Sangoma Vega 5000

The Vega 5000 Analog VoIP Gateway connects up to 50 analog handsets & devices to an IP network. The entire Sangoma Resiliency Enablement Suite (SRES) is available and the rich set of telephony features create an outstanding resilient solution that maintains key communications functions even in the event of a loss of connectivity to the IP-PBX or Hosted Solution. This is all supplemented by two FXO connections for direct connection to the PSTN or to a PBX. The GUI and Web-based SNAP Tool make the Vega 5000 one of the easiest gateways to provision and deploy.


SIP Trunking - Carrier

Carriers offering SIP trunking services must provide a secure environment that their customers can trust, especially when these services are delivered over the internet. Carriers also aim to reduce or eliminate interoperability difficulties between their equipment and that of their clients.

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SIP Trunking - Carrier

Branch Office Survivability

One of the main cost savings associated with the installation of an IP-PBX and SIP Trunks is that certain functions can be consolidated at a centralized IP-PBX installation. Services such as Voice Message Storage, Directory Services, International Call Routing and Endpoint Administration can all be handled at a central location, reducing the need for additional equipment and personnel at the remote or branch office. However, unless properly implemented, this can leave the remote or branch office vulnerable in the event that the IP Connection between the central installation and branch office is lost. In many cases, if this connection is lost, the endpoint becomes useless and all communication capabilities are lost.

With the Sangoma Resiliency Enablement Suite (SRES) features available on the Vega Gateways, a remote or branch office installation can have true resiliency in the event of a WAN failure. Even in the event of a WAN failure, IP Phones behind the Vega Gateway can continue to call each other, be routed to a backup switch or connected directly to the PSTN.

Branch Office Survivability

Connecting Legacy Equipment to an IP PBX

As enterprises continue to evolve their communications infrastructures into a pure SIP/VoIP environment, they often discover that there is still the need to connect older non-VoIP equipment to their VoIP infrastructure or directly to the PSTN. Many of these older, but necessary endpoints (i.e. fax machines, postage machines, alarm systems) simply will not function reliably over a VoIP network.

There may also be instances where it is more cost effective to use a "Plain Old Telephone" as an endpoint on a VoIP network, rather than an expensive VoIP phone. Ruggedized phones may already be in place on construction sites or in other locations where the risk of damage is great. Entry and elevator phones may already be integrated into existing systems that cannot be easily replaced by an IP endpoint.

A Sangoma Vega VoIP gateway can be configured to deliver a PSTN dial tone to endpoints that need it, or it can be configured to act as a proxy for plain old telephony, making it appear to the network as an IP endpoint.

Connecting Legacy Equipment to an IP PBX

SIP Trunking - Carrier

Our lower Density SIP Gateways are the only ones in their price class that deliver great voice quality, ease of configuration and failover/fallback features.

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SIP Trunking – Enterprise

SIP Dial Tone/Multi-Dwelling Units

Older Hotels, College Campuses and other Multi-Tenant installations have a huge investment in their structured wiring plans and existing Legacy Telephones. Replacing the wiring and every handset in a facility may not be a financial option, even when a core infrastructure upgrade to an IP-PBX would deliver an outstanding ROI.

The Sangoma Vega 5000 Gateway can be configured to allow Legacy Telephones to appear as IP Endpoints to an IP-PBX or Hosted IP-PBX Provider. This allows property owners to achieve the ROI of upgrading the core of their communications system to VoIP, without upgrading the legacy endpoints connected to the system or the structured wiring plan that connects these endpoint to the communications closet of each unit.

SIP Dial Tone/Multi-Dwelling Units

 


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3450 Lakeside Dr, Ste 145,
Miramar, FL 33027, USA

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3450 Lakeside Dr, Suite 145,
Miramar, FL33027, USA
T: +1 954.894.6181 F: +1 954.697.9707
 
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