If you are tired of paying for high-end Cisco or Polycom phones, and I know some of you are, here’s a list of low cost commodity SIP phones and adapters. So, if you’re on a budget these sub $100 phones can help.

Aastra

9112i SIP Phone $89.99

Aastra 9112i Features

  • Enhanced Call Management — Large storage for personal directory, callers log, and redial list
  • Tight Integration — Support for multiple IP telephony systems including BroadWorks®, Nortel, Sylantro, and Asterisk SIP
  • Remarkable Audio — Quality speakerphone with excellent voice delivery
  • Protect Your Investments — Firmware upgrades can be downloaded and installed in the field as standards develop and protocols evolve

Aastra 9112i Specifications

Physical

    • 20.2 cm W x 19.2 cm D x 8.9 cm H (8.0”W x 7.6”D x 3.5”H)
    • 812 g (28.6 oz)
  • Power
    • AC wall adapter included
  • Handset / Headset
    • Modular RJ9 headset connector, compatible with amplified business headsets
    • Hearing aid compatible handset
    • Quality speakerphone
  • Display
    • 3 line backlit display
  • Feature Keys
    • 4 navigational keys
    • 2 programmable keys
    • 11 predefined hard keys including Callers log, Conference, Call Transfer, Redial, Options, Directory, Save, Delete, Speaker/Headset, Mute
  • Networking
    • 10/100 Mbps Ethernet port
    • Manual or Dynamic Host Configuration Protocol (DHCP) IP address setup
    • Time and date synchronization using SNTP
    • Built-in HTTP server for web administration and maintenance
    • Server provisioned user configuration files
  • Protocols / Codecs
    • IETF SIP (RFC3261)
    • G.711 ?-law / A-law
    • G.729
  • Feature Highlights
    • Personal directory
    • Call forward
    • Call transfer
    • Call waiting
    • Caller and calling line information
    • Callers log
    • Conference
    • Redial list
    • Live dial pad or pre-dial support

D-Link

DVG-2001S SIP Adapter MSRP $59.99

D-Link DVG-2100S Features:

  • (1) FXS Port
  • (1) RJ-45 Network Port
  • Enable VoIP Instantly
  • Secured Remote Web-Based Access For Configuration
  • Integrated QoS To Prevent Dropped Calls

Download D-Link DVG-2001S Product Datasheet The D-Link DVG-2001S VoIP Phone Adapter comes with one FXS port to connect to the existing analog telephone and one Fast Ethernet port to connect to the broadband router.

With a built-in secured provisioning feature, VoIP service providers can configure service settings such as a server address, CODEC and STUN settings via HTTPS/TFTP directly to the DVG-2001S.

DVG-2100S Specifications:

  • Standards
    • TCP
    • UDP
    • ARP
    • HTTP
  • Connection Port
    • RJ-11
    • FXS Port
    • RJ-45 Ethernet Port
  • Ethernet Port
    • IEEE 802.3 for 10M Ethernet
    • IEEE 802.3u for 100M Ethernet
  • Telephony Support
    • SIP Call Control Protocol
    • Supports Audio CODEC• G.711 (A-law and U-law)
    • G.723.1
    • G.726
    • G.729A
    • G.168 (Echo Cancellation)
    • DTMF Relay• G.711 (In Band)
    • RFC2833
  • Device Management
    • TFTP Client
    • HTTP Web Interface
  • Configuration/Management
    • DHCP (Dynamic Host Configuration Protocol) RFC2131
    • Embedded Web Server HTTP1.0 (RFC1945)
    • Auto-Provisioning Via Automated Centralized Configuration File
    • Configuration Restore/Backup
    • TELNET
    • TFTP Client
    • Performance Monitor DSP/Ethernet Statistics
  • Quality of Service (QoS)
    • TOS-Type of Service Supports 3 Levels:
    • Normal
    • Signaling
    • RTP Packets
  • Security
    • SIP Authentication with Password Encryption
    • HTTP Digest Authentication
    • Configuration Download Using HTTPS and SSL/TLS Clients Certification Encryption and Authentication
    • Encryption of Configuration File
    • VoIP NAT Traversal (SIP/STUN)
  • Fax Support
    • FAX Relay
    • PCM (G.711)
  • LEDs
    • Power ON/OFF
    • LAN Link & Activity
    • Phone ON/OFF Hook & Ringing
  • Power
    • External AC Power Adapter
    • Output: 12V AC, 1.2A
  • Temperature
    • Operating:0°C to 40°C
    • Storing: -10°C to 55°C
  • Humidity
    • 5%-95% Non-Condensing
  • Certifications
    • EMC: FCC Class B, VCCI Class B, CE Class B
    • UL/CUL
  • Dimensions
    • 90mm x 82.46mm x 31mm (WxDxH)
  • Warranty
    • 1 Year Limited Warranty

Grandstream

Budgetone GS-101 $44.99

Grandstream GS-101 Features:

  • Support SARP/RARP, ICMP, DNS, DHCP, NTP, TFTP protocols
  • Support NAT traversal via STUN & symmetric RTP
  • Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products
  • Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
  • Advanced and patent pending adaptive jitter buffer control, packet delay & loss concealment technology
  • Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.726, G.728, and wide-band G.722 (Model 102D). Dynamic negotiation of codec and voice payload length
  • Support standard voice features such as Caller ID Display or Block, Call Waiting, Hold, TransfForward, FLASH, in-band and out-of-band DTMF (RFC2833), Dial Plans, off-hook auto dial, configurable emergency dialing (e.g., 911), early dial, click-to-dial
  • Support 3-way conferencing (Model 102D), full duplex hands-fredomain acoustic echo cancellation (pending), redial, call log, volume control, voice mail with indicator, downloadable ring tone (pending)
  • Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
  • Support DIGEST authentication and encryption using MD5 and MD5-sess.
  • Provide easy configuration thru manual operation (phone keypad and Web interface) or personalized automated provisioning via central configuration file for mass deployment.
  • Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
  • NAT-friendly remote software upgrade capability (via tftp) even from behind firewalls/NATs.
  • Support for fail-over SIP server and DNS server (pending)

The Grandstream GS-101 has one Ethernet port.

Bugdetone GS-102 $49.99

Grandstream GS-102 Features:

  • Support SARP/RARP, ICMP, DNS, DHCP, NTP, TFTP protocols
  • Support NAT traversal via STUN & symmetric RTP
  • Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products
  • Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
  • Advanced and patent pending adaptive jitter buffer control, packet delay & loss concealment technology
  • Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.726, G.728. Dynamic negotiation of codec and voice payload length
  • Support standard voice features such as Caller ID Display or Block, Call Waiting, Hold, TransfForward, FLASH, in-band and out-of-band DTMF (RFC2833), Dial Plans, off-hook auto dial, configurable emergency dialing (e.g., 911), early dial, click-to-dial
  • Support 3-way conferencing (Model 102D), full duplex hands-fredomain acoustic echo cancellation (pending), redial, call log, volume control, voice mail with indicator, downloadable ring tone (pending)
  • Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
  • Support DIGEST authentication and encryption using MD5 and MD5-sess.
  • Provide easy configuration thru manual operation (phone keypad and Web interface) or personalized automated provisioning via central configuration file for mass deployment.
  • Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
  • NAT-friendly remote software upgrade capability (via tftp) even from behind firewalls/NATs.
  • Support for fail-over SIP server and DNS server (pending)

The Grandstream GS-102 has two Ethernet ports.

Budgetone GS-200 $64.95

Grandstream GS-200 Features:

  • Support SIP, TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, DNS, DHCP, NTP, TFTP protocols
  • Support NAT traversal via STUN & symmetric RTP
  • Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products
  • Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
  • Advanced and patent pending adaptive jitter buffer control, packet delay & loss concealment technology
  • Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.726, G.728, and wide-band G.722 (Model 102D). Dynamic negotiation of codec and voice payload length
  • Support standard voice features such as Caller ID Display or Block, Call Waiting, Hold, TransfForward, FLASH, in-band and out-of-band DTMF (RFC2833), Dial Plans, off-hook auto dial, configurable emergency dialing (e.g., 911), early dial, click-to-dial
  • Support 3-way conferencing (Model 102D), full duplex hands-fredomain acoustic echo cancellation (pending), redial, call log, volume control, voice mail with indicator, downloadable ring tone (pending)
  • Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
  • Support DIGEST authentication and encryption using MD5 and MD5-sess.
  • Provide easy configuration thru manual operation (phone keypad and Web interface) or personalized automated provisioning via central configuration file for mass deployment.
  • Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
  • NAT-friendly remote software upgrade capability (via tftp) even from behind firewalls/NATs.
  • Support for fail-over SIP server and DNS server (pending)

Linksys

Sipura SPA-2002 SIP Adapter $59.99

Linksys SPA2002 Features:

  • Terminating Impedance Agnostic - 8 Settings
  • Call Waiting, Cancel Call Waiting
  • Caller ID with Name / Number
  • Caller ID Blocking
  • Call Waiting Caller ID with Name / Number
  • Call Forwarding: No Answer / Busy / All
  • Do Not Disturb
  • Call Transfer
  • Three-Way Conference Calling with Local Mixing
  • Message Waiting Indication - Visual and Tone Based
  • Call Return
  • Call Back on Busy
  • Call Blocking with Toll Restriction
  • Delayed Disconnect
  • Distinctive Ringing
  • Off-Hook Warning Tone
  • Selective / Anonymous Call Rejection
  • Hot Line and Warm Line Calling
  • Speed Dialing of 8 Numbers / Addresses
  • Music On Hold

Linksys SPA901 SIP PhoneSPA 901 SIP Phone $79.95

Linksys SPA901 Telephony Features

  • One service provider line
  • Two call appearances accessed via Flash Key or Hook Flash
  • Shared line appearance**
  • Line status indicator
  • Call Hold
  • Music on Hold**
  • Call Waiting
  • Outbound CallerID Blocking
  • Call transfer - Atended and Blind
  • Three Way conferencing with local mixing
  • Multi-Party Call Conferencing via external Conference Bridge**
  • Call Pick Up - Selective and Group**
  • Call park and UnPark**
  • Call back on Busy
  • Call Blocking - Anonymous and Selective
  • Call Forwarding - Unconditional, No Answer, On Busy
  • Call Return - Redial Last Caller
  • Hot Line and Warm Line Automatic Calling
  • Call Logs (60 Entries Each) Made, Answered and Missed Calls. Accessed via HTTP Server
  • Redial Last Called Number
  • Do Not Disturb (Caller Hears Busy Line tone)
  • Block Anonymous Incoming Calls
  • URI (IP) Dialing support (Vanity NUmbers)
  • Built-in Web Server for Administration and Configuration, with User and Admin Access Levels
  • Built-In Interactive Voice Response (IVR) System to check status and change configuration
  • Date and Time w/Intelligent Daylight Savings Support
  • Call Start Time stored in Call Logs
  • Distinctive Ringing
  • 10 User-Downloadable Ring Tones - Ring Tone Generator free from www.Linksys.com
  • Speed Dial (8 entries)
  • Group Paging (Outbound Only)**
  • Intercom (Outbound Only)**
  • Set preferred CODEC, Per Call, All Calls
  • Configurable Dial/Numbering Plan Support
  • Ringer and Handset Voluem Controls
  • DNS SRV and Multiple A Records for Proxy Lookup and Proxy redundancy
  • Syslog, Debug, Report Generation, an Event Logging
  • Secure Call Encrypted Voice Communication Support
  • NAT Traversal
  • Automated Provisioning, Multiple Methods. Up to 256Bit encryption: (HTTP, HTTPS, TFTP)
  • Support Linksys Voice System Automatic Configuration
  • Optionally require Admin Password to Reset unit to factory defaults
  • **Feature requires support by call server.

    Hardware

    • Voice Mail Message Waiting Indicator Light
    • Redial Button
    • Dedicated Flash Button
    • Volume Control button cycles through Voluem Levels. Controls Ringer and Handset Volume.
    • Standard 12-Button dialing pad
    • High Quality Handset and Cradle
    • Ethernet LAN - 10Base-T RJ-45
    • 5v DC Universal (100-240v) Switching Power Adapter

    Specifications:

    • Data Networking
      • MAC Address (IEEE 802.3)
      • IPv4
      • ARP
      • DNS
      • DHCP Client
      • ICMP
      • TCP
      • UDP
      • RTP
      • RTCP
      • DiffServ
      • VLAN Tagging
      • SNTP
    • Voice Gateway
      • SIPv2
      • SIP Proxy redundancy
      • Re-Registration with Primary SIP Proxy Server
      • SIP Support in NAT Networks (including STUN)
      • SIPFrag
      • Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP
      • CODEC Name Assignment
      • G.711
      • G.726
      • G.729
      • G.723.1
      • Dynamic Payload Support
      • Adjustable Audio Frames Per Packet
      • DTMF: In-Band and Out-of-Band
      • Flexible Dial Plan Support with Inter-Digit Timers
      • IP Address / URI Dialing Support
      • Call Progress Tone Generation
      • Adaptive Jitter Buffer
      • Frame Los Concealment
      • VAD
      • Attenuation / Gain Adjustments
      • MWI and VMWI
      • Third Party Call Control

    • Zoom Telephonics

    5801 SIP Adapter $59.99

    Zoom 5801 Specifications:

    • (1) FXS Loop-start interface with RJ-11
    • (1)FXO analog interface with Teleport and RJ-11
    • Up to 5 REN (Ringer Equivalence Number), supports more than 5 typical telephones
    • Programmable ring patterns
    • Call progress tones supported; Initial dial tone, Secondary dial tone, Stuttered dial tone, Message waiting dial tone, Call forward dial tone, Pre-Ringback dial tone, Ring back tone, Call waiting tone, Call holding tone, Call disconnect tone, Call conference tone, Busy tone, reorder tone, Off hook warning
    • Power fail over
    • Auto switch to PSTN for emergency calling using 911 and other programmed 3 digit codes
    • FXS to FXO call bridging
    • Supports SIPv2

    Can be configured remotely using a TFTP or HTTP download from the service provider and updates to the firmware can be automatically delivered. Local configuration is done with a browser based interface.