Connect to analog PBX or the PSTN lines of telecom carriers
8 FXO ports
SIP protocol
Delivery policy
Tarifat e Transportit :
1- Tirana : Falas
2- Rrethet : 360 leke
3-Kosova : 6 Eur
Return policy
(edit with the Customer Reassurance module)
Guarantee safe & secure checkout
DAG1000-8O FXO Analog VoIP Gateway is a type of access gateway offering seamless connectivity between IP-based telephony networks and legacy telephones (POTS) and PBX systems. The analog gateway has 8 FXO ports and is used to connect to analog PBX or the PSTN lines of telecom carriers. With the standard SIP protocol, it's compatible with leading IMS/NGN platforms and SIP-based IP Phone systems. It provides low-cost and easy-to-use VoIP solutions for small and medium businesses, call centers, SOHO, remote offices as well as enterprises with multiple branches.
Data sheet
Capacity:
500 SIP users, 30 concurrent calls
Ports:
2 FXS, 2 FXO
Features:
IP/SIP Failover
Multiple SIP trunks
Fax over IP (T.38 and Pass-through)
Built-in VPN
TLS / SRTP security
Call waiting
Call transfer
Voicemail
Call queue
Ring group
Paging
Voicemail to Email
Event report
Conference Call
8 FXS ports
8 FXO ports
SIP protocol
32 SIP users
8 Concurrent Calls
Multiple SIP trunks
Mobile Extension
Voice over LTE (VoLTE)
Built-in VPN
Wi-Fi Hotspot
TLS / SRTP security
The design of Fanvil PA3 fits to every internal installation, especially for public broadcasting. With HD audio and function-rich interfaces, Fanvil PA3 can be used for real-time & fix-time MP3 broadcasting, and one-touch intercom when connecting to the external devices. Fanvil PA3 is your perfect choice to DIY the integrated broadcasting solution for the campus, shopping mall, railway station, building, etc.
8 GSM/CDMA/WCDMA Channels
Support to different mobile networks:
GSM: 850/900/1800/1900MHz;
CDMA: 800MHz; WCDMA: 850/900/1900/2100MHz
HTTP API for SMS Application Integration
SMS to Email and Email to SMS
Flexible Dial Rules and Manipulation Rules
60 SIP users, 15 concurrent calls
2 FXO
IP/SIP Failover
Multiple SIP trunks
Fax over IP (T.38 and Pass-through)
Built-in VPN
TLS / SRTP security
48 FXS Ports
Softswitches
IP PBXs
SIP trunks
4 FXS ports
4 FXO ports
SIP protocol
Capacity:
60 SIP users, 30 concurrent calls
Ports:
2 FXS, 2 FXO
Features:
IP/SIP Failover
Multiple SIP trunks
Fax over IP (T.38 and Pass-through)
Built-in VPN
TLS / SRTP security
Call waiting
Call transfer
Voicemail
Call queue
Ring group
Paging
Voicemail to Email
Event report
Conference Call
Product type: IP-Phone System
CPU: DualCore 500MHz DSP
Storage: 4GB (Default SD Card)
SDRAM: 128MB DDR2
System: Asterisk 1.8
Connect to analog PBX or the PSTN lines of telecom carriers
8 FXO ports
1 WAN
3 LAN
8 RJ11
SIP protocol
Connect to analog PBX or the PSTN lines of telecom carriers
4 FXO ports
4 FXS ports
1 WAN
3 LAN
8 RJ11
SIP protocol
Capacity:
200 SIP users, 30 concurrent calls
Ports:
2 FXS, 2 FXO
Features:
IP/SIP Failover
Multiple SIP trunks
Fax over IP (T.38 and Pass-through)
Built-in VPN
TLS / SRTP security
Call waiting
Call transfer
Voicemail
Call queue
Ring group
Paging
Voicemail to Email
Event report
Conference Call
8 GSM/CDMA/WCDMA Channels supported on UC2000-VE
Support to different mobile networks:
GSM: 850/900/1800/1900MHz;
CDMA: 800MHz; WCDMA: 850/900/1900/2100MHz
HTTP API for SMS Application Integration
SMS to Email and Email to SMS
Flexible Dial Rules and Manipulation Rules
PBX Features Up to 200 Extensions
Up to 60 Simultaneous Calls
Up to 40 Simultaneous Conference Attendees
Up to 36 Call Queues
Up to 36 Conference Rooms
Recording up to 150 hrs (WAV)
Recording up to 1500 hrs (GSM)
72 FXS Ports
Softswitches
IP PBXs
SIP trunks
