IP PBX Phone System with 16 to 48 FXS/FXO Analog Extension or Trunks, Support 168 IP Users and 216 IP Trunks
The RT80 is a highly integrated hybrid IP-PBX with up to 200 analog and/or IP extensions. Due to its high-performance and XML/HTTP based API. It is a cost-effective solution for mid-size contact centers with up to 60 agents as well. Powered by innovative VoIP technology from Real Tone, the RT80 provides seamless connectivity to not only PSTN and analog terminals, but also IP-based SIP trunks and SIP voice stations.
Always Connected with Your Customers
With features such as remote extension, call forking to your mobile phone, and call-back, the RT80 assures you that wherever you are--whether in your office, traveling abroad, or working at home--you will never miss any important business calls.
Never Miss a Call
The auto-attendant provides 24×7 service to customers, and efficiently directs incoming calls to the extensions or cell phones of the correct person through configurable voice menus for business/non-business hours and holidays. The smart auto-attendant can even deliver the calls from VIP or repeat customers to the appropriate agents directly, bypassing voice menus (caller-agent binding).
Always in Touch
The RT80 has all the features needed to fully support the remote worker, business travel, and communications with branches, all through the Internet. International or long-distance calls are turned into local toll-free calls. With advanced-routing capabilities, the RT80 can route calls intelligently according to call type.
Remote Management and NAT Traversal
The RT80 includes an embedded Real Tone Cloud client that allows the RT80 located behind an enterprise NAT or firewall to be accessed from the Internet for maintenance and troubleshooting or by voice devices from remote branch sites, remote IP extensions, or far-end value-adding applications. Real-time monitoring, alarm notification, remote packet captures and software upgrades can be performed with the Real Tone or third-party Element Management Systems (EMS) with TR-069 or SNMP.
Features
- Dual-redundant Ethernet ports
- Dual AC/DC power supplies (optional)
- Multi-site deployment with up to 300 sites
- Call recordings on the internal storage or external server
- Add-on call recording management system
- Add-on middle-ware for hospitality applications
- XML/HTTP-based RESTful API interworking with and supporting third-party applications
- Multi-level auto-attendant
- Support TR-069, SNMP
- Interoperability with popular SIP servers, such as Cisco Unified CallManager (CUCM), Broadsoft, Microsoft Skype for Business (Lync), Huawei IMS, and Asterisk/Elastix
- Class I lightning protection