private communication;

denied 8: incoming calls are allowed, outgoing calls are allowed only for local and private and zone communication;

denied 9: incoming calls are allowed, outgoing calls are allowed only for local, private, zone and national communication;

ignore: the number is excluded from the dial plan. The number is completely excluded from the subscriber number list of the dial plan. If this number is called, the call will be rejected with the no route to destination cause, or it will be routed to the appropriate prefix in the dial plan.

VAS settings:

Enable VAS ¹ – connection of VAS services for subscriber. When checked, the ‘VAS activation’ table is available.

VAS activation:

Subscribers → PRI Subscribers → Configuration → Object → Enable VAS

Call Forward (Unconditional) – enables the Call Forwarding Unconditional (CF Unconditional) service;

Call Forward (Busy) – enables the Call Forwarding Busy (CF Busy) service;

Call Forward (No Reply) – enables the Call Forwarding No Reply (CF No Reply) service;

Call Forward (Out of Service) – enables the Call Forwarding Out of Service (CF Out of Service);

Call Forward (Time) – enables the Call Forwarding by time.

A detailed description of the operation and configuration of VAS services is given in Appendix H. Working with VAS services. 

RingBack settings:

Allows configuring audio file playback individually.

Modes:

Default – refer to the settings in the system parameters.х;

RingBack – play the default Ringback, ignoring the settings in the system parameters;

Audio file – replacing the standard Ringback sound with a sound of your choice, which was uploaded during the Ringback configuration stage under the ‘System Settings’ menu. 

Internal resources

CDR settings

This section describes parameters configuration to save call detail records. CDR is a call detail record, which allows the system to save the history of calls performed through SMG gateway. If the primary server is unavailable, CDR records are sent to the backup server (with appropriate configuration of the backup server) until communication with the primary server is restored. After the connection is restored, the CDR records sent to the backup server, will not be loaded to the primary server. Go to the ‘Internal Resources’ section and to the ‘CDR Records’ tab.

Internal resources → CDR settings

CDR settings:

Enable CDR – when this option is checked, the gateway will generate CDRs.

CDR files settings:

Create files – select the mode to create CDR files:

• periodically– CDR file is created after the specified period has elapsed since the device boot;
• once per day – CDR file is created once a day at the specified time;
• once per hour– CDR file is created once an hour at the specified time.

Saving period: Days, Hours, minutes – time period for CDR generation and saving in the device RAM;

Add header – when this option is checked, the following header will be written at the beginning of the CDR file: SMG1016. CDR. File started at “YYYYMMDDhhmmss”, where “YYYYMMDDhhmmss” is the records saving start time;