Sark PBX SARK1200 IP PBX (60-150 users) Single PRI

0 Write Review No reviews

Shipping Brand Sark SKU SARK1200-60-150-1PRI

£4,405.50 exc. VAT £5,286.60 inc. VAT Select currency Quantity

Availability: In Stock

Note: This product comes with Echo Cancellation Daughterboard

SARK UCS is a new kind of PBX platform, built to cope with medium to high workloads on existing copper and fibre-based TDM networks. It is equally at home in the 21CN world of SIP and VoIP. Designed and developed for the UK market by UK telephony people, the platform is reliable, fast and has a low cost-of-ownership when compared with traditional TDM and proprietary IP offerings.

SARK UCS also has many unique features which give it a substantial edge over its competitors, particularly in the areas of high-availability and remote platform support. Issues that are of particular importance to those users whose businesses depend upon high levels of customer interaction and care.

Switching Performance

SARK UCS uses its own on-board high-speed logic and rules engine (the HSLE). The HSLE is very small and fast, being optimised to switch high volumes of in-bound and out-bound calls in the minimum number of cycles. The end result is much faster switching decisions from SARK when compared to its competitors. Thanks to HSLE, SARK UCS can comfortably sustain high call arrival rates on relatively low CPU power, making it ideal for deployment into high volume or high “spike” environments.

''FlatPack'' Turn-Key Solution for Fast and Easy Deployment

With the VoIPVn developed configurator documentation, VoIPVn resellers can forward order PBX systems with all key provisioning and routing information pre-loaded. This combined with the PBX’s autoprovisioning software (available for most popular SIP phone types) means installation and deployment is painless, error free and normally completed in just a few hours.

Remote Support / Management

VoIPVn systems have been developed with the most advanced remote management, configuration and diagnostic capabilities of any PBX in their class. Tools like the unique SIP Dynamic Proxy capability plus automatic recognition and adoption of new SIP devices, enable effective system and terminal management, and make on-going adds, moves and changes much easier for both customer and supplier

Reliability – High Availability (HA)

Where high levels of reliability and uptime are of the utmost importance, as for example in call centre applications, the VoIPVn SARK advanced High Availability option can be of huge business value. The HA cluster comprises two PBX servers with identical configurations running side by side and connected by a ''heart beat'' mechanism. In the event of a failure of the primary system, fail-over to the secondary takes just a few seconds and requires no reprogramming or restart of either the phones, the VoIP accounts, gateways or ISDN circuits. The total downtime in such situations is usually around 12 to 20 seconds from initial failure to resumption of operations. When compared to the minimum 4 hour callout, or next business day support terms available from our competitors, this one feature alone can make a huge difference to the bottom line in call-critical businesses.

Feature List

Call Handling

• Call hold (with music)
• Blind/Attended Transfer with optional return on no answer
• Call “camp” on extension
• Call re-direct or shunt
• Call pickup groups
• Call parking
• Call Forwarding - (on busy, no answer or unconditional)
• Call waiting (multi line handsets)
• Do not disturb
• IVR/Automated attendant
• Multi-level automated attendants for routing of calls more efficiently (i.e. press 1 for etc...)
• Direct dial to extension
• Interactive directory name/number lookup
• Direct record from handset or upload messages
• Multilevel user access for security

Call queuing

• Call queuing with strict answer ordering
• Choice of call distribution methods including (ring all, round robin, round robin memory, least recently called first, fewest number of calls first or random)
• Static and dynamic agents to allow users to log-in/out of the call queues
• Caller announcements including queue position and estimated hold time.
• Link to IVR for in queue options
• Min. and Max. people in queue settings
• Max. wait time for queued calls
• Choice of hold music

Voice mail

• Voicemail per extension (no port limits)
• Message retrieval by phone (local or remote) & web browser
• Email notification (with sound file if required)
• Visual and stutter dial tone message waiting indication (handset dependant)
• Group mailboxes
• Busy and Unavailable Messages
• Message forwarding and append options
• Multiple mailboxes for filing / storage

Advanced call routing

• Automatic fail over for outbound calls
• Provider preferencing for least cost routing
• Dialled number manipulation pattern match, add/subtract digits)
• Route password protection (class of service)
• Inbound call routing (either number called, or calling number)
• Direct SIP inbound/outbound calling support or ENUM/SIP URI calling
• Transcoding between codecs for maximum compatibility

Handset features (model dependant)

• Compatible with SIP compliant handsets
• 3 way conferencing
• Busy lamp field support
• Stutter dial tone for message waiting
• Missed/Dialled and Received calls logging
• Remote workers
• Caller ID name & number support
• Multiple codec support
• Personal phone book with distinctive ring
• Global address book linked to LDAP/XML existing database

Call distribution

• Flexible extension numbering
• Ring groups with timers and fail overs for call distribution including off site calling
• DDI inbound routing support
• Route based on Caller ID and/or DDI
• Time/Day/Date routing

Conferencing

• 3 way conferencing (handset dependent)
• Conference rooms for larger conferences
• Pin access
• Member announcements (join/leave)
• Mute/Un-mute per user

Reporting

• Detailed call logging with selection, search and CSV export
• Call comparison over time graphs
• Monthly traffic graphing
• Daily load graphing

Administration/usability

• Easy web based interface for administration local/remote
• Multi user/multi level access for delegated/departmental control
• Web based operator’s console

Music on hold

• Multiple music on-hold tracks (mp3)
• Multiple music categories

Call Recording Option

• Selective recording at system, group or individual extension
• One-Touch on-demand recording
• One-Touch Retrospective on-demand recording option
• Real time pause and resume option
• High volume in-RAM record-and-offload option

High Availability Option

• Tandem system
• Fully automatic failover in the event of software or hardware failure
• ISDN BRI/PRI failover option (hardware card)
• Automatic failback when prime node recovers
• Failover in less than 20 seconds (on regular ISDN30 circuits)

System integration

• Tapi compliant (multiple drivers available)
• Inbuilt Manager API for custom integration/control
• Outlook click to call support (via Tapi)

Multi provider/trunk support

• Multiple inbound phone lines/trunks
• Multiple technologies including:
• Analogue, ISDN2e, ISDN30e, IP Trunks (SIP), ENUM (e.164.arpa & .org)

Expandable

• Custom programming language for system expansion and integration
• Media Link—connect multiple managers together across multiple sites providing one large integrated phone system across multiple sites/locations
• Say/Spell Engine (custom programming)
• Software upgradeable