Kexint Optional Call Center Device 48 64 72 Port Support Rj21 Connector FXS Analog VoIP Gateway

• Customization: Available
• Type: FXS Analog VoIP Gateway
• Certification: CE, ISO, RoHS
• Model NO.: KXT2500-48/64/72S
• Condition: New
• Voice Codec: G.711A/μ Law, G.723.1, G.729A/B, G.726
• Protocol: Pptp/L2tp VPN/Openvpn .etc
• Support: SIP Protocol
• Power Supply: 100-240VAC
• Transport Package: Package
• Specification: 46*30*6
• Trademark: KEXINT
• Origin: Guangdong, China

Product Description

Product Description

KXT2500 is a functional analog gateway offering seamless connectivity between VoIP and traditional phone system. It's a highly-integrated gateway which provides 48/64/72 FXS ports with standard R21 interface in 1U size. Users can easily connect their analog phone, fax, PBX via standard voice interface.KXT2500 supports the standard SIP protocol and it's compatible with leading IMS/NGN platform and SIP-based IP telephony system. It is ideally suited for small and medium businesses, call centers and multi-location environments that need VoIP services.
 

Feature

High density gateway, 48/64/72 FXS ports in 1U size
Open-standard SIP support
Support IPv4 and IPv6
3KM Maximum Cabling Length
Elastix/Broadsoft Certification
Multiple codecs: G.711A/U,G.723.1,G.729A/B, iLBC,AMR
Fully compatible with leading IMS/NGN, SIP based IP telephony system
 

Specification

Physical Interfaces
Phone Connector:
DAG2500-72S: 72 FXS with 3* RJ21, 50 PIN
DAG2500-64S: 64 FXS with 3* RJ21, 50 PIN
DAG2500-48S: 48 FXS with 2* RJ21, 50 PIN
Console:1* RS232, 115200bps
SFP:N/A, Reserve
Network Interface: 2* 10/100/1000 Base-T
 

Voice & FAX
G.711A/U law, G.723.1, G.729A/B,G.726,iLBC,AMR
Silence Suppression
Comfort Noise Generation(CNG)
Voice Activity Detection(VAD)
Echo Cancellation(G.168), with up to 128ms
Adaptive (Dynamic) Jitter Buffer
Programmable Gain Control
T.38/Pass-through
Modem/POS
DTMF mode: SIP Info/RFC2833/In-band
VLAN 802.1P/802.1Q
Layer3 QoS and DiffServ
 

FXS Features
Connector: RJ21
Dial Mode: DTMF and Pulse
Pulse: 10 and 20 PPS
Caller ID: DTMF/FSK CLI Presentation
Max Cable Length: 5KM
Reversed Polarity
Programmable Call Progress Tone
 

VoIP
Protocol: TLS / SRTP SIP v2.0 (UDP/TCP),RFC3261,SDP,RTP(RFC2833), RFC3262, 3263,3264,3265,3515,2976,3311
RTP/RTCP, RFC2198, 1889
RFC4028 Session Timer
RFC3266 IPv6 in SDP
RFC2806 TEL URI
RFC3581 NAT,rport
Primary/Backup SIP Server
Outbound Proxy
DNS SRV/ A Query/NATPR Query
Early Media/Early Answer
NAT:STUN, Static/Dynamic NAT
 

Supplementary Service
Call Waiting,Blind Transfer,Attend Transfer,Call Pickup,Call Forward on Busy,Call Forward on No Reply,Unconditional Call
Forward,Warm/Immediately Hotline,Call Hold ,Do-not-disturb,3-Way Conference,Message Waiting Indicator(MWI)
 

Software Features
•Hunting Group
•Web/Telnet ACL
•Action URL
•PPPoE/IPv4/IPv6
•Digitmap
•Bandwidth Optimization
•Routing Rules based Prefixes
•Caller/Called Number Manipulation
 

Maintenance
• SNMP v1/v2/v3, TR069
• Auto Provisioning
• Web/Telnet
• Configuration Backup/Restore
• Firmware Upgrade via Web
• CDR, Syslog
• Ping/Tracert Test
• Network Capture
• Outward Test(GR909)
• NTP/Daylight Saving Time
• IVR local Maintenance
• Cloud-based Management

Power Supply	100-240VAC, 50-60 Hz
Power Consumption	75W
Operating Temperature	0 ºC ~ 45 ºC
Storage Temperature	-20 ºC ~80 ºC
Humidity	10%-90% Non-Condensing
Dimensions(W/D/H)	440*280*44mm(1U)
Unit Weight	4.0kg
Compliance	CE, FCC

Our Company