D--OPTION - VoIP System and Network Infrastructure Installation

AMENDMENT NOTICE: This is a combined synopsis/solicitation for commercial items prepared in accordance with the format in FAR Subpart 12.6, as supplemented with additional information included in this notice. The solicitation number is N6671520RC72025 and is issued as a Request for Quote (RFQ), unless otherwise indicated herein. The solicitation document and incorporated provisions and clauses are those in effect through Federal Acquisition Circular 2020-04. The associated North American Industrial Classification System (NAICS) code for this procurement is 517919 with a small business size standard of $32.50M. This requirement is a Small Business Set-Aside and only qualified offerors may submit quotes. The solicitation pricing on www.UnisonMarketplace.com will start on the date this solicitation is posted, and, unless otherwise displayed at www.UnisonMarketplace.com, will end on: 2020-02-27 17:00:00.0 Eastern Time. This time supersedes the Offers Due Time listed above. FOB Destination shall be in the Statement of Work. The FLC - San Diego requires the following items, Meet or Exceed, to the following: Base Period of Performance: 03/10/2020 - 03/09/2021 LI 001: Provider Hosted and/or On-Premises Internet Protocol (IP) - Private Branch Exchange (PBX) Telecommunications System *Additional details in SOW, 1, EA; LI 002: Monthly maintenance plan to diagnose and repair IP telephones, wiring and jacks for all IP telephones used with the system owned or rented., 67, EA; LI 003: Monthly flat rate for the use of 67 individual telephone numbers used with an IP Telecommunications System hosted or owned., 67, EA; LI 004: A 6Mb SIP Trunking Port to handle 67 individuals lines that uses the protocol to provision voice over IP (VoIP) connectivity between an on-premises phone system and the public switched telephone network (PSTN). A service offered by a communications service provider., 1, EA; LI 005: Call Detail Reporting - A system generated program featuring a history of incoming, outgoing and internal extension to extension calls. Multiple report types present call trends, such as call volumes by day, week, time of day and more., 1, EA; LI 006: IP Phone or VoIP System compatible phones: Four voice lines, Four Independent SIP Registrations, Line status: active line indication, with name and number, Menu-driven user interface, Shared line appearance**, Speakerphone, Call hold, Call waiting, Caller ID name and number, Outbound caller, ID blocking, Call transfer: attended and blind, Three-way call conferencing with local mixing, Multiparty conferencing via external conference bridge, Call swap, Call back on busy, Call blocking: anonymous and selective, Call forwarding: unconditional, no answer, on busy, Hot line and warm line automatic calling, Redial from call logs, Do not disturb, Anonymous caller blocking, Uniform Resource Identifier (URI) (IP) dialing support (vanity numbers), On-hook default audio configuration (speakerphone and headset), Configurable dial/numbering plan support, Intercom, Group paging, Network Address Translation (NAT) Traversal, including Simple Traversal of UDP Through NATs (STUN) support, DNS SRV and multiple A records for proxy lookup and proxy redundancy, Syslog, debug, report generation, and event logging, Highly secure call encrypted voice communications support, Built-in web server for administration and configuration with multiple security levels, Automated remote provisioning, multiple methods; up to 256-bit encryption (HTTP, HTTPS, Trivial File Transfer Protocol [TFTP]), Option to require administrator password to reset unit to factory defaults., 65, EA; LI 007: IP Conference Phones - Power - IEEE 802.3af Power over Ethernet - Display Size (pixels): 248 x 68 (W x H) - Standard 12-key keypad - Context-dependent soft keys: 4 - On-hook/Off-hook, redial, mute, volume up/down - 5-way navigation - Menu Audio features - Loudspeaker - Frequency: 250 - 7,000 HZ - Volume: Adjustable to peak volume 84 dB at 1/2 meter distance - Voice activity detection - Comfort noise fill - DTMF tone generation / DTMF event RTP payload - Low-delay audio packet transmission - Adaptive jitter buffers - Packet loss concealment - Acoustic echo cancellation - Background noise suppression - Supported Codecs - G.711 (A-law and -law) - G.729a (Annex A, B) - G.722 - iLBC Call handling features - Shared call / bridged line appearance - Busy Lamp Field (BLF) - Distinctive incoming call treatment / call waiting - Call timer - Call transfer, hold, divert (forward), pickup - Called, calling, connected party information - Advanced Local three-way conferencing (conference, join, split, hold, resume) - One-touch speed dial, redial - Call waiting - Remote missed call notification - Automatic off-hook call placement - Do not disturb function Other features - Local feature-rich GUI - Time and date display - Corporate Directory Access (search, dial, save to local directory) - Convenient volume adjustment keys - User-configurable contact directory and call history (missed, placed, and received) - Customizable call progress tones - Wav file support for call progress tones - Unicode UTF-8 character support. Multilingual user interface encompassing Network and provisioning - Ethernet 10/100 Base-T - IP Address Configuration: DHCP and Static IP - Time synchronization with SNTP server - FTP / TFTP / HTTP / HTTPS server based central provisioning for mass deployments. Provisioning server redundancy supported. - Web portal for individual unit configuration - QoS Support - IEEE 802.1p/Q tagging (VLAN), Layer 3 TOS and DSCP - Network Address Translation (NAT) support - static - RTCP support (RFC 1889) - Event logging - Local dial plan - Hardware diagnostics - Status and statistics - User selectable ringer tones - Field upgradeable Security - Transport Layer Security (TLS) - Encrypted configuration files - Digest authentication - Password login - Support for URL syntax with password for boot server - HTTPS secure provisioning - Support for signed software executables Safety - CE Mark - EN60950-1 - IEC60950-1 - UL60950-1 - CAN/CSA C22.2 No.60950-1-03 - AS/NZS60950-1 - RoHS Compliant EMC - FCC Part 15 (CFR 47) Class B - ICES-003 Class B - EN55022 Class B - CISPR22 Class B - AS/NZS CISPR22 Class B - VCCI Class B - EN22024 Protocol support - IETF SIP (RFC 3261 and companion RFCs), 2, EA; LI 008: Unlimited Phone Services, 67, EA; LI 009: Multi-Level Auto Attendant - Advanced, 1, EA; LI 010: A service plan available by a communications provider that will connect the automated system attendant protocol to a private branch exchange (PBX) server, or a voice over internet protocol (VoIP) system. The plan will cover program and system updates, telephone line maintenance, telephone replacments and service agreements with minimal cost and time., 1, EA; LI 011: Additional Lines for overflow call volume and future expansion., 4, EA; LI 012: Long Distance International Calling Ability in case of emergencies requiring ability to contact personnel as many members are from foreign countries and visit family in other countries., 1, EA; LI 013: 2 Hour minimum Customer Education Training - Training will consist of the overall PBX/VoIP/hosted IP console in the operation of internal programs and telephone setup and operation., 1, EA; LI 014: Labor - Activation *Additional details in SOW, 67, EA; LI 015: Labor - Wiring *Additional details in SOW, 67, EA; LI 016: Labor - Install *Additional details in SOW, 1, EA; Option 1 Period of Performance: 03/10/2021 - 03/09/2022 LI 001: Provider Hosted and/or On-Premises Internet Protocol (IP) - Private Branch Exchange (PBX) Telecommunications System *Additional details in SOW, 1, EA; LI 002: Monthly maintenance plan to diagnose and repair IP telephones, wiring and jacks for all IP telephones used with the system owned or rented., 67, EA; LI 003: Monthly flat rate for the use of 67 individual telephone numbers used with an IP Telecommunications System hosted or owned., 67, EA; LI 004: A 6Mb SIP Trunking Port to handle 67 individuals lines that uses the protocol to provision voice over IP (VoIP) connectivity between an on-premises phone system and the public switched telephone network (PSTN). A service offered by a communications service provider., 1, EA; LI 005: Call Detail Reporting - A system generated program featuring a history of incoming, outgoing and internal extension to extension calls. Multiple report types present call trends, such as call volumes by day, week, time of day and more., 1, EA; LI 006: IP Phone or VoIP System compatible phones: Four voice lines, Four Independent SIP Registrations, Line status: active line indication, with name and number, Menu-driven user interface, Shared line appearance**, Speakerphone, Call hold, Call waiting, Caller ID name and number, Outbound caller, ID blocking, Call transfer: attended and blind, Three-way call conferencing with local mixing, Multiparty conferencing via external conference bridge, Call swap, Call back on busy, Call blocking: anonymous and selective, Call forwarding: unconditional, no answer, on busy, Hot line and warm line automatic calling, Redial from call logs, Do not disturb, Anonymous caller blocking, Uniform Resource Identifier (URI) (IP) dialing support (vanity numbers), On-hook default audio configuration (speakerphone and headset), Configurable dial/numbering plan support, Intercom, Group paging, Network Address Translation (NAT) Traversal, including Simple Traversal of UDP Through NATs (STUN) support, DNS SRV and multiple A records for proxy lookup and proxy redundancy, Syslog, debug, report generation, and event logging, Highly secure call encrypted voice communications support, Built-in web server for administration and configuration with multiple security levels, Automated remote provisioning, multiple methods; up to 256-bit encryption (HTTP, HTTPS, Trivial File Transfer Protocol [TFTP]), Option to require administrator password to reset unit to factory defaults., 65, EA; LI 007: IP Conference Phones - Power - IEEE 802.3af Power over Ethernet - Display Size (pixels): 248 x 68 (W x H) - Standard 12-key keypad - Context-dependent soft keys: 4 - On-hook/Off-hook, redial, mute, volume up/down - 5-way navigation - Menu Audio features - Loudspeaker - Frequency: 250 - 7,000 HZ - Volume: Adjustable to peak volume 84 dB at 1/2 meter distance - Voice activity detection - Comfort noise fill - DTMF tone generation / DTMF event RTP payload - Low-delay audio packet transmission - Adaptive jitter buffers - Packet loss concealment - Acoustic echo cancellation - Background noise suppression - Supported Codecs - G.711 (A-law and -law) - G.729a (Annex A, B) - G.722 - iLBC Call handling features - Shared call / bridged line appearance - Busy Lamp Field (BLF) - Distinctive incoming call treatment / call waiting - Call timer - Call transfer, hold, divert (forward), pickup - Called, calling, connected party information - Advanced Local three-way conferencing (conference, join, split, hold, resume) - One-touch speed dial, redial - Call waiting - Remote missed call notification - Automatic off-hook call placement - Do not disturb function Other features - Local feature-rich GUI - Time and date display - Corporate Directory Access (search, dial, save to local directory) - Convenient volume adjustment keys - User-configurable contact directory and call history (missed, placed, and received) - Customizable call progress tones - Wav file support for call progress tones - Unicode UTF-8 character support. Multilingual user interface encompassing Network and provisioning - Ethernet 10/100 Base-T - IP Address Configuration: DHCP and Static IP - Time synchronization with SNTP server - FTP / TFTP / HTTP / HTTPS server based central provisioning for mass deployments. Provisioning server redundancy supported. - Web portal for individual unit configuration - QoS Support - IEEE 802.1p/Q tagging (VLAN), Layer 3 TOS and DSCP - Network Address Translation (NAT) support - static - RTCP support (RFC 1889) - Event logging - Local dial plan - Hardware diagnostics - Status and statistics - User selectable ringer tones - Field upgradeable Security - Transport Layer Security (TLS) - Encrypted configuration files - Digest authentication - Password login - Support for URL syntax with password for boot server - HTTPS secure provisioning - Support for signed software executables Safety - CE Mark - EN60950-1 - IEC60950-1 - UL60950-1 - CAN/CSA C22.2 No.60950-1-03 - AS/NZS60950-1 - RoHS Compliant EMC - FCC Part 15 (CFR 47) Class B - ICES-003 Class B - EN55022 Class B - CISPR22 Class B - AS/NZS CISPR22 Class B - VCCI Class B - EN22024 Protocol support - IETF SIP (RFC 3261 and companion RFCs), 2, EA; LI 008: Unlimited Phone Services, 67, EA; LI 009: Multi-Level Auto Attendant - Advanced, 1, EA; LI 010: A service plan available by a communications provider that will connect the automated system attendant protocol to a private branch exchange (PBX) server, or a voice over internet protocol (VoIP) system. The plan will cover program and system updates, telephone line maintenance, telephone replacments and service agreements with minimal cost and time., 1, EA; LI 011: Additional Lines for overflow call volume and future expansion., 4, EA; LI 012: Long Distance International Calling Ability in case of emergencies requiring ability to contact personnel as many members are from foreign countries and visit family in other countries., 1, EA; LI 013: 2 Hour minimum Customer Education Training - Training will consist of the overall PBX/VoIP/hosted IP console in the operation of internal programs and telephone setup and operation., 1, EA; LI 014: Labor - Activation *Additional details in SOW, 67, EA; LI 015: Labor - Wiring *Additional details in SOW, 67, EA; LI 016: Labor - Install *Additional details in SOW, 1, EA; Option 2 Period of Performance: 03/10/2022 - 03/09/2023 LI 001: Provider Hosted and/or On-Premises Internet Protocol (IP) - Private Branch Exchange (PBX) Telecommunications System *Additional details in SOW, 1, EA; LI 002: Monthly maintenance plan to diagnose and repair IP telephones, wiring and jacks for all IP telephones used with the system owned or rented., 67, EA; LI 003: Monthly flat rate for the use of 67 individual telephone numbers used with an IP Telecommunications System hosted or owned., 67, EA; LI 004: A 6Mb SIP Trunking Port to handle 67 individuals lines that uses the protocol to provision voice over IP (VoIP) connectivity between an on-premises phone system and the public switched telephone network (PSTN). A service offered by a communications service provider., 1, EA; LI 005: Call Detail Reporting - A system generated program featuring a history of incoming, outgoing and internal extension to extension calls. Multiple report types present call trends, such as call volumes by day, week, time of day and more., 1, EA; LI 006: IP Phone or VoIP System compatible phones: Four voice lines, Four Independent SIP Registrations, Line status: active line indication, with name and number, Menu-driven user interface, Shared line appearance**, Speakerphone, Call hold, Call waiting, Caller ID name and number, Outbound caller, ID blocking, Call transfer: attended and blind, Three-way call conferencing with local mixing, Multiparty conferencing via external conference bridge, Call swap, Call back on busy, Call blocking: anonymous and selective, Call forwarding: unconditional, no answer, on busy, Hot line and warm line automatic calling, Redial from call logs, Do not disturb, Anonymous caller blocking, Uniform Resource Identifier (URI) (IP) dialing support (vanity numbers), On-hook default audio configuration (speakerphone and headset), Configurable dial/numbering plan support, Intercom, Group paging, Network Address Translation (NAT) Traversal, including Simple Traversal of UDP Through NATs (STUN) support, DNS SRV and multiple A records for proxy lookup and proxy redundancy, Syslog, debug, report generation, and event logging, Highly secure call encrypted voice communications support, Built-in web server for administration and configuration with multiple security levels, Automated remote provisioning, multiple methods; up to 256-bit encryption (HTTP, HTTPS, Trivial File Transfer Protocol [TFTP]), Option to require administrator password to reset unit to factory defaults., 65, EA; LI 007: IP Conference Phones - Power - IEEE 802.3af Power over Ethernet - Display Size (pixels): 248 x 68 (W x H) - Standard 12-key keypad - Context-dependent soft keys: 4 - On-hook/Off-hook, redial, mute, volume up/down - 5-way navigation - Menu Audio features - Loudspeaker - Frequency: 250 - 7,000 HZ - Volume: Adjustable to peak volume 84 dB at 1/2 meter distance - Voice activity detection - Comfort noise fill - DTMF tone generation / DTMF event RTP payload - Low-delay audio packet transmission - Adaptive jitter buffers - Packet loss concealment - Acoustic echo cancellation - Background noise suppression - Supported Codecs - G.711 (A-law and -law) - G.729a (Annex A, B) - G.722 - iLBC Call handling features - Shared call / bridged line appearance - Busy Lamp Field (BLF) - Distinctive incoming call treatment / call waiting - Call timer - Call transfer, hold, divert (forward), pickup - Called, calling, connected party information - Advanced Local three-way conferencing (conference, join, split, hold, resume) - One-touch speed dial, redial - Call waiting - Remote missed call notification - Automatic off-hook call placement - Do not disturb function Other features - Local feature-rich GUI - Time and date display - Corporate Directory Access (search, dial, save to local directory) - Convenient volume adjustment keys - User-configurable contact directory and call history (missed, placed, and received) - Customizable call progress tones - Wav file support for call progress tones - Unicode UTF-8 character support. Multilingual user interface encompassing Network and provisioning - Ethernet 10/100 Base-T - IP Address Configuration: DHCP and Static IP - Time synchronization with SNTP server - FTP / TFTP / HTTP / HTTPS server based central provisioning for mass deployments. Provisioning server redundancy supported. - Web portal for individual unit configuration - QoS Support - IEEE 802.1p/Q tagging (VLAN), Layer 3 TOS and DSCP - Network Address Translation (NAT) support - static - RTCP support (RFC 1889) - Event logging - Local dial plan - Hardware diagnostics - Status and statistics - User selectable ringer tones - Field upgradeable Security - Transport Layer Security (TLS) - Encrypted configuration files - Digest authentication - Password login - Support for URL syntax with password for boot server - HTTPS secure provisioning - Support for signed software executables Safety - CE Mark - EN60950-1 - IEC60950-1 - UL60950-1 - CAN/CSA C22.2 No.60950-1-03 - AS/NZS60950-1 - RoHS Compliant EMC - FCC Part 15 (CFR 47) Class B - ICES-003 Class B - EN55022 Class B - CISPR22 Class B - AS/NZS CISPR22 Class B - VCCI Class B - EN22024 Protocol support - IETF SIP (RFC 3261 and companion RFCs), 2, EA; LI 008: Unlimited Phone Services, 67, EA; LI 009: Multi-Level Auto Attendant - Advanced, 1, EA; LI 010: A service plan available by a communications provider that will connect the automated system attendant protocol to a private branch exchange (PBX) server, or a voice over internet protocol (VoIP) system. The plan will cover program and system updates, telephone line maintenance, telephone replacments and service agreements with minimal cost and time., 1, EA; LI 011: Additional Lines for overflow call volume and future expansion., 4, EA; LI 012: Long Distance International Calling Ability in case of emergencies requiring ability to contact personnel as many members are from foreign countries and visit family in other countries., 1, EA; LI 013: 2 Hour minimum Customer Education Training - Training will consist of the overall PBX/VoIP/hosted IP console in the operation of internal programs and telephone setup and operation., 1, EA; LI 014: Labor - Activation *Additional details in SOW, 67, EA; LI 015: Labor - Wiring *Additional details in SOW, 67, EA; LI 016: Labor - Install *Additional details in SOW, 1, EA; Option 3 Period of Performance: 03/10/2023 - 03/09/2024 LI 001: Provider Hosted and/or On-Premises Internet Protocol (IP) - Private Branch Exchange (PBX) Telecommunications System *Additional details in SOW, 1, EA; LI 002: Monthly maintenance plan to diagnose and repair IP telephones, wiring and jacks for all IP telephones used with the system owned or rented., 67, EA; LI 003: Monthly flat rate for the use of 67 individual telephone numbers used with an IP Telecommunications System hosted or owned., 67, EA; LI 004: A 6Mb SIP Trunking Port to handle 67 individuals lines that uses the protocol to provision voice over IP (VoIP) connectivity between an on-premises phone system and the public switched telephone network (PSTN). A service offered by a communications service provider., 1, EA; LI 005: Call Detail Reporting - A system generated program featuring a history of incoming, outgoing and internal extension to extension calls. Multiple report types present call trends, such as call volumes by day, week, time of day and more., 1, EA; LI 006: IP Phone or VoIP System compatible phones: Four voice lines, Four Independent SIP Registrations, Line status: active line indication, with name and number, Menu-driven user interface, Shared line appearance**, Speakerphone, Call hold, Call waiting, Caller ID name and number, Outbound caller, ID blocking, Call transfer: attended and blind, Three-way call conferencing with local mixing, Multiparty conferencing via external conference bridge, Call swap, Call back on busy, Call blocking: anonymous and selective, Call forwarding: unconditional, no answer, on busy, Hot line and warm line automatic calling, Redial from call logs, Do not disturb, Anonymous caller blocking, Uniform Resource Identifier (URI) (IP) dialing support (vanity numbers), On-hook default audio configuration (speakerphone and headset), Configurable dial/numbering plan support, Intercom, Group paging, Network Address Translation (NAT) Traversal, including Simple Traversal of UDP Through NATs (STUN) support, DNS SRV and multiple A records for proxy lookup and proxy redundancy, Syslog, debug, report generation, and event logging, Highly secure call encrypted voice communications support, Built-in web server for administration and configuration with multiple security levels, Automated remote provisioning, multiple methods; up to 256-bit encryption (HTTP, HTTPS, Trivial File Transfer Protocol [TFTP]), Option to require administrator password to reset unit to factory defaults., 65, EA; LI 007: IP Conference Phones - Power - IEEE 802.3af Power over Ethernet - Display Size (pixels): 248 x 68 (W x H) - Standard 12-key keypad - Context-dependent soft keys: 4 - On-hook/Off-hook, redial, mute, volume up/down - 5-way navigation - Menu Audio features - Loudspeaker - Frequency: 250 - 7,000 HZ - Volume: Adjustable to peak volume 84 dB at 1/2 meter distance - Voice activity detection - Comfort noise fill - DTMF tone generation / DTMF event RTP payload - Low-delay audio packet transmission - Adaptive jitter buffers - Packet loss concealment - Acoustic echo cancellation - Background noise suppression - Supported Codecs - G.711 (A-law and -law) - G.729a (Annex A, B) - G.722 - iLBC Call handling features - Shared call / bridged line appearance - Busy Lamp Field (BLF) - Distinctive incoming call treatment / call waiting - Call timer - Call transfer, hold, divert (forward), pickup - Called, calling, connected party information - Advanced Local three-way conferencing (conference, join, split, hold, resume) - One-touch speed dial, redial - Call waiting - Remote missed call notification - Automatic off-hook call placement - Do not disturb function Other features - Local feature-rich GUI - Time and date display - Corporate Directory Access (search, dial, save to local directory) - Convenient volume adjustment keys - User-configurable contact directory and call history (missed, placed, and received) - Customizable call progress tones - Wav file support for call progress tones - Unicode UTF-8 character support. Multilingual user interface encompassing Network and provisioning - Ethernet 10/100 Base-T - IP Address Configuration: DHCP and Static IP - Time synchronization with SNTP server - FTP / TFTP / HTTP / HTTPS server based central provisioning for mass deployments. Provisioning server redundancy supported. - Web portal for individual unit configuration - QoS Support - IEEE 802.1p/Q tagging (VLAN), Layer 3 TOS and DSCP - Network Address Translation (NAT) support - static - RTCP support (RFC 1889) - Event logging - Local dial plan - Hardware diagnostics - Status and statistics - User selectable ringer tones - Field upgradeable Security - Transport Layer Security (TLS) - Encrypted configuration files - Digest authentication - Password login - Support for URL syntax with password for boot server - HTTPS secure provisioning - Support for signed software executables Safety - CE Mark - EN60950-1 - IEC60950-1 - UL60950-1 - CAN/CSA C22.2 No.60950-1-03 - AS/NZS60950-1 - RoHS Compliant EMC - FCC Part 15 (CFR 47) Class B - ICES-003 Class B - EN55022 Class B - CISPR22 Class B - AS/NZS CISPR22 Class B - VCCI Class B - EN22024 Protocol support - IETF SIP (RFC 3261 and companion RFCs), 2, EA; LI 008: Unlimited Phone Services, 67, EA; LI 009: Multi-Level Auto Attendant - Advanced, 1, EA; LI 010: A service plan available by a communications provider that will connect the automated system attendant protocol to a private branch exchange (PBX) server, or a voice over internet protocol (VoIP) system. The plan will cover program and system updates, telephone line maintenance, telephone replacments and service agreements with minimal cost and time., 1, EA; LI 011: Additional Lines for overflow call volume and future expansion., 4, EA; LI 012: Long Distance International Calling Ability in case of emergencies requiring ability to contact personnel as many members are from foreign countries and visit family in other countries., 1, EA; LI 013: 2 Hour minimum Customer Education Training - Training will consist of the overall PBX/VoIP/hosted IP console in the operation of internal programs and telephone setup and operation., 1, EA; LI 014: Labor - Activation *Additional details in SOW, 67, EA; LI 015: Labor - Wiring *Additional details in SOW, 67, EA; LI 016: Labor - Install *Additional details in SOW, 1, EA; Option 4 Period of Performance: 03/10/2024 - 03/09/2025 LI 001: Provider Hosted and/or On-Premises Internet Protocol (IP) - Private Branch Exchange (PBX) Telecommunications System *Additional details in SOW, 1, EA; LI 002: Monthly maintenance plan to diagnose and repair IP telephones, wiring and jacks for all IP telephones used with the system owned or rented., 67, EA; LI 003: Monthly flat rate for the use of 67 individual telephone numbers used with an IP Telecommunications System hosted or owned., 67, EA; LI 004: A 6Mb SIP Trunking Port to handle 67 individuals lines that uses the protocol to provision voice over IP (VoIP) connectivity between an on-premises phone system and the public switched telephone network (PSTN). A service offered by a communications service provider., 1, EA; LI 005: Call Detail Reporting - A system generated program featuring a history of incoming, outgoing and internal extension to extension calls. Multiple report types present call trends, such as call volumes by day, week, time of day and more., 1, EA; LI 006: IP Phone or VoIP System compatible phones: Four voice lines, Four Independent SIP Registrations, Line status: active line indication, with name and number, Menu-driven user interface, Shared line appearance**, Speakerphone, Call hold, Call waiting, Caller ID name and number, Outbound caller, ID blocking, Call transfer: attended and blind, Three-way call conferencing with local mixing, Multiparty conferencing via external conference bridge, Call swap, Call back on busy, Call blocking: anonymous and selective, Call forwarding: unconditional, no answer, on busy, Hot line and warm line automatic calling, Redial from call logs, Do not disturb, Anonymous caller blocking, Uniform Resource Identifier (URI) (IP) dialing support (vanity numbers), On-hook default audio configuration (speakerphone and headset), Configurable dial/numbering plan support, Intercom, Group paging, Network Address Translation (NAT) Traversal, including Simple Traversal of UDP Through NATs (STUN) support, DNS SRV and multiple A records for proxy lookup and proxy redundancy, Syslog, debug, report generation, and event logging, Highly secure call encrypted voice communications support, Built-in web server for administration and configuration with multiple security levels, Automated remote provisioning, multiple methods; up to 256-bit encryption (HTTP, HTTPS, Trivial File Transfer Protocol [TFTP]), Option to require administrator password to reset unit to factory defaults., 65, EA; LI 007: IP Conference Phones - Power - IEEE 802.3af Power over Ethernet - Display Size (pixels): 248 x 68 (W x H) - Standard 12-key keypad - Context-dependent soft keys: 4 - On-hook/Off-hook, redial, mute, volume up/down - 5-way navigation - Menu Audio features - Loudspeaker - Frequency: 250 - 7,000 HZ - Volume: Adjustable to peak volume 84 dB at 1/2 meter distance - Voice activity detection - Comfort noise fill - DTMF tone generation / DTMF event RTP payload - Low-delay audio packet transmission - Adaptive jitter buffers - Packet loss concealment - Acoustic echo cancellation - Background noise suppression - Supported Codecs - G.711 (A-law and -law) - G.729a (Annex A, B) - G.722 - iLBC Call handling features - Shared call / bridged line appearance - Busy Lamp Field (BLF) - Distinctive incoming call treatment / call waiting - Call timer - Call transfer, hold, divert (forward), pickup - Called, calling, connected party information - Advanced Local three-way conferencing (conference, join, split, hold, resume) - One-touch speed dial, redial - Call waiting - Remote missed call notification - Automatic off-hook call placement - Do not disturb function Other features - Local feature-rich GUI - Time and date display - Corporate Directory Access (search, dial, save to local directory) - Convenient volume adjustment keys - User-configurable contact directory and call history (missed, placed, and received) - Customizable call progress tones - Wav file support for call progress tones - Unicode UTF-8 character support. Multilingual user interface encompassing Network and provisioning - Ethernet 10/100 Base-T - IP Address Configuration: DHCP and Static IP - Time synchronization with SNTP server - FTP / TFTP / HTTP / HTTPS server based central provisioning for mass deployments. Provisioning server redundancy supported. - Web portal for individual unit configuration - QoS Support - IEEE 802.1p/Q tagging (VLAN), Layer 3 TOS and DSCP - Network Address Translation (NAT) support - static - RTCP support (RFC 1889) - Event logging - Local dial plan - Hardware diagnostics - Status and statistics - User selectable ringer tones - Field upgradeable Security - Transport Layer Security (TLS) - Encrypted configuration files - Digest authentication - Password login - Support for URL syntax with password for boot server - HTTPS secure provisioning - Support for signed software executables Safety - CE Mark - EN60950-1 - IEC60950-1 - UL60950-1 - CAN/CSA C22.2 No.60950-1-03 - AS/NZS60950-1 - RoHS Compliant EMC - FCC Part 15 (CFR 47) Class B - ICES-003 Class B - EN55022 Class B - CISPR22 Class B - AS/NZS CISPR22 Class B - VCCI Class B - EN22024 Protocol support - IETF SIP (RFC 3261 and companion RFCs), 2, EA; LI 008: Unlimited Phone Services, 67, EA; LI 009: Multi-Level Auto Attendant - Advanced, 1, EA; LI 010: A service plan available by...