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

SPA 901 SIP Phone $79.95

Linksys SPA901 Telephony Features

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.