Hasil pencarian untuk 'gsm gateway'

The Yeastar EX08 expansion board has 4 onboard module slots for the future expansion and 8 port RJ-11 for interfaces on the panel. This board is well supported by the Yeastar S-Series models (S100 and S300) and Yeastar P-Series models (P560 and P570) IP PBX systems.

• EX08 Expansion Board support Yeastar S-Series models (S100 and S300) and Yeastar P-Series models (P560 and P570) 
• Supports up to 4 optional modules(8 RJ-11 Interface port), supporting Yeastar Module: O2, S2, BRI, GSM, CDMA, and UMTS/3G modules
• Support up to 2 expansion cards to S100 and P560 and 3 expansion card to the S300 and P570

**^(optional) If required ?, You can add or customize up to 4 optional module cards: (8 analog port (FXO/FXS/BRI)) and/or 4 channel of GSM/CDMA/3G  in your EX08 Expansion Boards according to your requirements below.

Grandstream UCM6304 Key Features

• Built in 4 FXO ports (PSTN Line) and 4 FXS ports (Analog Device); All ports have lifeline capability in case of power outage.
• Up to 15 Simultaneous Video Conference Rooms, up to 200 simultaneous participants in all room combined, up to 9 video feeds in all conference room
• Voice Conference : Up to 200 parties (G.711)
• Supports up to 2000 users and up to 300 concurrent calls
• Up to 200 SIP trunk accounts
• Zero configuration provisioning of Grandstream SIP endpoints
• Built-in conferencing & meetings platform; supports desktop, Wave app, and SIP endpoints
• Wave for Android, iOS, Chrome and Firefox browsers allows communication with all UCM6300 users & solutions
• API available for third-party integrations, including CRM and PMS platforms
• Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
• Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
• Automated NAT firewall traversal service facilitates secure remote connections
• Supports Full-Band Opus voice codec and H.264/H.263/ H.263+/VP8 video codec, jitter resilience up to 50% packet loss
• Compatible with GDMS for cloud setup, management and monitoring
• Based on Asterisk* version 16 open source telephony operating system

Grandstream UCM6308A Key Features

• Supports up to 1500 users and up to 200 concurrent calls
• Zero configuration provisioning of Grandstream SIP endpoints
• Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints
• Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices
• API available for third-party integrations, including CRM and PMS platforms
• Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
• Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
• Automated NAT firewall traversal service facilitates secure remote connections
• Enhanced reliability with support for Hot Standby High-Availability and local dual deployment
• Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss
• Compatible with GDMS for cloud setup, management, and monitoring
• Based on Asterisk* version 16 open source telephony operating system

Grandstream UCM6304A Key Features

• Supports up to 1000 users and up to 150 concurrent calls
• Zero configuration provisioning of Grandstream SIP endpoints
• Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints
• Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices
• API available for third-party integrations, including CRM and PMS platforms
• Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
• Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
• Automated NAT firewall traversal service facilitates secure remote connections
• Enhanced reliability with support for Hot Standby High-Availability and local dual deployment
• Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss
• Compatible with GDMS for cloud setup, management, and monitoring
• Based on Asterisk* version 16 open source telephony operating system

Grandstream UCM6302A Key Features

• Supports up to 500 users and up to 75 concurrent calls
• Zero configuration provisioning of Grandstream SIP endpoints
• Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints
• Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices
• API available for third-party integrations, including CRM and PMS platforms
• Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
• Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
• Automated NAT firewall traversal service facilitates secure remote connections
• Enhanced reliability with support for Hot Standby High-Availability and local dual deployment
• Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss
• Compatible with GDMS for cloud setup, management, and monitoring
• Based on Asterisk* version 16 open source telephony operating system

Grandstream UCM6300A Audio Series IP PBX

The Grandstream UCM6300A allows businesses to build powerful and scalable unified communication and collaboration solutions. This series of IP PBXs provide a platform that unifies fundamental business communications needs, including voice, instant messaging (IM), voice meetings, audio web meetings, data, analytics, mobility, facility access,intercoms and more.

Grandstream UCM6300A Key Features

• Supports up to 250 users and up to 50 concurrent calls
• Zero configuration provisioning of Grandstream SIP endpoints
• Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints
• Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices
• API available for third-party integrations, including CRM and PMS platforms
• Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
• Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
• Automated NAT firewall traversal service facilitates secure remote connections
• Enhanced reliability with support for Hot Standby High-Availability and local dual deployment
• Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss
• Compatible with GDMS for cloud setup, management, and monitoring
• Based on Asterisk* version 16 open source telephony operating system

Grandstream UCM6308 Key Features

• Built-in 8 FXO ports (PSTN Line) and 8 FXS ports (Analog Device); All ports have lifeline capability in case of power outage.
• Up to 25 Simultaneous Video Conference Rooms, up to 300 simultaneous participants in all room combined, up to 9 video feeds in all conference room
• Voice Conference : Up to 300 parties (G.711)
• Supports up to 3000 users and up to 450 concurrent calls
• Up to 200 SIP trunk accounts
• Zero configuration provisioning of Grandstream SIP endpoints
• Built-in conferencing & meetings platform; supports desktop, Wave app, and SIP endpoints
• Wave for Android, iOS, Chrome and Firefox browsers allows communication with all UCM6300 users & solutions
• API available for third-party integrations, including CRM and PMS platforms
• Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
• Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
• Automated NAT firewall traversal service facilitates secure remote connections
• Supports Full-Band Opus voice codec and H.264/H.263/ H.263+/VP8 video codec, jitter resilience up to 50% packet loss
• Compatible with GDMS for cloud setup, management and monitoring
• Based on Asterisk* version 16 open source telephony operating system