The Grandi GIX100 VoIP ATAs  works great. It is complies with SIP v2.0 and very easy to configure, excellent audio and great selection of codecs including G.711, G.723.1, G.729A/B. It is inexpensive with dual 10/100M Ethernet ports that can be configured as either switched or routed ports, built-in NAT router/DHCP server and support message waiting indicator.

The GIX-100 connects your standard telephones and PBX to the IP-based data networks. With the GIX-100, individuals and companies are able to protect and extend their past investments in existing equipments as well as control their migration to IP with an extremely affordable, incremental investment.

Grandi adopts advanced single ARM9 CPU technology in GIX100 design. So, it can be delivered with high performance at competitive price points thus helping service providers realize rapid payback on CPE investment.

GIP100 is just at $28.00 USD for 100K units order.

GIX100 Specification

VoIP SoC    CPU: MARVELL 88E6218@150MHZ

Flash           2MBytes Flash

RAM             8MBytes SDRAM   

Power Supply 
                      AC/DC Adapter   AC Input: 100~240V, 50~60Hz
                      DC Output: 5V, 1.2A

I/O Interfaces 
                      2 RJ-45 for 10/100 Ethernet Ports that can be configured as either switched or routed ports
                      1 RJ-11 for FXS port

Audio Codec
                     G.711 (A-Law, ?Law)
                     G.723.1
                     G.729A/AB
                     Adaptive Jitter Buffer Management
                     Voice Activity Detection
                     Comfort Noise Generation
                     Echo Cancellation

Protocols 
                    SIP V2 (RFC 3261,3262,3263,3264)
                    Backward Compatible with RFC2543
                    Session Timer (RFC4028)
                    SDP (RFC2327)
                    RTP/RTCP (RFC1889 and RFC1890)
                    NAPTR for SIP URI Lookup (RFC2915)
                    STUN (RFC 3489)
                    ARP/RARP (RFC 826/903)
                    SNTP (RFC 2030)
                    DHCP/PPPoE
                    HTTP Server for Web Management
                    TFTP/HTTP for Auto Provisioning
                    Message Waiting Indicator (RFC3842)
                    DHCP Option Codes for SIP (RFC3361)
                    DNS/DNS SRV (RFC1706 and RFC 2782)
                    IEEE802.1Q VLAN/802.1p and IP TOS

Applications
 Mac address cloning
                     SIP proxy redundancy
                                   - dynamic via DNS SRV, A records
                     Direct IP to IP calling
                     NAT Traversal
                                   -Static NAT Route
                                   - Traversal by STUN
                     Build-in NAT Router/DHCP server

Management
 Firmware Upgradeable
                     Web Management Interface
                     IVR-driven Management Interface
                     Auto Provisioning

Call Features
                     3-way Conference
                     Music on hold
                     Call Hold
                     Call Forwarding
                     Call Mute
                     Call Park
                     Call Progress Tone
                     Call Transfer
                     Call Waiting
                     Caller ID
                     DTMF Relay
                           - In-band, Out-band(RFC2833) and SIP Info
                     Hotline

Dimension
          (L) 110 x (W) 75 x (H) 30 mm

Weight
          366g