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Grandstream UCM6108 8 port fxo fxs card asterisk elastix voip ip pbx

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Grandstream UCM6108 8 port fxo fxs card asterisk elastix voip ip pbx

Grandstream UCM6108 8 port fxo fxs card asterisk elastix voip ip pbx

Update: 2015/1/14View: 26
  • Brand: Grandstream
  • Type: UCM6108
  • Price: RMB
  • Promotion: RMB ( 0 Customers Bought)


1.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

Grandstream UCM6108 8 port fxo fxs card asterisk elastix voip ip pbx

Product Description

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.



Analog Telephone FXS Ports

2 ports (both with lifeline capability in case of power outage)

PSTN Line FXO Ports

2 ports (UCM6102); 4 ports (UCM6104); 8 ports (UCM6108); 16 ports (UCM6116)

Network Interfaces

Single or Dual (UCM6102/6104) 10/100/1000Mbps RJ45 Ethernet port(s) with integrated PoE Plus (IEEE 802.3at-2009)

NAT Router

Yes (UCM6102 only)

Peripheral Ports


LED Indicators

Power/Ready, Network, PSTN Line, USB, SD

LCD Display

128x32 graphic LCD with DOWN & OK button

Reset Switch


Voice/Video Capabilities

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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