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Overview
Designed to provide a centralised solution for the communication needs of businesses, the UCM6200 series IP PBX appliance combines enterprise-grade voice, video, data, and mobility features in an easy-to-manage solution. This IP PBX series allows businesses to unify multiple communication technologies, such voice, video calling, video conferencing, video surveillance, data tools,and facility access management onto one common network that that can be managed and/or accessed remotely. The secure and reliable UCM6200 series delivers enterprise-grade features without any licensing fees, costs-per-feature or recurring fees.
Features
- The UCM6208 is a new and improved replacement version of the popular UCM6108
- Zero configuration provisioning of Grandstream SIP endpoints
- Strongest-possible security protection using SRTP, TLS and HTTPS encryption
- Dual Gigabit network ports with integrated PoE
- Supports up to a 5-level IVR (Interactive Voice Response)
- Built-in call recording server; recordings accessed via web user interface
- Supports call queue for efficient call volume management
- Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc
- Integrated LDAP and XML phonebooks, flexible dial plan
- Supports any SIP video endpoint that uses the H.264, H.263 or H.263+ codecs
- Supports voicemail and fax forwarding to email
| Details | |
|---|---|
| Manufacturer's Product Code | UCM6208 |
| Analog Telephone FXS Ports | 2 ports (both with lifeline capability in case of power outage) |
| PSTN Line FXO Ports | 8 ports |
| Network Interfaces | Dual Gigabit RJ45 ports with integrated PoE Plus (IEEE 802.3at-2009) |
| NAT Router | Yes (supports router mode and switch mode) |
| Peripheral Ports | USB, SD |
| Voice-over-Packet Capabilities | LEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier grade Line Echo Cancellation, Dynamic Jitter Buffer, Modem detection & auto-switch to G.711 |
| Voice and Fax Codecs | G.711 A-law/U-law, G.722, G.723.1 5.3K/6.3K, G.726, G.729A/B, iLBC, GSM, AAL2-G.726-32, ADPCM; T.38 |
| Video Codecs | H.264, H.263, H263+ |
| QoS | Layer 3 QoS, Layer 2 QoS |
| DTMF Methods | In Audio, RFC2833, and SIP INFO |
| Provisioning Protocol & Plug-and-Play | TFTP/HTTP/HTTPS, auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP Option 66 multicast SIP SUBSCRIBE mDNS), eventlist between local and remote trunk |
| Network Protocols | TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH, HTTP/HTTPS, PPPoE, SIP (RFC3261), STUN, SRTP, TLS, LADP |
| Disconnect Methods | Call Progress Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect, Busy Tone |
| Dimensions | 226mm L x 155mm W x 34.5mm H |
| Caller ID | Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT |
| Polarity Reversal/ Wink | Yes, with enable/disable option upon call establishment and termination |
| Call Center | Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/busy level, in-queue announcement |
| Customizable Auto Attendant | Up to 5 layers of IVR (Interactive Voice Response) |
| Concurrent Calls | Up to 800 registered SIP devices/users, Up to 100 concurrent calls or 66% performance if calls are SRTP encrypted |
| Conference Bridges | Up to 6 password-protected conference bridges allowing up to 32 simultaneous PSTN or IP participants |
| Call Features | Call park, call forward, call transfer, DND, ring/hunt group, paging/intercom etc. |
| Universal Power Supply | Output: 12VDC, 1.5A; Input: 100 ~ 240VAC, 50 ~ 60Hz |
