Grandstream UCM6202
DESCRIPTION
- Supports up to 500 users, 50 SIP trunk accounts, up to 30 concurrent calls
- 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.
- Multi-language auto-attendant to efficiently handle incoming calls
- 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
Datasheet:
Analog Telephone FXS Ports |
2 ports (both with lifeline capability
in case of power outage) |
PSTN Line FXO Ports |
2 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 |
LED Indicators |
Power/Ready, Network, PSTN Line, USB,
SD |
LCD Display |
128x32 graphic LCD with DOWN & OK
button |
Reset Switch |
Yes |
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 Zero-Config (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 |
Media Encryption |
SRTP, TLS, HTTPS, SSH |
Universal Power Supply |
Output: 12VDC, 1.5A; Input: 100 ~
240VAC, 50 ~ 60Hz |
Dimensions |
226mm L x 155mm W x 34.5mm H |
Weight |
Unit weight 0.51kg, Package weight
0.94kg |
Environmental |
Operating: 32 ~ 104ºF / 0 ~ 40ºC, 10 ~
90% (non-condensing); Storage: 14 ~ 140ºF / -10 ~ 60ºC |
Mounting |
Wall mount & Desktop |
Multi-Language Support |
Web UI: en,cn,es,fr,pt,de,ru,it,pl,cs |
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) |
Maximum Call Capacity |
Registered SIP devices: supports up to
500 registered SIP devices/users |
Conference Bridges |
Up to 3 password-protected conference
bridges allowing up to 25 simultaneous PSTN or IP participants |
Call Features |
Call park, call forward, call transfer,
DND, ring/hunt group, paging/intercom etc. |
|
FCC: Part 15 (CFR 47) Class B, Part 68 |
Designed to provide a centralized 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 as voice, video calling, video conferencing, video surveillance, data tools, mobility options and facility access management onto one common network 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.