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Enhanced Cloud-based Unified Communications

PBXact UCC (Unified Cloud Communications) delivers cloud-based telephony services to small and medium sized businesses who are looking for a feature-rich, unified communications solution that can be tailored to meet their business demands.

For a monthly flat fee, PBXact UCC removes the burden of owning...

Enhanced Cloud-based Unified Communications

PBXact UCC (Unified Cloud Communications) delivers cloud-based telephony services to small and medium sized businesses who are looking for a feature-rich, unified communications solution that can be tailored to meet their business demands.

For a monthly flat fee, PBXact UCC removes the burden of owning and maintaining a costly on-premise PBX solution and takes ownership of infrastructure maintenance so you don’t have to, providing a full suite of services to your organization. Since PBXact UCC is a hosted service, no matter where you are your PBX is always available, from desk phone, softphone, mobile or laptop.

PBXact comes with an extensive set of Unified Communications features such as Phones Apps productivity features with Sangoma IP phones, UCP for web based communications and Zulu UC for desktop and CRM integration, all facilitating collaboration and productivity.

Benefits

Why PBXact UCC?

PBXact UCC offers not only the same benefits as our popular on-premise PBXact UC solution, but incorporates many powerful features that only the cloud can bring.

Voicemail

Each extension comes complete with voicemail. Users can call retrieve their voicemails from any device or through a web browser and their personal user control panel.

User Control

Each user gets access to their unique User Control Panel. This allows them to manage voicemails, set call forwards and configure other phone features like presence and do not disturb

Auto Attendant

Auto Attendants can be used to allow callers to directly dial an extension select or select a job function using their phone keypad.

Groups

Groups can be used to efficiently handle calls to a particular job function like sales or support. Team member’s phones can configured to ring either sequentially or simultaneously.

Conferences

With no limit on the number of conference rooms that can be created, it’s easy to empower your workforce to run conference calls with no extra expense

Open Standards

PBXact UCC works best with Sangoma phones, but it’s also possible to use any other type of standard SIP phone.

Why Cloud?

  • No PBX Hardware

PBXact is a fully hosted system, we take care of all the hassle of the infrastructure maintenance so you don’t have to.

  • Accessible

No matter where you are your PBX is always available. From deskphone, softphone, mobile or laptop.or through web browser with webrtc.

  • Reliability

Using the latest data centre technology together with rigourous back-ups and resiliency means that moving to the cloud reduces risk.

  • Future Proof

Using a managed service in the cloud means that you are buying a future proof solution. We’ll take care of all the upgrades so that you benefit from features as they become available.

  • Multi-Site

For businesses with more than one site a single instance of PBXact UCC can give you telephony service across all your sites with all users experiencing the same features from the same PBX.

Installation Wizard

The quick-start Installation Wizard for PBXact UC & PBXact UCC makes it incredibly easy to get your PBX set up with basic configuration in just a few minutes!

Here are some of the items it configures:

  • Extensions. Choose the number of extension and the range
  • AutoAttendant / IVR. Select how inbound calls should be routed at different times of day and even record voice prompts directly from your computer via the web browser.
  • Ring Groups. Decide how phones for each team should ring and what will happen to the call if no one picks up.
  • Fax. PBXact UCC can deal with FAX calls as well and it’s easy to decide how to route them.

Bring us your phone number or choose your own

PBXact UCC allows you to select new numbers or bring your own numbers with you – a service called porting.

PBXact UCC gives you the freedom to select as few or as many numbers as you need. If you need one inbound number, also known as a DID, you can have all your customers call that number and then use an IVR (Interactive Voice Response) to allow callers to dial an extension or a business function like sales or support. If your business requires you to use a public number to advertise, no problem, it’s easy to associate an inbound number directly with an extension. The caller will be routed directly to the extension.

Of course, it’s no problem to mix the two and have some users with DIDs and others accessible through an IVR

Build your solution with Sangoma IP Phones

Designed specifically for PBXact UCC, Sangoma’s line of IP Phones autoprovision themselves out-of-box using our Redirection/Auto-configuration. This eliminates the requirement of involving IT resources to manually configure network and user details. Auto-deployment also works for remote users too!

Included with all Sangoma IP Phones are Phone apps which allow user to control complex features directly from the phone’s colour display. Features such as Call Parking, Follow me, Conference rooms, Hot desking, Presence and many more. No more feature codes to remember!

Best of all, you can order your phones at the same time you purchase & configure your PBXact UCC services. This way you can have your phones delivered to you already pre-provisioned to work with your new setup.

PBXact UCC Feature Support Included in All Systems

Business Features

  • Flexible Time-Based Call Routing
  • Built-In Conference Bridge
  • Fax to E-mail
  • Hunt/Ring Groups
  • Music on Hold
  • Voicemail Blasting
  • Find Me / Follow Me Calling
  • Personal IVRs
  • Wake Up Calls
  • Support for Video Calling
  • Secure Communications (SRTP/TLS)
  • Announcements
  • Text to Speech
  • Calling Queues (ACD)
  • Interactive Voice Response (IVR)

Calling Features

  • Three-Way Calling Support
  • Voicemail
  • Voicemail to E-mail
  • Caller ID Support
  • Call Transfer
  • Call Recording
  • Do Not Disturb
  • Call Waiting
  • Call History / Call Detail Records
  • Call Event Logging
  • Speed Dials
  • Caller Blacklisting
  • Call Screening

Telephony Support

  • Open Standards Support for Multiple Protocols
  • SIP, IAX2, PRI, T1, E1, J1, R2, POTS/Analog, ISDN, GSM
  • WebRTC
  • Softphone Support
  • Specialty Device Support
  • Door Phones
  • Overhead Paging
  • Strobe Alerts
  • Paging Gateways
  • Voice Gateways
  • Failover Devices
  • Desktop/Mobile Phone Support

Administration

  • Upgrade System with Granular Control
  • Bulk Import Utilities (Trunks, Extensions, Users, DIDs)
  • Localization in both GUI and Sound Files for Multiple Languages
  • Backup and Restore Utilities
  • Custom Destination Administration
  • Web-based Config File Management When Needed
  • System Recording Management
  • GUI Controls for DNS, Network Settings, and More!

User Control Panel

  • Responsive GUI (Desktop, Tablet, and Mobile Device)
  • WebRTC Softphone
  • Call History (Details and Recording Playback / Download)
  • Contact Management
  • Presence Management
  • Conference Room Management
  • Settings Management
  • Find Me / Follow Me
  • Call Forwarding, Call Waiting, Do Not Disturb
  • Call Confirmation
  • Voicemail
  • Visual Voicemail – Playback and Management
  • Notification Options
  • Greetings Management

Add-ons

The Base Platform includes a base of system enhanced features (see chart below)

Additional functionality can be added as needed:

  • High Availability (License Required per PBX Node)
  • Call/Contact Center Features (Enhanced Call Center Functionality)
  • Operator Panel / Wall Boards
  • Third Party Phone Support (for Non-Sangoma IP Phones)

Why Choose Sangoma?

Sangoma’s customer-centric approach, product innovations, and worldwide network of distribution partners deliver the industry’s best-engineered, highest quality, IP and Unified Communications solutions, supporting “any app, anywhere” for businesses and service providers of all sizes.

All Sangoma products are backed by more than 30 years of IP communications experience, expert engineering and technical resources, and a comprehensive 1-year warranty. Extended warranties are also available.

More

Sangoma There are 27 products.

Subcategories

  • Analog VoIP Gateways
    Analog VoIP Gateways

    The Most Resilient VoIP Analog Gateways in Their Class

    Connecting your legacy telephony infrastructure to modern VoIP systems has been made easy with Sangoma’s line of Analog Vega VoIP Gateways. The Vega VoIP gateways seamlessly connect your analog phone system to IP networks. This is great for businesses with legacy phone equipment (such as an analog PBX) who want to connect to SIP trunking services without having to spend money altering their network infrastructure. They are also great for businesses that are already VoIP enabled at the core (with an IP- PBX) that need PSTN connectivity for emergency fallback. Simply place the Vega Gateway at the edge of your network, plug in your existing internet cable for VoIP connectivity and analog cables from your phone system and let the Vega automatically handle the SIP signaling and voice media conversion for seamless voice and T.38 Fax integration.

    Advanced Web GUI

    Features an intuitive Quick Wizard which does all the hard work for you for new deployments. Flexible dialplan to allow you to make your own routes, including automatic failure detection with failover routing.

    Diagnostic Tools

    Web GUI based PCAP tracing tool to capture full signaling and media, eliminating the need to connect equipment for line tracing, fully compatible with wireshark.

    Low and High Density Models

    Our low density analog models called Vega 50 are offered with 2-10 ports of FXS and FXO combinations. Our High Density Vega 3000G and Vega 3050G offer 24 and 50 FXS ports, respectively.

    Long Line Length

    Connect your analog lines up to 8 kms from the Vega Gateway. Great for applications where the IT closet is located far away from the analog devices

    Built-in Local Survivability

    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.

    Qualified for Microsoft

    Qualified for Lync 2013 and Skype for Business 2015.

    Which one do I Choose?

    • If you want to connect to SIP trunking service and have an analog PBX, the Vega 50 FXS is right for you Simply connecting the FXO lines from your PBX to the Vega 50 FXS ports for seamless connectivity.
    • If you want to provide emergency fallback to your VoIP network, the Vega 50 FXO model is right for you. Connect the PSTN telco lines to the FXO ports for automatic fallback routing when your VoIP network goes down.
    • If you want to connect a large quantity of analog phones to your IP-PBX, the Vega 3000G and Vega 3050G are right for you. These models are typically used hotels, schools, apartment buildings and other multi-dwelling environments to eliminate the need of change all the analog phones and infrastructure to IP.

    Vega Analog Gateway Models

    Vega Gateway models are one of the most reliable fault tolerant SIP-to-TDM media Gateways on the market, sized for your business needs. All Sangoma hardware carries a one year warranty with options to extend.

    Use Cases

    Enterprise VoIP Networking

    Use a Vega gateway to allow site to site networking for toll bypass between sites or to connect legacy and new infrastructure together.

    PSTN Trunking

    Use a Vega gateway to allow IP-PBX to route calls to traditional PSTN connections like PRI, BRI or analogue. Could be primary route or for resilience. 

    Low Density PSTN

    Connect your offices fax machines, door entry and modems to your company’s IP-PBX.

    Enterprise IP Telephony Gateway

    Using Enhanced Network Proxy (ENP) feature allows for local survivability of IP telephones. In case of SIP failure those onsite phones can talk to each other, the PSTN (if connected) and place emergency calls.

    SIP Trunking

    Vega Gateways allow traditional, legacy TDM based PBXs to replace expensive connections like PRI (E1T1) BRI or analogue with SIP trunking to allow reduced call costs, reduced line rental, and bring extra flexibility and disaster recovery.

    Multi-Dwelling

    For schools, hotels, apartment buildings, and military having large infrastructure of analog phones needing to connect to IP-PBX eliminating the need to replace any hardware or cabling.

    Analog Gateway Features

    Telephony Features

    Call Waiting
    Call Forward – Unconditional, Busy, No-Answer
    Call Transfer – Blind, Consultative
    3-Way Conference
    Do Not Disturb
    Message Waiting Indicator – Audible, Visual
    Music on Hold
    Executive Barge
    Caller ID presentation – UK, DTMF, Bellcore GR30, ETSI
    Caller ID screening
    SIP registration & digest authentication

    Call Quality

    Adaptive jitter removal
    Comfort noise generation
    Silence suppression
    802.1p/Q VLAN tagging
    Differentiated Services (DiffServ)
    Type of Service (ToS)
    QoS statistics reporting
    Echo cancellation (G.168 up to 128ms)

    Operations, Maintenance & Billing

    HTTP(S) Web Server
    RADIUS Accounting & Login
    Remote firmware upgrade:
    Auto code upgrade
    Auto configuration upgrade
    SNMP V1, V2 & V3
    Syslog
    TFTP/FTP support
    VT100 – RS232/Telnet/SSH
    Voice readback of IP parameters

    Security & Encryption

    Media – SRTP (Optional)
    SIP – TLS (Optional)
    Management – HTTPS, SSH, Telnet
    Configurable user login passwords
    Enhanced Network Proxy (ENP)

    Routing & Numbering

    Dial Planner – sophisticated call routing capabilities, standalone or gatekeeper/proxy integration
    Direct Dialing In (DDI)
    SIP registration to multiple proxies

    Survivability

    Automatic re-routing of call traffic to backup PSTN connection
    Local Survivability – Business Continuity during WAN/SIP outage

    Analog Gateway Specifications

    Interfaces

    VoIP Interface

    SIP
    Fax support – up to G3 FAX, using T.38
    Modem support – up to V.90, using G.711
    H.323 version 4 (vega50 only)
    H.323 version 4 (vega50 only)
    VoIP channel support:
    Vega 50:16
    Vega 3000G:24
    Vega 3050G:50
    Vega 4×4:16
    Audio Codecs:
    G.711 (a-law/µ-law) (64 kbps)
    G.729a (8kbps)
    G.723.1 (5.3/6.4 kbps)
    G.726
    Clearmode

    Telephony Interface

    Vega 50
    4 or 8 FXS ports with 2 FXO ports presented on RJ11
    4 or 8 FXO ports presented on RJ11
    600R, 900R or CTR-21 line impedance
    Vega 3000G
    24 FXS ports presented on 1 Amphenol connector
    600R, 900R or CTR-21 line impedance
    Vega 3050G
    50 FXS ports presented on 2 Amphenol connector
    2 FXO ports presented on optional expansion board
    600R, 900R or CTR-21 line impedance
    Vega 4×4
    FXS: Up to 16 ports presented on RJ45 600R, 900R or CTR-21 line impedance
    FXO: Up to 16 ports presented on RJ45 600R, 900R or CTR-21 line impedance
    GENGEN: Up to 16 ports presented on RJ45 75V RMS ringing
    AUDIO: Up to 16 ports presented on RJ45 Unbalanced audio input, receive (mic) and transmit (speaker)

    LAN Interface

    Vega 50
    2 RJ-45s, 10 BaseT / 100 BaseTX, full / half duplex
    Vega 3000G
    1 RJ-45, 1000BaseT/100 BaseTx/10 BaseT, full/half duplex
    Vega 3050G
    1 RJ-45, 1000BaseT/100 BaseTx/10 BaseT, full/half duplex
    Vega 4×4
    2 RJ-45s, 1000BaseT / 100 BaseTX / 10 BaseT, full / half duplex

    Hardware

    Certification
    • EMC (CLASS B)
      • EN55022
      • EN55024
      • FCC Part 15
      • AS/NZS3548
    • TELECOMS (ISDN)
      • E1: TBR4
      • T1: FCC Part 68
      • T1: CS-03
      • VCCI
    • Safety
      • EN60950
      • IEC60950
      • UL60950
      • AS/NZS60950
    Dimensions
    • Vega 50
      • 300mm (w) x 237mm (d) x 45mm (h)
      • weight 1kgs (2.2lbs)
      • Rackmount ears supplied
    • Vega 3000G
      • 1U: 270mm (W) x 155mm (D) x 43mm (H)
      • Weight: 1.5kgs (3.31lbs)
      • Rackmount ears supplied: 107mm (2 pieces)
    • Vega 3050G
      • 1U: 437mm (W) x 275mm (D) x 43mm (H)
      • Weight: 6.5kgs
      • Rackmount: brackets supplied 483mm (19″)
    • Vega 4×4
      • 1U: 437mm (W) x 275mm (D) x 43mm (H)
      • Weight: 6.5kgs
      • Rackmount: brackets supplied 483mm (19″)
    Power Supply
    • Vega 50
      • External AC adapter
      • 100–240 VAC (50/60 Hz)
      • DC output 12V/5A (60W)
    • Vega 3000G
      • External AC adapter
      • 100–240 VAC (50/60 Hz)
      • DC output 12V/5A (60W)
    • Vega 3050G
      • Internal PSU
      • 24 ports: 100..240 VAC, 47..63 Hz, 1..0.5 A
      • 48 ports: 1100..240 VAC, 47..63 Hz, 2..1 A
      • -48V DC, 1.2A (Max) available subject to MOQ (Optional)
    • Vega 4×4
      • Internal PSU
      • 24 ports: 100..240 VAC, 47..63 Hz, 1..0.5 A
      • 48 ports: 1100..240 VAC, 47..63 Hz, 2..1 A
      • -48V DC, 1.2A (Max) available subject to MOQ (Optional)

    LED Indicators

    • System: Power/System Ready/Activity
    • LAN: Speed/activity
    Environmental
    • 0° .. 40°C
    • 0% .. 90% humidity (non-condensing)
    FXS Line Length
    • Up to 8km at 1 REN

  • VoIP Digital Gateways
    VoIP Digital Gateways

    The Most Resilient VoIP Digital Gateways in Their Class

    Connecting your legacy telephony infrastructure to modern VoIP systems has been made easy with Sangoma’s line of SIP-to-TDM Vega VoIP Gateways. The Vega VoIP gateways seamlessly connect your PRI (T1/E1) or BRI phone system to IP networks. This is great for businesses with legacy phone equipment (such as a TDM PBX) who want to connect to SIP trunking services without having to spend money altering their network infrastructure. They are also great for businesses that are already VoIP enabled at the core (with an IP- PBX) that need PSTN connectivity. Simply place the Vega Gateway at the edge of your network, plug in your existing internet cable for VoIP connectivity and E1/T1 or BRI cables from your phone system and let the Vega automatically handle the SIP signaling and voice media conversion for seamless voice and T.38 Fax integration.

    Advanced Web GUI
    Features an intuitive Quick Wizard which does all the hard work for you for new deployments. Flexible dialplan to allow you to make your own routes, including automatic failure detection with failover routing.

    Diagnostic Tools
    Web GUI based PCAP tracing tool to capture full signaling and media, eliminating the need to connect equipment for line tracing, fully compatible with wireshark.

    Low and High Density Models
    The Vega 100G and Vega 200G are our low density models with a maximum capacity for 30 and 60 SIP-TDM simultaneous calls. The Vega 400G is our high density model and the most flexible field upgradable unit for a maximum capacity of 120 simultaneous SIP-TDM calls.

    E1/T1 & BRI Interface
    Each E1/T1 interface (for Vega 100G, 200G, 400G) and BRI interface (Vega 50 BRI) can be independently configured as network side or terminal side. The Vega gateway can therefore be connected to a PBX or the PSTN.

    Built-in Local Survivability
    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.

    Qualified for Microsoft
    Qualified for Lync 2013 and Skype for Business 2015.

    Vega Digital Gateway Models

    Vega Gateway models are one of the most reliable fault tolerant SIP-to-TDM media Gateways on the market, sized for your business needs. All Sangoma hardware carries a one year warranty with options to extend.

    Use Cases

    Enterprise VoIP Networking
    Use a Vega gateway to allow site to site networking for toll bypass between sites or to connect legacy and new infrastructure together.

    PSTN Trunking
    Use a Vega gateway to allow IP-PBX to route calls to traditional PSTN connections like PRI, BRI or analogue. Could be primary route or for resilience.

    Low Density PSTN
    Connect your offices fax machines, door entry and modems to your company’s IP-PBX. 


    Local Survivability
    Using Enhanced Network Proxy (ENP) feature allows for local survivability of IP telephones. In case of SIP failure those onsite phones can talk to each other, the PSTN (if connected) and place emergency calls. Discover More

    SIP Trunking
    Vega Gateways allow traditional, legacy TDM based PBXs to replace expensive connections like PRI (E1T1) BRI or analogue with SIP trunking to allow reduced call costs, reduced line rental, and bring extra flexibility and disaster recovery.

    PSTN connectivity for MS Lync / SFB Infrastructure
    Vega Media Gateways qualified and tested for use with Microsoft Lync allow Lync to reliably and securely connect to PSTN trunks and legacy PBXs.

    Digital Gateway Features

    Telephony Features

    Caller ID presentation
    Caller ID screening allows connections to be accepted only from selected call sources
    SIP Registration & Digest Authentication

    Call Quality

    Adaptive jitter removal
    Comfort noise generation
    Silence suppression
    802.1p/Q VLAN tagging
    Differentiated Services (DiffServ)
    Type of Service (ToS)
    QoS statistics reporting
    Echo cancellation (G.168 up to 128ms)

    Operations, Maintenance & Billing

    HTTP(S) Web Server
    RADIUS Accounting & Login
    Remote firmware upgrade:
    Auto code upgrade
    Auto configuration upgrade
    SNMP V1, V2 & V3
    TFTP/FTP support
    VT100 – RS232

    Security & Encryption

    Management – HTTPS, SSH, Telnet
    Configurable user login passwords
    SIP/TLS and SRTP

    Routing & Numbering

    Dial Planner – sophisticated call routing capabilities, standalone or gatekeeper/proxy integration
    Direct Dialing In (DDI)
    SIP registration to multiple proxies
    NAT traversal

    Redundancy/Survivability:

    Hardware failover using port bypass (Vega 400G only)
    Local Survivability – Business Continuity during WAN/SIP outage

    Digital Gateway Specifications

    Interfaces

    VoIP Interface

    SIP
    Fax support – up to G3 FAX, using T.38
    Modem support – up to V.90, using G.711
    VoIP channel capacity:
    Vega 100G:30
    Vega 200G:60
    Vega 400G up to 120
    Vega 50 BRI
    2 BRI interfaces (4 channels)
    4 BRI interfaces (8 channels)
    8 BRI interfaces (16 Channels)
    Audio Codecs:
    (a-law/µ-law) (64 kbps)
    G.723.1 (5.3/6.4 kbps)
    G.729a (8kbps)
    G.726
    T.38

    Telephony Interface

    Primary Rate ISDN (User configurable NT/TE):

    T1
    NI1/NI2
    AT&T 5ESS
    CAS (RBS)
    DMS100
    ISO QSIG
    CAS Private Wire (*400G)
    E1
    Euro-ISDN
    ISO QSIG
    VN4
    CAS R2MFC
    CAS Private Wire (*400G)
    BRI
    2, 4 or 8 S/T interfaces presented on RJ45
    Point-to-point or point-to-multipoint
    Each interface can be configured NT or TE

    Open-Non Proprietary

    ETSI, VN4, ISDN
    NI1, NI2, AT&T 5ESS, DMS100
    ISO QSIG Basic Call and QSIG Feature Transparency
    Channel Associated Signalling (CAS)
    R2 MFC

    LAN Interface

    2x RJ-45, 1000BaseT / 100BaseTx / 10BaseT, full / half duplex

    Hardware

    Certification

    EMC (CLASS B)
    EN55022
    EN55024
    FCC Part 15
    AS/NZS3548
    TELECOMS (ISDN)
    E1: TBR4
    BRI: TBR3
    T1: FCC Part 68
    T1: CS-03
    VCCI
    Safety
    EN60950
    IEC60950
    UL60950
    AS/NZS60950

    Dimensions

    Vega 100G/ 200G
    70mm (W) x 155mm (D) x 43mm (H)
    Weight: 1.20kgs (2.64lbs)
    Rackmount ears supplied: 107mm (2-pieces)
    Vega 400G
    437mm (W) x 153mm (D) x 43.5mm (H)
    Weight: 1.97kgs (4.35lbs)
    Rackmount ears supplied
    Vega 50 BRI
    299.73mm (W) x 237.06mm (D) x 44.45mm (H)
    Weight: 1.02kg (unit only)
    Rackmount ears supplied

    Power Supply

    Vega 100G/ 200G
    External AC-to-DC power brick 100..240 VAC, 47..63 Hz, 1..0.5 A
    Vega 400G
    Internal PSU 100..240 VAC, 47..63 Hz, 1..0.5 A
    Vega 50 BRI
    External AC-to-DC power brick 100..240 VAC, 47..63 Hz, 1..0.5 A

    LED Indicators

    Power
    ISDN: NT/TE & link up
    LAN: Speed/activity

    Environmental

    0° .. 40°C
    0% .. 90% humidity (non-condensing)

  • NetBorder VoIP Gateways
    NetBorder VoIP Gateways

    The Most Flexible Carrier-Grade High Density VoIP Gateway

    Although the telecommunications industry is rapidly adopting the Internet Protocol to provide Voice-over-IP (VoIP) services, the legacy PSTN network (using T1/E1 telco lines) is still prevalent for world-wide communication. As such there is a growing need for equipment that can seamlessly bridge legacy PSTN and VoIP services, whilst offering a variety of protocol support.

    The Sangoma VoIP Media Gateway is the answer to this challenge offering seamless interconnection of next generation VoIP and the legacy PSTN networks, scalable up to 32 T1/E1 lines (or 960 simultaneous calls) in a 1 U form factor. Supporting concurrent use of ISDN, CAS (R2), SIP, T.38 fax Sangoma’s on-premise VoIP Gateway appliances are perfect for service providers migrating from legacy services to IP-centric architectures and large enterprises & call centers to deliver high-volume SIP trunking. Integrated transcoding capabilities also guarantees NVG to interop with any open source or propriety PBX solution and increase ROI by compression VoIP traffic.

    Advanced Web GUI

    Browser-based GUI use for configuration, control and monitoring of PRI links and network connections and load. Easy to use dialplan routing for basic and advanced call routing configurations. System logs and reports are also available directly from the GUI.


    Diagnostic Tools

    Browser based dashboard featuring time based graphing, system and session error reporting and email notifications upon faults. Onboard PCAP training tool capturing signaling and media, eliminating the need for extra port mirrors or hubs. The Hardware Crash protection feature reboots the system on lockup or hardware fault.


    Scalable and Flexible


    NVG is future proof with field upgradable licensing for up to 32 T1/E1s or 960 simultaneous calls in a 1 U platform and flexible with routing between any PRI, CAS (R2) and SIP interconnection using its advanced XML routing engine.


    Simplified Licensing

    All NSG appliances are field upgradable and licensing is per channel, which means all the features are always included – No sticker shock. Predictable NSG capacity and cost in every use case.


    Proven Interoperability with Global installations

    NVG has been deployed globally, which can attest to its highly successful interoperability and integration with PRI and VoIP carriers. On-board transcoding capabilities ensure interoperability with any open-source or proprietary PBX. So you never have to worry about new deployments wherever they may be.


    Enhanced Media Capability

    NVG has extensive support for various wireline, mobile and audio codecs which guarantees interoperability with service providers, enriched audio quality and improved ROI. NVG comes with support for AMR, AMR-WB (G.722), GSM, G729AB, G.723, G.726 and iLBC, all configurable on a per channel basis. Extensive support for T.38 protocol which is used for Fax-Over-IP comes standard.

    Use Cases


    SIP Network to ORI Interconnect

    Provide High value VoIP services to your customers.

    High-Density Call Completion Platform

    Provide competitive long distance and large scale call completion applications using cost-effective.

    Migrate Legacy Infrastructure to VoIP Services

    For large Enterprises, Carriers and Call Centers using legacy PRI/MFCR2 equipment, reduce telephony costs by seamlessly connecting an NVG appliance to access VoIP based services.

    PSTN Trunking for Large Call Center

    Connect your Large IP-PBX Based call centers to your trusted PRI/MFCR2 TDM trunks.

    NetBorder Gateway Features

    Echo Cancellation

    DTMF detection and generation:
    RFC2833 tone relay
    In-band
    SIP info
    G.168-2002 with 128ms tail

    VoIP Protocols

    SIP V2 / RFC3261
    SCTP RFC 2960

    PSTN Protocols

    Primary Rate ISDN (PRI)
    E1: Euro-ISDN, CAS MFCF2
    T1: Q.931, NI2, 4ESS, 5ESS, DSM-100

    Call Routing

    Configurable and extendable XML-based dial plan and routing rules

    Capacities

    Flexible 8 port appliance
    8 E1/T1 ports, or 240 channels
    16 port appliance
    16 E1/T1 ports, or 480 channels
    32 port appliance
    32 E1/T1 ports, or 960 channels

    Debugging

    Dedicated Browser interface for capturing full RTP media and signaling information
    Onboard browser-based PCAP tracing, signalling and media – wireshark compatible
    Large onboard storage capacity for long term tracing

    Audio Codecs

    G.711
    G.723.1
    G.726
    iLBC
    G.729AB
    GSM
    G.722
    AMR
    G.722.1
    Fax Support; T.38 Fax Relay

    Management & Configuration

    Web GUI
    Command line interface
    Detailed logs with con gurable le size and autorotation
    SNMP
    Radius
    Call detail records in XML format
    Remote firmware upgrade

    NetBorder Gateway Specifications

    Interfaces

    PSTN Interfaces

    8 port appliance
    8 E1/T1; One 8-port telephony interface (RJ45 ports)
    16 port appliance
    16 E1/T1; Either two 8-port telephony interface (RJ45
    ports) or one 16-port telephony interface (16-port interface requires 68-pin SCSI type interface)
    32 port appliance
    32 E1/T1: Two 16-port telephony interfaces (16-port interface requires 68-pin SCSI type interface)

    Network Interface

    8 port appliance
    5x LAN 10/100/1000 BaseT Ethernet ports
    IPV4, IPV6
    1x IPMI remote management Interface
    2x USB ports
    1x console port
    16/32 port appliance
    7x LAN 10/100/1000 BaseT Ethernet ports
    IPV4, IPV6
    1x IPMI remote management Interface
    2x USB ports
    1x console port

    Hardware

    8 Port Appliance

    1U: 430mm (W) x 305mm (D) x 45mm (H)
    7.2kgs (16lbs)
    60W universal AC power

    16 Port Appliance

    1U: 430mm (W) x 470mm (D) x 45mm (H)
    15kgs (33lbs)
    250W universal AC power

    32 Port Appliance

    1U: 430mm (W) x 470mm (D) x 45mm (H)
    15kgs (33lbs)
    250W universal AC power

  • NetBorder SS7 VoIP Gateways
    NetBorder SS7 VoIP...

    The Most Flexible Carrier-Grade Solution for SS7 to VoIP interconnect

    Although the telecommunications industry is rapidly adopting the Internet Protocol to provide Voice-over-IP (VoIP) services, the legacy PSTN SS7 network (using T1/E1 telco lines) is still prevalent for world-wide communication. As such there is a growing need for equipment that can seamlessly bridge legacy PSTN and VoIP services, whilst offering a variety of protocol support.

    The Sangoma NetBorder SS7 VoIP Media Gateway is the solution to this challenge offering seamless interconnection of next generation VoIP and the legacy SS7 networks. Sangoma’s line of Carrier-Grade SS7 VoIP Gateways are scalable, cost-effective TDM and SIP-over-IP telephony solutions supporting up to 64 T1/E1 lines, or 1920 concurrent calls. Supporting concurrent use of ISDN, SS7 CAS (R2), SIP, and SIGTRAN signaling protocols, Sangoma’s on-premise SS7 VoIP Gateway appliances connect traditional SS7 edge, core and peer zones with new-generation, MEGACO and SIP networks with transcoding capabilities. Bundling together the signaling and media gateway controllers makes this an ideal solution to reduce footprint and need to source components from multiple vendors.

    Advanced Web GUI

    Browser-based GUI use for configuration, control and monitoring of SS7 links and network connections and load. Easy to use dialplan routing for basic and advanced call routing configurations. System logs and reports are also available directly from the GUI.


    Software Options

    For integration within existing data centre infrastructures, NSG is also offered in a software platform version which contains the same feature- rich capabilities as our scalable appliances and compatible in all commercially available servers and Linux distributions.


    Scalable and Flexible

    NSG is future proof with field upgradable licensing for up to 64 T1/E1s or 1920 simultaneous calls in a 1 U platform and flexible with routing between any PRI, SS7 and/or SIP interconnection using its advanced XML routing engine.


    Simplified Licensing

    All NSG appliances are field upgradable and licensing is per channel, which means all the features are always included – No sticker shock. Predictable NSG capacity and cost in every use case.


    Proven Interoperability with Global installations

    NSG has been deployed globally, which can attest to its highly successful interoperability and integration with SS7 and VoIP carriers. On-board transcoding capabilities ensure interoperability with any open-source or proprietary PBX. So you never have to worry about new deployments wherever they may be.


    Diagnostic Tools

    Browser based dashboard featuring time based graphing, system and session error reporting and email notifications upon faults. Onboard PCAP training tool capturing signaling and media, eliminating the need for extra port mirrors or hubs. The Hardware Crash protection feature reboots the system on lockup or hardware fault.

    NetBorder SS7 VoIP Gateway Models

    Scalable and flexible, Sangoma’s NetBorder SS7 VoiP Gateways offer concurrent use of ISDN, SS7 CAS (R2), SIP, and SIGTRAN signaling protocols all in one package, eliminating the need for third party media gateway controllers or protocol converters lowering operation and maintenance costs.

    NetBorder SS7 Gateway Software

    For integration within existing data centre infrastructures, NSG is also offered in a software platform version which contains the same feature- rich capabilities as our scalable appliances and compatible in all commercially available servers and Linux distributions.

    Software Platform:

    • Operating System: 32-bit and 64-bit Linux (CentOS recommended)
    • Server: Varies with size of deployment

    Use Cases

    Mobile Based Services

    Provide SMS, MMS and location bases services (through HLR) by interconnecting directly into the mobile network which uses SS7 signaling.

    High-Density Call Completion Platform

    Provide competitive long distance and large scale using cost-effective NetBorder SS7 Gateways with signaling and media combined in a single platform.

    SIP Network to SS7 Interconnect

    Provide High value VoIP services to your customers.

    Caller Ring Back Tone (CRBT) Solution

    Carriers can provide Color Ring Back Tone to their customers as an additional service.

    SS7 Gateway Features

    Capacities

    Flexible 8 port appliance
    Up to 8 E1/T1 ports, or 240 channels
    8-32 port appliance
    Up to 32 E1/T1 ports, or 960 channels
    64 port appliance
    Up to 64 E1/T1 ports, or 1920 channels
    Software platform
    Up to 64 E1/T1 ports, or 1920 channels

    VoIP Protocols

    SIP V2 / RFC3261
    SCTP RFC 2960
    SIGTRAN M2UA RFC 3331
    Megaco / H.248

    PSTN Protocols

    SS7-ISUP
    ITU, ANSI, Bellcore, UK, China, India, France and
    Russian variants
    Up to 16 A or F signaling links & linksets
    Up to 16 originating & destination point codes
    ISUP relay for larger distributed network configurations
    E1: Euro-ISDN, CAS MFCF2
    T1: Q.931, NI2, 4ESS, 5ESS, DSM-100

    Audio Codecs

    G.711
    G.723.1
    G.726
    iLBC
    G.729AB
    GSM
    G.722
    AMR
    G.722.1
    Fax Support; T.38 Fax Relay

    Echo Cancellation

    DTMF detection and generation:
    RFC2833 tone relay
    In-band
    SIP info
    G.168-2002 with 128ms tail

    Debugging

    Dedicated Browser interface for capturing full RTP media and signaling information
    Onboard browser-based PCAP tracing, signalling and media – wireshark compatible
    Large onboard storage capacity for long term tracing

    Call Routing

    Configurable and extendable XML-based dial plan and routing rules
    Management & Configuration
    Web GUI
    Command line interface
    Detailed logs with con gurable le size and autorotation
    SNMP
    Radius
    Call detail records in XML format

    SS7 Gateway Specifications

    Interfaces

    PSTN Interfaces

    Flexible 8 port appliance / 8-32 port appliance
    4 E1/T1; One 4-port telephony interface (RJ45 ports)
    8 E1/T1; One 8-port telephony interface (RJ45 ports)
    16 E1/T1; Either two 8-port telephony interface (RJ45 ports) or one 16-port telephony interface (16-port interface requires 68-pin SCSI type interface)
    32 E1/T1: Two 16-port telephony interfaces (16-port interface requires 68-pin SCSI type interface)
    64 port appliance
    64 E1/T1; Four 16-port telephony interfaces (16-port interface requires 68-pin SCSI type interface)

    Network Interface

    Flexible 8 port appliance

    5x LAN 10/100/1000 BaseT Ethernet ports
    IPV4, IPV6
    1x IPMI remote management Interface
    2x USB ports
    1x console port
    8-32 port appliance
    7x LAN 10/100/1000 BaseT Ethernet ports
    IPV4, IPV6
    1x IPMI remote management Interface
    2x USB ports
    1x console port

    64 port appliance

    4x LAN 10/100/1000 BaseT Ethernet ports
    IPV4, IPV6
    1x IPMI remote management Interface
    Video: 1x VGA port

    Hardware

    Flexible 8 Port Appliance

    1U: 430mm (W) x 305mm (D) x 45mm (H)
    7.2kgs (16lbs)
    60W universal AC power

    8-32 Port Appliance

    1U: 430mm (W) x 470mm (D) x 45mm (H)
    15kgs (33lbs)
    250W universal AC power

    64 Port Appliance

    2U: 723mm (W) x 347mm (D) x 89mm (H))
    40.8kgs (90lbs)
    1000W universal AC power

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