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Obihai

OBi110
Voice Service Bridge & Telephone Adapter

with SIP, Google Voice™, OBiTALK, Telephone & Phone Line Interfaces

The Obihai OBi110 Voice Service Bridge and VoIP Telephone Adapter allows you to make and receive phone calls using your high-speed Internet, your regular home phone and a VoIP Internet phone service provider like Google Voice, Callcentric or VoIP.ms. With the Obihai VoIP, no computer is needed. You can also call for free from one OBi to another OBi anywhere in the world. With the Google Voice OBi110, a traditional phone service is never needed to make the connections so talking becomes truly free. Use this Obihai VoIP service to lower your phone bills. Add this Obihai OBi110 Voice Service Bridge and VoIP Telephone Adapter to make and receive phone calls to and from regular numbers. All you need is your existing home phone and broadband router.
How Does the OBi Work?
The OBi allows you to make phone calls using your high-speed internet, your regular home phone and a VoIP provider like Google Voice™. No computer is needed. You can also call for FREE from one OBi to another OBi anywhere in the world. In this case, a traditional phone service is never needed to make the connections. Talking becomes truly free.
The OBi110 Is Complemented by Other OBi Products & Services
  • OBiTALK: A customer portal for device management allowing members to add people and associated OBi endpoints to "circles of trust" such that additional functionality can be shared amongst authorized users. The OBiTALK portal is also where members can download the OBiAPP and OBiON applications for PCs and the iPhone, iPad, iPod touch & Android devices, respectively.
  • OBiON iPhone, iPad, iPod touch & Android Devices: An application for iPhone, iPad, iPod touch and Android devices which makes possible placing and receiving calls to/from other OBi endpoints.
  • OBiOn PC: A middleware application for a PC that facilitates placing and receiving calls to/from other OBi endpoints.

Product Specifications:

Obihai Part Number

  • OBi110

System Requirements

  • Active Internet Connection
  • Analog Touch Tone Phone
  • Access to Internet Via a Switched Ethernet Port on Home or Office Router
  • (Optional) Access to an Analog Telephone (POTS) Line.
  • (Optional) Active Internet Phone Service Subscription with All Required SIP
  • Credentials to Make & Receive Calls

Microprocessor

  • Obihai SoC, 32 bit, MIPS

Phone Port Configuration Settings

  • Recursive Digit Map & Associated Outbound Call Routing
  • On-Hook Tip Ring Voltage: 30v - 52v
  • Off-Hook Current Max: 15mA - 45mA
  • Impedance: 12 Independent Settings
  • DTMF Playback Level: -90 dBm - 3dBm
  • Caller ID Method (FSK Bell202, FSK V.23, DTMF FI/SE/DK)
  • Caller ID Trigger (Before / After First Ring, Polarity Reversal)
  • Channel Tx Gain: -12dB - 12dB
  • Channel Rx Gain: -12dB - 12dB
  • Silence Detect Sensitivity
  • Hook-Flash Time Max
  • Hook-Flash Time Min
  • CPC Delay Time
  • CPC Duration
  • Idle Polarity
  • Connect Polarity

Line Port Configuration Settings

  • Recursive Digit Map & Associated Inbound Call Routing
  • Ring Delay
  • Detect CPC
  • CPC Time Threshold
  • Detect Polarity Reversal
  • Detect Far End Long Silence
  • Detect Near End Long Silence
  • Silence Detect Sensitivity
  • Silence Time Threshold
  • Detect Disconnect Tone
  • Disconnect Tone Pattern - Programmable
  • AC Impedance: 16 Settings
  • On-Hook Speed: 0.5ms, 3ms (ETSI), 26 (AU)
  • Tip-Ring Voltage: 3.1v, 3.2v, 3.35v, 3.5v
  • Min Operational Loop Current: 10mA, 12mA, 14mA, 16mA
  • Current Limiting Enable
  • Channel Tx Gain
  • Channel Rx Gain
  • Line In-Use Voltage Threshold
  • Line In-Use Current Threshold
  • Caller ID Detect Method: FSK (Bell 202), FSK (V.23), DTMF (FI, SE, DK)
  • DTMF Playback Level

Local Access Interface

  • IVR, Web Page - Password Protected (Admin & User Level Log-in)

Remote Access Interface

  • Syslog (Multi-Level Granularity), Invokable via SIP Notify, Web, Provisioning

LED Indications

  • Power, Device Status, Upgrade in Progress Status, Ethernet Activity, Phone Port Status, Line Port Status

Ports

  • 1 x 10/100BaseT Ethernet Port (802.3)
  • 1 x RJ-11 FXS Analog Phone Port
  • 1 x RJ-11 FXO Analog Line Port
  • Reset Button - Located on Bottom of Case

VoIP

  • SIPv2 (RFC 3261, 3262, 3263, 3264)
  • SIP over UDP
  • SIP over TCP
  • SIP over TCP with TLS
  • 2 SIP Service Provider Service Sessions - Concurrent Operation
  • 2 XMPP (Google Voice) Sessions
  • 1 OBiTALK Service Session
  • SIP Proxy Redundancy - Local or DNS Based SVR, Primary & Secondary Fallback List
  • Restrict Source IP Address
  • Fail-over to FXO on Primary and/or Secondary SP Reg Failure - Selectable
  • Maximum Number of Sessions - Independent per Service
  • Trunk Groups (4)
  • Voice Gateway - Direct Dialing (8)
  • G.711 A-Law
  • G.711 μ-Law
  • G.726 (40/32/24/16)
  • G.729a
  • Codec Pre-selection Code
  • Voice Processing per SIP Service - TX/RX Audio Gain, Echo Cancellation
  • Adjustable Audio Frames per Packet
  • Codec Name Assignment
  • Codec Profile per SIP SP (2) & OBiTALK Service
  • Dynamic Audio Payload
  • Packet Loss Concealment
  • Jitter Buffer (Adaptive)
  • STUN
  • ICE
  • SUBSCRIBE / NOTIFY Framework (RFC 3265)
  • NOTIFY Dialog, Line Status
  • SUBSCRIBE Message Summary
  • VoIP NAT Interworking
  • DATE Header Support
  • ALERT-INFO Header Support
  • Remote-Party-ID (RPID) P-Asserted-Identity (PAID)
  • RTP Statistics in BYE Message
  • Media Loopback Support

Telephony

  • Configurable Contact List (Inbound Call Routing)
  • Automatic Attendant (English) with Configurable Answer Delay
  • PIN Access Control to AA (Up to 4 PINs)
  • Recursive Digit Map for Call Routing (AA, Line, Phone, Voice Gateways, Trunk Groups)
  • AA Configurable Outbound Call Routing Rule
  • SIP Service Configurable Inbound Call Routing Rule (2)
  • Direct / Single-Stage Dialing (Route to Voice Gateway)
  • Fax Pass Through (G.711)
  • Modem Pass Through (G.711)
  • In-Band DTMF (G.711)
  • Out of Voice Band DTMF (RFC 2833)
  • Out of Voice Band DTMF (INFO Method)
  • Call Progress Tone Generation
  • Tone Profile per SIP SP and OBiTALK service
  • Ring Profile per SIP SP and OBiTALK service
  • Star Code Profile per SIP SP and OBiTALK service
  • Full Duplex Audio
  • G.165, 168 Echo Cancelation
  • VAD - Voice Activity Detection
  • Silence Suppression
  • Comfort Noise Generation
  • Three Way Conference Calling with Local Mixing
  • Hook Flash Event Signaling
  • Flash Hook Timer
  • Caller ID - Name & Number per Bellcore, ETSI and DTMF
  • MWI - Message Waiting Indicator
  • Visual Message Waiting Indication (VMWI)
  • Daylight Savings Time Support - North & South Hemispheres
  • Caller ID Enable /Disable
  • Caller ID Number
  • Caller ID Name (Alphanumeric)
  • Caller ID Spoofing
  • Call Waiting
  • Maximum Session Control
  • Call Forward - Unconditional
  • Call Forward on Busy
  • Call Forward on No Answer (Ring Count Configurable)
  • Call Transfer Enable / Disable
  • Anonymous Call Block
  • Anonymous Call
  • Do Not Disturb
  • Call Return
  • Repeat Dialing

Certifications

  • FCC Part 15 Class B
  • FCC Part 68
  • A-Tick
  • CE
  • ICES-003
  • RoHS
  • WEEE
  • UL/cUL

Operating Temperature

  • 0° to 45° C (32° to 113° F)

Storage Temperature

  • -25° to 85° C (-13° to 185° F)

Operating Humidity

  • 10% to 90% Non-condensing

Non-operating Humidity

  • 10% to 90% Non-condensing

Dimensions

  • 4.5" x 4.2" x 1.2" (11.5 x 11.0 x 3.0 cm)

Weight

  • 9.0 oz. (255 g)

Warranty

      Manufacturer provided specifications for Obihai Technology, Inc. p/n OBi110 UPC#: 804879286677
Obihai OBi110 Voice Service Bridge & Telephone Adapter Mfr P/N: OBi110 OWC SKU: OBI110 Use your existing home phone and your broadband router to make and receive calls over the Internet. Obihai Technology 1 Year Limited Warranty
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