OpenVox iAG802 Analog VoIP Gateway
Supports 8 FXS ports;
Use standard SIP protocol;
Compatible with Leading IMS/NGN platform, IPPBX and SIP servers.
Request a Quote-
Specifications
OpenVox iAG802 is a multifunctional analog gateway that provides 8 FXS ports for seamless connection between IPPBX, fax machines, analog phones and operators. In addition, it has excellent full concurrent voice/fax processing capabilities, strong performance and high stability, and provides high-quality call services for operators, enterprises, call centers and residential users in residential communities.
The iAG802 Analog Gateway, a cost-effective product of the iAG Series, is an ideal analog VoIP gateway solution for SMBs and SOHOs. With friendly GUI and unique design, users may easily customize and configure their gateway. Also secondary development can be completed through API.
The iAG802 Analog Gateway is developed for interconnecting a wide selection of codecs including G.711A/U, G.723.1, G.729A, G.722, iLBC, OPUS, AMR and AMR-WB. iAG802 series use standard SIP protocol and compatible with leading VoIP platform, IPPBX and SIP servers, such as Asterisk, Issabel, 3CX, FreeSWITCH, BroadSoft and VOS VoIP operating platform.
Technical Specifications:
Models iAG802 Ports 8 FXS ports Ethernet Ports 2 * 10/100/1000Mbps Ports Weight 400g Dimension 170*98*26mm Universal Power Supply DC 12V/2A Maximum Power 12W Operation Temperature 0°C ~ 50°C Operation Humidity Range 10% ~ 90% NON-CONDENSING Storage Temperature Range -20°C ~ 70°C Short/Long Haul Ring Load 2 REN, up to 3km on 24 AWG line Certification CE & FCC Software & Platform:
- Linux OS
-
Features
Physical Interfaces (Capacity & Ethernet Interfaces)
- 8 FXS with RJ11
- 1*WAN, 10/100/1000Mbps, RJ-45
- 1*LAN, 10/100/1000Mbps, RJ-45
Voice & FAX
- G.711A/U, G.723.1, G.729A, G.722, iLBC, OPUS, AMR and AMR-WB
- Comfort Noise Generation (CNG)
- Modem/POS
- Adaptive (Dynamic) Jitter Buffer
- Programmable Gain Control
- DTMF mode: Signal/RFC2833/INBAND
- T.38/Pass-through
- Echo Cancellation (G.168), with up to 128ms
Call Features
- Call Waiting
- Blind Transfer
- Attend Transfer
- Call Forward on Busy
- Call Forward on No Reply
- Unconditional Call Forward
- Warm/lmmediately Hotline
- Message Waiting Indicator
- SlP registration to different servers
- Call Hold
- Do-not-disturb
- 3-Way Conference
- Speed Dial
FXS
- Connector: RJ11
- Dial Mode: DTMF and Pulse
- Pulse: 10 and 20 PPS
- Caller ID: DTMF/FSK CLI Presentation
- Max Cable Length:3 km
- Reversed Polarity Programmable Call Progress Tone
Software Features
- Hunting Group
- Web ACL
- Telnet ACL
- Action URL
- PPPoE
- OpenVPN
- Digitmap
- Routing Rules based Prefixes
- Caller/Called Number Manipulation
- VEX (Advanced Peer To Peer Call)
VOIP (Protocol)
- SIP v2.0 (UDP/TCP), RFC3261
- SDP, RTP(RFC2833), RFC3262, RFC3263, RFC3264, RFC3265, RFC3515, RFC2976, RFC3311
- RTP/RTCP, RFC2198, RFC1889
- IPv4 and lPv6
- Outbound Proxy
- RFC2806 TEL URI
- RFC3581 NAT, rport
- VLAN 802.1P/802.1Q
- RFC4028 Session Timer
- DNS SRV/A Query/NATPR Query
- NAT: STUN, Static/Dynamic NAT
Maintance
- SNMP v1/v2/v3
- Auto Provisioning
- Web Configuration
- Backup/Restore
- Firmware Upgrade via Web
- CDR
- TR069
- IPoE
- Syslog(Emerg, alert, critical, error, warning, notice, info, debug)
- Ping/Tracert Test
- Network Capture
- NTP/Daylight Saving Time
- lVR local Maintenance
- Cloud-based Management
-
Download
PDF Manual
Online Manual