Description
- supports up to 2000 users, 50 SIP trunk accounts, up to 200 concurrent calls
- Zero configuration provisioning of Grandstream SIP endpoints
- Strongest-possible security protection using SRTP, TLS and HTTPS encryption
- Gigabit network ports with integrated PoE+; Integrated NAT router
- 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 using the H.264, H.263 and H.263+ codecs
- supports voicemail and fax forwarding to email
Details | |
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Manufacturer's Product Code | UCM6510 |
Analog Telephone FXS Ports | 2 RJ11 ports (both with lifeline capability in case of power outage) |
PSTN Line FXO Ports | 2 RJ11 ports (both with lifeline capability in case of power outage) |
T1/E1/J1 Interface | 1 RJ45 port |
Network Interfaces | Dual Gigabit ports (switched or routed) with PoE+ |
NAT Router | Yes (user configurable) |
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, LDAP, HDLC, HDLC-ETH, PPP, Frame Relay (pending) |
Disconnect Methods | Call Progress Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect, Busy Tone |
Dimensions | 440mm(L) x 185mm(W) x 44mm(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/work-load, in-queue announcement |
Customizable Auto Attendant | Up to 5 layers of IVR (Interactive Voice Response) |
Concurrent Calls | Up to 2000 registered SIP endpoints, up to 200 concurrent calls |
Conference Bridges | Up to 8 bridges, up to 64 simultaneous conference attendees |
Call Features | Call park, call forward, call transfer, DND, DISA, ring group, pickup group, blacklist, paging/intercom etc. |
Universal Power Supply | Input: 100 ~ 240VAC, 50/60Hz; Output: DC+12V, 1.5A |
Details | |
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Power Supply | Included |
Mounting | 1U Rackmount |
Specifications
Details | |
---|---|
Manufacturer's Product Code | UCM6510 |
Analog Telephone FXS Ports | 2 RJ11 ports (both with lifeline capability in case of power outage) |
PSTN Line FXO Ports | 2 RJ11 ports (both with lifeline capability in case of power outage) |
T1/E1/J1 Interface | 1 RJ45 port |
Network Interfaces | Dual Gigabit ports (switched or routed) with PoE+ |
NAT Router | Yes (user configurable) |
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, LDAP, HDLC, HDLC-ETH, PPP, Frame Relay (pending) |
Disconnect Methods | Call Progress Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect, Busy Tone |
Dimensions | 440mm(L) x 185mm(W) x 44mm(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/work-load, in-queue announcement |
Customizable Auto Attendant | Up to 5 layers of IVR (Interactive Voice Response) |
Concurrent Calls | Up to 2000 registered SIP endpoints, up to 200 concurrent calls |
Conference Bridges | Up to 8 bridges, up to 64 simultaneous conference attendees |
Call Features | Call park, call forward, call transfer, DND, DISA, ring group, pickup group, blacklist, paging/intercom etc. |
Universal Power Supply | Input: 100 ~ 240VAC, 50/60Hz; Output: DC+12V, 1.5A |
Expected Delivery Times
North Island - 2-6 Working Days
South Island - 3-10 Working Days