VOICE CALLS OVER SIP TRUNKS

What are SIP Trunks?
Session Initiation Protocol (SIP) is a signaling protocol. It is widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP). SIP (Session Initiation Protocol) Trunks allow businesses to retain their existing telecoms equipment, whilst gaining the additional uses and benefits of VoIP Telephony. Typically, SIP Trunks enable costly standard or ISDN (BRI or PRI) telephone lines to be replaced by a single IP connection. SIP Trunking can be linked directly to an onsite IP-PBX or to a traditional PBX using a Gateway. Outgoing calls are then routed through the PBX and out via the SIP Trunk, rather than using costly traditional phone lines. This efficient way of combining voice and data onto one Internet connection will make your life easier, while enjoying free on-network calling between your corporate locations.
Who are they for?
SiP Connect is ideal for companies with multiple sites. It can scale easily and with your needs and is capable of supporting organisations with any number of employees. Companies looking to utilise their existing PBX and internal infrastructure but reduce monthly connection charges, calling costs and maintenance fees. Companies who want to benefit from the feature rich advantages of VoIP such as online and instant call records & bills, self-managed call forwarding and number diversion.
Panasonic KX-UT670
  • SIP Tablet Phone
  • 7" Color Touchscreen
  • Android Operating System
  • USB Port
  • SD Card Slot
  • Power Over Ethernet (PoE) Support
  • 2 Ethernet Ports
  • 6 SIP Accounts
  • HD Video and Voice
  • Bluetooth Enabled
  • 24 Programmable Keys
  • 100 Entry Phonebook
  • 3 Way Conferencing
  • Integration w/ IP Cameras
Panasonic KX-UT248
  • Executive SIP Phone
  • 4.4" Grayscale Graphical LCD
  • 6 SIP Accounts
  • HD Voice
  • 24 Flexible Buttons
  • 2 Ethernet Ports
  • 3 Way Conference Call
  • Built-In Bluetooth
  • Plug and Play Configuration
  • Full Duplex Speaker Phone
Panasonic KX-UT133
  • SIP Corded Phone
  • 3-Line Backlit LCD Display
  • 4 SIP Accounts
  • HD Voice & Power Over Ethernet (PoE) Support
  • Two Ethernet Ports
  • 24 Programmable Buttons
  • 12 Feature Buttons
  • Hearing Aid Compatible
  • 500 Phone Book Entries
  • Environmentally Friendly
Panasonic KX-UT136
  • SIP Corded Phone
  • 6-Line Backlit LCD Display
  • 4 SIP Accounts
  • HD Voice & Power Over Ethernet (PoE) Support
  • Two Ethernet Ports
  • 24 Programmable Buttons
  • 12 Feature Buttons
  • Hearing Aid Compatible
  • 500 Phone Book Entries
  • Environmentally Friendly
Polycom 335
Entry-level SIP phone with Polycom HD Voice
  • 2 Lines
  • Certified to inter-operate with a range of SIP platforms
  • Integrated PoE support (IEEE 802.3af)
  • Polycom HD Voice for all audio paths
  • Full-duplex speakerphone with Acoustic Clarity Technology
  • 102 x 33 pixel graphic LCD
  • Built-in XML micro-browser
  • Two port 10/100 Ethernet switch
Polycom 550
Cutting edge SIP feature set with Polycom HD Voiceâ„¢
  • 4 Lines
  • Integrated PoE support (IEEE 802.3af)
  • Full-duplex speakerphone with Acoustic Clarity Technology
  • Polycom HD Voice for all audio paths
  • 320 x 160 pixel backlit greyscale graphical LCD
  • 26 dedicated hard keys and 4 context sensitive soft keys
  • Inbuilt Gigabit Ethernet support
Polycom 670
A premium, SIP desktop phone with colour display delivering a rich voice, visual and applications experience.
  • 4 Lines
  • Integrated PoE support (IEEE 802.3af)
  • Full-duplex speakerphone with Acoustic Clarity Technology
  • Polycom HD Voice for all audio paths
  • 320 x 160 pixel backlit greyscale graphical LCD
  • 26 dedicated hard keys and 4 context sensitive soft keys
  • Inbuilt Gigabit Ethernet support