Hasil pencarian untuk 'gsm gateway'

Synway UC200 Series IP PBX available in 4 Variants System Capacity :

- UC200-15 : up to 60 User Extension and 15 Concurrent Call
- UC200-30 : up to 120 User Extension and 30 Concurrent Call
- UC200-45 : up to 220 User Extension and 45 Concurrent Call
- UC200-60 : up to 300 User Extension and 60Concurrent Call
* All variants are equipped with 4 FXO ports and 2 FXS ports.

Product Features :

• Maximum Registered Users : 60
• Maximun Concurrent Call : 15
• Built-in Call Recording Server; recordings accessed via web user interface
• Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
• Supports call queque for efficient call volume management
• Supports voicemail and fax forwarding to email
• Integrated LDAP and XML Phonebooks, flexible dial plans
• Zero Configuration Povisioning of Mainstream SIP Endpoints
• Highest level of security protection using SRTP, TLS, and HTTPS encryption
• HI-Speed network ports with Integrated NAT router and buit-in firewall, supporting 3 modes : Dual, Bridge, and Route
• Multi-Language auto-attendant to efficiently handle incoming calls
• Built-in with multi-level IVR Systems, supporting teleconferencing, call queuing, CDR, sound monitoring, call broadcasting, etc.
• Has 8GB internal storage with extendable external storage via TF and USB cards

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