Grandstream UCM6204
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DESCRIPTION
- Supports up to 500 users, 50 SIP trunk accounts, up to 45 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 | 4 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 Customizable IVR/voice prompts for en,cn,de,es,gr,fr,it,nl,pl,pt,ru,sv,tr,he,ar Customizable language pack to support any other languages |
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 Concurrent SIP calls: Up to 45 or 66% performance if calls are SRTP encrypted |
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. |
Compliance | FCC: Part 15 (CFR 47) Class B, Part 68 CE: EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1, TBR21, RoHS A-TICK: AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, AS/NZS 60950, AS/ACIF S002 ITU-T K.21 (Basic Level); UL 60950 (power adapter) |