Grandstream UCM6100 VOIP Gateway Router GSM PBX System
Grandstream UCM6100 VOIP Gateway Router GSM PBX System
The UCM6100 series IP PBX appliance is designed to bring enterprise-grade voice, video, data, and mobility features to small-to-medium businesses (SMBs) in an easy-to-manage fashion. By incorporating industry-leading features, the UCM6100 series offers quick and easy setup and deployment using the web-browser user interface – which features auto-discovery of Grandstream endpoints and Zero-Configuration provisioning. The UCM6100 series allows businesses to unify multiple communication technologies, such comprehensive voice, video calling, video conferencing, video surveillance, data tools, and facility access management onto one common network that that can be managed and/or accesed remotely. The secure and reliable UCM6100 series delivers enterprise-grade features to SMBs at an unprecedented price point without any licensing fees, costs-per-feature, or recurring fees.
- Integrated 2/4/8/16 PSTN trunk FXO ports, 2 analog telephone FXS ports, and up to 50 SIP trunk options
- Gigabit network port with integrated PoE, USB, SD; integrated NAT router with advanced QoS support (UCM6102 only)
- Supports up to 60 concurrent calls and up to 32 conference attendees
Grandstream SIP Devices can be configured via Web interface as well as via configuration file through TFTP/HTTP/HTTPS download. All Grandstream SIP devices support a proprietary binary format configuration file and XML format configuration file. The UCM6100 provides a Plug and Play mechanism to auto-provision the Grandstream SIP devices in a zero configuration manner by generating XML config file and having the phone to download it within LAN area. This allows users to finish the installation with ease and start using the SIP devices in a managed way.
To provision a phone, three steps are involved, i.e., discovery, assignment and provisioning. The UCM6100 creates XML config file to the detected/assigned Grandstream device and accomplishes the following configurations on the device after the provisioning:
1 A UCM6100 extension will be assigned and registered on the phone.
2 SIP-related network settings such as "NAT traversal" and "Use Random Port" are configured on the phone.
3 Call feature settings such as "Public Mode", "Voicemail User ID", "Dial Plan" and "Auto Answer".
4 LDAP client configurations will be set up automatically on the phone to use the default LDAP directory generated in the UCM6100 LDAP server.
5 Date format, time format and time zone settings for the phone to be provisioned.
UCM6102
UCM6104
UCM6108
UCM6116
1.voip ip pbx
2.Integrated 2/4/8/16 PSTN trunk FXO ports
3.up to 50 SIP trunk options
4.Supports up to 60 concurrent calls
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Technical Specifications |
UCM6102 |
UCM6104 |
UCM6108 |
UCM6116 |
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INTERFACES |
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PSTN Line FXO Ports |
2 |
4 |
8 |
16 |
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Analog Telephone FXS Ports |
2 |
2 |
2 |
2 |
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Network Interfaces |
Single or Dual (UCM6102 only) 10M/100M/1000M RJ45 Ethernet port(s) with integrated PoE Plus (IEEE 802.3at-2009) |
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Data Router |
Yes |
No |
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Peripheral Ports |
USB, SD |
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LED Indicators |
Power/Ready, Network, PSTN Line, USB, SD |
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LCD Display |
128x32 graphic LCD with DOWN & OK button |
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Reset Switch |
YES |
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VOICE/VIDEO CAPABILITIES |
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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 |
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Voice Compression & Fax |
G.711 A-law/U-law, G.722, G.723.1 5.3K/6.3K, G.726, G.729A/B, iLBC, GSM; T.38 |
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Video Compression |
H.264, H.263, H263+ |
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QoS |
Layer 3 QoS. |
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SIGNALING & CONTROL |
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DTMF Method |
In Audio, RFC2833 and SIP INFO |
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Provisioning Protocol & Plug-and-Play |
TFTP/HTTP/HTTPS, auto-discovery & auto-provisioning of Grandstream IP endpoints |
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Network Protocols |
TCP/UDP/IP, RTP/RTCP, ICMP, ARP/RARP, DNS, DDNS, DHCP, NTP, TFTP, SSH, HTTP/HTTPS, PPPoE, SIP (RFC3261), STUN, SRTP, TLS/SIP |
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Disconnect Methods |
Call Progress Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect, Busy Tone |
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SECURITY |
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Media |
SRTP, TLS, HTTPS, SSH, Syslog |
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PHYSICAL |
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Universal Power Supply |
Output: 12VDC, 1.5A; Input: 100 ~ 240VAC, 50 ~ 60Hz |
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Environmental |
Operating: 32 ~ 104ºF / 0 ~ 40ºC, 10 ~ 90% (non-condensing); Storage: 14 ~ 140ºF / -10 ~ 60ºC |
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Dimensions (L*W*H) |
226mm(L)x 155mm(W)x 34.5mm(H) |
440mm(L)x 185mm(W)x 44mm(H) |
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Mounting |
Wall mount & Desktop (UCM6102/6104); Rack mount & Desktop (UCM6108/6116) |
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Concurrent Calls |
Up to 30 (UCM6102), 45 (UCM6104), or 60 (UCM6108/6116) simultaneous calls |
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