VoIP Шлюз Grandstream GXW4232

Telephone connections
GXW4216/4224/4232: 16/24/32 x RJ11 FXS ports and 1/1/2 50-pin connectors
Telco GXW4248: 2 50–pin Telco Connectors

Network interfaces
1 x 10M/100M/1000Mbps RJ45 port with automatic identification

LED indicators
LAN Link, LAN Activity, connection for each phone port

LCD display
128x32 backlit graphic LCD display with multi
-language support

Speech transmission functions in batch mode
Windows-based operator-class line echo cancellation, dynamic
delay oscillation buffer, modem detection and automatic switching
to G.711

Speech signal compression
G.711, G.723.1, G.726 (40/32/24/16), G.729 A/B, iLBC

Fax by IP
Fax relay of the third group of the standard T.38 with support for data transfer rates
up to 14.4 Kbit/s and automatic switching to G.711 for fax transmission,
a functional unit for transmitting and receiving data by fax V.17, V.21, V.27ter, V.29
for fax relays of standard T.38

Telephone functions
Displaying or blocking the caller's number, notification
of an incoming call, blind or accompanied call transfer,
forwarding, banning incoming calls, three-way conference call,
dialing the last incoming call, search call, LED indicator
of an incoming message (NEON LED) and intermittent signal, auto-call

Quality of service
DiffServ, TOS, 802.1P/Q VLAN tagging

Network protocols
TCP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS, DHCP, NTP, TFTP, TELNET, PPPoE, STUN, LLDP (expected)

DTMF method
Configurable DTMF transmission methods, including In-audio, RFC2833, and/or SIP INFO

Signal transmission
SIP (RFC 3261) over UDP/TCP/TLS

SIP server profiles and accounts for each system
4 separate SIP server profiles for each system and separate accounts
SIP for each phone port

Registration
TFTP, HTTP, HTTPS, TR069

Safety
SRTP*, TLS/SIPS, HTTPS (AES-128 encryption for SRTP*, TLS and HTTPS)

Management
Syslog, HTTPS, Web browser, Telnet, Voice Reminders, TR-069

Universal power supply
GXW4232/4224/4216: Output: DC12V, 5 A; Input: 100~240V
AC, 50~60Hz; GXW4248: Output: DC24V, 6.25A; input:
100~240 vac, 50~60 Hz

Environmental characteristics
Operation: 0°C ~45°C; Storage: -20°C ~60°C; Humidity: 10％ ~ 90% (non
-condensing)
Electrical protection Overvoltage and overvoltage protection (ITU-T K.21 recommendation, basic
level of verification)

Physical characteristics
Device Size: 440 mm (L) x 255 mm (W) x 44 mm (H) (1U) (GXW4248)
440 mm (L) x 185 mm (W) x 44 mm (H) (1U) (GXW4216/4224/4232)
Device weight: 3.21 kg; package weight: 4.31 kg (GXW4248)
2.63 kg; 3.68 kg (GXW4232)
2.57 kg; 3.62 kg (GXW4224)
2.39 kg; 3.48 kg (GXW4216)

Installation
Installation on a table and in a rack with front brackets
Near and far communication 2 REN, up to 6000 feet for 24 AWG wire

Caller's number
Bellcore Type 1&2, ETSI, BT, NTT, and DTMF based CID

Disabling methods
Busy signal, polarity change/instability, current in the circuit

Compliance with regulatory requirements
US Federal Communications Commission: Part 15 (CFR 47) Class B, Council of Europe:
EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1, RoHS, C-TICK:
AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, AS/NZS 60950, ITU-T K.21 (basic
verification level); UL 60950 (power adapter)

Telephone connections
GXW4216/4224/4232: 16/24/32 x RJ11 FXS ports and 1/1/2 50-pin connectors
Telco GXW4248: 2 50–pin Telco Connectors

Network interfaces
1 x 10M/100M/1000Mbps RJ45 port with automatic identification

LED indicators
LAN Link, LAN Activity, connection for each phone port

LCD display
128x32 backlit graphic LCD display with multi
-language support

Speech transmission functions in batch mode
Windows-based operator-class line echo cancellation, dynamic
delay oscillation buffer, modem detection and automatic switching
to G.711

Speech signal compression
G.711, G.723.1, G.726 (40/32/24/16), G.729 A/B, iLBC

Fax by IP
Fax relay of the third group of the standard T.38 with support for data transfer rates
up to 14.4 Kbit/s and automatic switching to G.711 for fax transmission,
a functional unit for transmitting and receiving data by fax V.17, V.21, V.27ter, V.29
for fax relays of standard T.38

Telephone functions
Displaying or blocking the caller's number, notification
of an incoming call, blind or accompanied call transfer,
forwarding, banning incoming calls, three-way conference call, dialing
the number of the last incoming call, search call, LED indicator
of the incoming message (NEON LED) and intermittent signal, auto-call

Quality of service
DiffServ, TOS, 802.1P/Q VLAN tagging

Network protocols
TCP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS, DHCP, NTP, TFTP, TELNET, PPPoE, STUN, LLDP (expected)

DTMF method
Configurable DTMF transmission methods, including In-audio, RFC2833, and/or SIP INFO

Signal transmission
SIP (RFC 3261) over UDP/TCP/TLS

SIP server profiles and accounts for each system
4 separate SIP server profiles for each system and separate SIP accounts for each phone port

Registration
TFTP, HTTP, HTTPS, TR069

Safety
SRTP*, TLS/SIPS, HTTPS (AES-128 encryption for SRTP*, TLS and HTTPS)

Management
Syslog, HTTPS, Web browser, Telnet, Voice Reminders, TR-069

Universal power supply
GXW4232/4224/4216: Output: DC12V, 5 A; Input: 100~240V
AC, 50~60Hz; GXW4248: Output: DC24V, 6.25A; Input:
100~240 vac, 50~60 Hz

Environmental characteristics
Operation: 0°C ~45°C; Storage: -20°C ~60°C; Humidity: 10％ ~ 90% (non-condensing)

Electrical protection
Overvoltage and overvoltage protection (ITU-T Recommendation K.21, basic level of verification)

Physical characteristics
Device Size:
440 mm (L) x 185 mm (W) x 44 mm (H)
Device weight: 3.21 kg; package weight:
2.57 kg; 3.62 kg (GXW4224)

Installation
Installation on a table and in a rack using front brackets

Near and far communication
2 REN, up to 6000 feet for 24 AWG wire

Caller's number
Bellcore Type 1&2, ETSI, BT, NTT, and CID based

Disabling methods
Busy signal, polarity change/instability, current in the circuit

Compliance with regulatory requirements
US Federal Communications Commission: Part 15 (CFR 47) Class B, Council of Europe:
EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1, RoHS, C-TICK:
AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, AS/NZS 60950, ITU-T K.21 (basic
verification level); UL 60950 (power adapter)

Grandstream UCM6304 — корпоративная IP‑АТС для унифицированных коммуникаций

Grandstream UCM6304 — это мощная IP‑АТС (VoIP‑PBX) корпоративного уровня, предназначенная для построения современной телефонии и унифицированных коммуникаций в бизнес‑среде. Система объединяет голосовую связь, видеовызовы, конференции, интеграцию с облачными и локальными приложениями, а также поддержку как цифровых, так и аналоговых линий связи — всё это без дополнительных лицензионных затрат на пользователей.

Основные характеристики

• Поддержка до 2000 внутренних пользователей (SIP‑абонентов) — масштабируемое решение для крупных сетей.
• Обработка до 300 одновременных голосовых вызовов для высокой нагрузки.
• 4 FXS‑порта для подключения аналоговых телефонов и устройств.
• 4 FXO‑порта для подключения исходящих/входящих PSTN‑линий.
• Три адаптивных Gigabit Ethernet порта с поддержкой PoE+ и режимов маршрутизатора/коммутатора.
• Порты USB 3.0 и слот SD‑карты для расширения хранилища данных, записи звонков или резервного копирования.
• Встроенный LCD‑экран и удобный веб‑интерфейс для локального управления.

Функции телефонии и коммуникаций

• Поддержка широкого набора голосовых кодеков (Opus, G.711, G.722, G.729 и др.) для оптимального качества речи.
• Поддержка Fax‑over‑IP (T.38) для передачи факсов по IP‑сетям.
• Гибкое маршрутизирование вызовов, интерактивное голосовое меню (IVR), голосовая почта и очереди вызовов.
• Автоматическое Zero‑Config провижнинг для IP‑устройств Grandstream.
• Интегрированная платформа конференций и встреч, совместимая с приложением Grandstream Wave.

Сеть, безопасность и управление

• Поддержка современных сетевых протоколов и NAT‑обхода для удалённого доступа.
• Шифрование и безопасность: **SRTP, TLS, HTTPS, SSH** для защиты вызовов и интерфейсов.
• Удалённое управление через облачную платформу **GDMS** и API для интеграции с CRM/PMS.
• Поддержка IPv6 и продвинутых услуг QoS для приоритизации голосового трафика.

Преимущества

• Отсутствие лицензий на пользователей — экономичное корпоративное решение.
• Гибкая интеграция аналоговых и цифровых линий связи на одной платформе.
• Широкие коммуникационные возможности: голос, видео, конференции и совместная работа.
• Масштабируемость и высокая производительность для растущих бизнес‑сред.
• Централизованное управление и простое развертывание IP‑инфраструктуры.

Область применения

• Корпоративные офисы и филиальные сети.
• Службы поддержки и call‑центры с высокой нагрузкой.
• Объекты с большими требованиями к видеосвязи и конференциям.
• Сети малого и среднего бизнеса, требующие гибкой и масштабируемой PBX‑платформы.

Analog FXS ports
8 RJ11 ports

FXO ports PSTN lines
8 RJ11 ports

Network interfaces
Three adaptive Gigabit network ports (dial-up, routable, or dual mode) with PoE+ support

NAT Router
Yes (supports router mode and switch mode)

Peripheral ports
2*USB 3.0, 1* SD Card Slot

The SIDs
Power supply 1/2, FXS, FXO, LAN, WAN,
Heartbeat

LCD screen
Matrix graphic LCD screen, 128x32 dots with buttons
DOWN (down) and OK

Reset button
Yes, long press to reset to factory settings, short press to reboot

VoP
LEC capabilities with NLP Packetized Voice Protocol, 128ms-tail-length Operator-grade echo cancellation, Dynamic jitter buffer,
modem detection and automatic switching to G.711, NetEQ, FEC 2.0, jitter tolerance (up to 50% loss of audio packets)

Voice and fax codecs
Opus, G.711 A-law/U-law, G.722, G722.1 G722.1C, G.723.1 5.3K/6.3K, G.726-32, G.729A/B, iLBC, GSM; T.38

QoS
Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS

API
API for integrating third-party platforms and applications

The operating system
Based on Asterisk version 16

DTMF methods
In-band audio, RFC4733 and SIP INFO

Automatic Configuration protocol
and Plug-and-Play

Mass configuration using an XML configuration file with AES encryption, automatic detection and configuration of subscriber data

IP devices
from Grandstream using ZeroConfig (DHCP Option 66 multicast SIP SUBSCRIBE mDNS), a list of events between the local and
remote backbone

Network protocols
TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH, HTTP/HTTPS, PPPoE, STUN, SRTP, TLS, LDAP, HDLC, HDLC-ETH, PPP, Frame Relay
(in development), IPv6, OpenVPN®

Hang-up methods
Busy/Overload/Howl Tone, Reverse polarity, Short-term shutdown, Circuit current shutdown

Data encryption
SRTP, TLS, HTTPS, SSH, 802.1X

The UCM6300 Audio series enables enterprises to create powerful and scalable solutions for unified
communications and collaboration. This series of IP PBX is a platform that combines all the main
types of communication for business, including voice communication, instant messaging (IM), voice conferencing, web meetings,
data transmission, analytics, mobility, access to premises, internal communication and much more. The UCM6300 Audio series
supports up to 1,500 users and includes a built-in instant messaging (IM) solution,
a voice/web conferencing platform and a free Wave application that provides communication and collaboration
of employees using desktop computers, mobile devices, IP phones and other subscriber devices. It has support for UCM RemoteConnect, a cloud service for remote users, offering a best-in
-class hybrid platform that combines the management of a local IP PBX with remote access and
the management capabilities of a cloud solution. Offering a high-quality solution for unified communications and
collaboration, containing a set of tools for mobility, security, instant messaging, voice
conferencing and collaboration, the UCM6300 Audio series provides a powerful professional platform for any
organization.

UCM6302A

Users: 500
Simultaneous calls (G. 711): 75
Maximum number of SRTP
calls (G. 711): 75
5 conference halls and up to 75 participants

Analog FXS ports: 2
FXO ports: 2