GrandStream IP-PBX for T1/E1 and 200 Concurrent Calls

button text=”ucm6510 setup guide” letter_case=”lowercase” color=”secondary” style=”outline” size=”small” radius=”99″ icon=”icon-plus” icon_pos=”left” link=”https://www.uslatstore.com/wp-content/uploads/2021/01/ucm6510-setup-guide.pdf” target=”_blank”]

Designed to bring leading-edge voice, video, data, and mobility features to medium and large businesses and small to medium enterprises. The UCM6510 IP PBX appliance creates a customizable, scalable and complete network solution. Powered by an advanced hardware and software functionalities, this converged voice, video, data, fax, security surveillance and mobility solution anchors any communications network.

Description

The UCM6510 creates an easily manageable on-premise anchor to your communications network. This enterprise-grade IP PBX comes equipped with a suite of advanced call handling and network data features, all with no licensing and no fees. Its scalability offers deployments that can support up to 2000 users, and it supports E1, T1, and J1. The UCM6510 series allows businesses to unify multiple communication technologies, such as voice, video, surveillance, data tools, and facilities access management into one common platform that can be managed and accessed remotely. With features such as customizable call-routing, multi-level IVRs, call queues, auto-attendant, call detail records, multi-site peering, SIP video support, voicemail/fax forwarding to email and more, the UCM6510 delivers complete unified communications.

Specs:

• Application processor Cortex A9 of 1 GHz of 4 nuclei, great memory capacity (1GB DDR3 RAM, 32GB NAND Flash) and processors DSP dedicated of multiple nuclei of high performance for advanced voice processing

• An integrated T1/E1/J1 interface, 2 FXO ports for PSN trunks, 2 FXS ports for analog telephones with a capacity of backup in case of interruption of electrical energy and 50 trunks SIP
• Ports of red Gigabit with integrated PoE, USB, SD card; integrated NAT router with advanced support of QoS
• Cancellation of local exchange carrier (LEC) of 128ms processed by DSP, providing levels of quality of telephony operator, ID caller, detection of progress tone, and detection of smart impedance to adapt itself to the local telephony provider
• Supports up to 2000 SIP terminal registrations, up to 200 concurrent calls (up to 100 encrypted simultaneous calls with SRTP) and up to 64 participants in conference
• Data communication through T1/E1/J1 and combined communication of data and voice through T1/E1/J1 with SS7 or PRI
• Plan of flexible dialing, call routing, connection with other IP-PBXs or providers of service, call recording, panel of central control for terminals, integrated NTP server, and directory of integrated LDAP contacts
• Automatic detection and telephone IP provisioning, video telephones, ATAs, gateways, SIP cameras, and other terminals for easy implementation.
• Hardware accelerator for encrypted with SRTP, TLS, and HTTPS to guarantee the max. protection of security

• Provides the application Fail2ban for the prevention of malicious attacks, is compatible with Blacklist and Whitelist for an efficient defense

• Manual and automatic recorder for each SIP call and each trunk

• Exports data of the day before automatically; cleans periodically the user’s data

• Supports QoS both in layer 2 in layer 3

• 19 supported languages, between them Spanish

• Graphic LCD screen of 128 x 32 with a button of ‘Selection’ and ‘OK’

• Codecs of supported video: H264, H263, and H263+

• Codecs of supported audio: G.711 (I and II), G.723.1A, G.729A/B, G.726, G.722 e IIbc

• Compatible with protocol T.38 for Fax