January 15 2008
Low Cost SIP Phones
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
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
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.
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.
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 Telephony Features
**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.







This company also has very good and cheap VoIP phones, Gateways and IPTV boxes http://www.konceptproducts.com/en/product.asp, most of their products are under $100. Right now they have 1FXS adapter for only $15.
Ring Tones…
I Googled for something completely different, but found your page…and have to say thanks. nice read….
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