PRODUCT SPECIFICATIONS
The DAG2500 Series FXS analog gateway is multi-functional voice gateway that provides seamless connectivity with analog phones, fax machine or analog PBXs to VoIP gateway. The overall frame of DAG2500 is compact and structured, supporting RJ45/RJ21 interface standards and up to 72 FXS voice interfaces at a size of 1U. In addition, DAG2500 provides high-performance hardware solution with good voice/fax processing capability. It is able to provide efficient and high-quality IP voice services for service provider, enterprises, community residents, and corporate telephone systems.
The DAG2500 supports standard SIP and RTP protocols and is compatible with multiple well-known platforms, for example, compatible with BroadSoft. It also works as extensions with various IPPBX products based on Asterisk and FreeSWITCH, and is also perfectly compatible with IMS/NGN softswitch platforms and call center systems. In addition, customers can build an enterprise branch communication network by using the DAG2500 to achieve interoperability between branches, which can provide an efficient, stable, and cost-effective voice communication solution for large enterprises.
Key Features
- High density gateway, 72 FXS ports in 1U size
- Support IPv4 and IPv6
- 5 km Line Length(MAX)
- Support SNMP and TR069
- Multiple codecs: G.711A/U,G.723.1,G.729A/B and iLBC
- Support Asterisk and FreeSWITCH platforms, and compatible with BroadSoft
- Fully compatible with leading IMS/NGN, SIP based IP telephony system
Dimensions
- 440 x 328 x 44mm (1U)
Weight
- 4.25kg
Technical Description
- Physical Interfaces
- Phone Connector
- DAG2500-72S: 72 FXS, RJ 45 or RJ21, 50 PIN
- Network Interfaces
- 2x 10/1000 Base-T
- 1x 10/100 Base-T
- Console:
- 1x RS232, 115200bps
- Call Features
- Digit map
- IMPU
- Action URL
- Voicemail
- Custom IVR
- Call Waiting
- Call Holding
- Do-not-disturb
- Blind Transfer
- Attend Transfer
- Call Forward on Busy
- Call Forward on No Reply
- Ringing Group, Hotline
- 3-way Conference
- Early Media/Early Answer
- Abbreviated Dialing
- Access and Peer Mode
- SIP PnP Auto Provisioning
- PBX IVR
- Unconditional Call Forward
- SIP registration to different servers
- Flexible Call Routing Policy
- Caller/Called Number Manipulation
- FXS
- Connector: RJ45 or RJ21
- Max Line Length: 5 km
- Reversed Polarity
- Pulse: 10 and 20 PPS
- Dial Mode: DTMF and Pulse
- Programmable Call Progress Tone
- Caller ID: DTMF/FSK CLI Presentation
- VoIP Protocol:
- SIP v2.0 (UDP/TCP), RFC3261
- SDP,RTP(RFC2833), RFC3262, RFC3263, RFC3264, RFC3265, RFC3515, RFC2976, RFC3311
- RTP/RTCP, RFC2198, RFC1889
- IPv4 and IPv6
- Outbound Proxy
- Master/Slave Server
- 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
- Maintenance
- CDR
- Ping/Tracert Test
- Network Capture
- Outward Test(GR909)
- SNMP, TR069, Provision
- NTP/Daylight Saving Time
- IVR local Maintenance
- RADUIS Authentication Management
- Cloud-based Management (NMS)
- Syslog: Debug, Info, Error, Warning, Notice
- Configuration Backup/Restore
- Firmware Upgrade via HTTP/HTTPS/TFTP/FTP/HTTP
- Voice & FAX
- Recording
- T.38/Pass-through
- Modem/POS
- Silence Suppression
- Bandwidth Optimization
- Comfort Noise Generation(CNG)
- Voice Activity Detection(VAD)
- Packet Loss Concealment
- Adaptive (Dynamic) Jitter Buffer
- Programmable Gain Control
- G.711A/U law, G.723.1, G.729A/B, G.726, iLBC, AMR, AMR-GSM
- DTMF mode: Signal/RFC2833/INBAND
- Echo Cancellation(G.168), with up to 128ms
- Security
- QoS
- TLS, SRTP
- Web ACL/SSH ACL
- RADIUS Authentication
- Certification Management
- Service Network Port: Isolated/BOND/Bridge Mode
- Signaling/Voice Encryption
- Phone Connector
- Physical Features
- Power Supply:
- 100-240VAC, 50/60Hz
- Dual power supplies (only 48S/72S support)
- Power Consumption: 95W
- Temperature
- Operating Temperature: 0 °C ~ 45 °C
- Storage Temperature: -20 °C ~ 80 °C
- Humidity: 10% - 90% Non-Condensing
- Compliance: CE, FCC
- Power Supply: