Fanvil

Fanvil Pack4 - PBX Asterisk a 2 Lignes avec 4 Postes IP

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PBX Asterisk a 2 Lignes avec 4 Postes IP Pack4
IPBX Asterisk avec 2 FXO (Ligne Fixe RTC) et 4 Postes IP un poste pour le standard avec 8 BLF et 3 postes pour de direction.. Le système peux allez jusqu'a 30 Postes.

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Disponibilité : Nous consulter sur mv.ma
Evaluation :
  • PBX Asterisk a 2 Lignes avec 4 Postes IP Pack4

FICHE TECHNIQUE IPBX:

FICHE TECHNIQUE

30 Extensions Registers
30 Extensions Registered (Jusqu'à Postes)
20 Concurrent Calls (Appels simultané) 
2 FXO ports (2 Ligne RTC)
Video Calls
Multiple Language
DID(Direct Inward Dialing Number)
Support SKYPE for SIP
Call Monitor / Record
Support POE
FXO to FXS life line
Codec: G.711-Ulaw,G.711-Alaw,G.726,G.729,GSM,SPEEX
H.261,H.263,H.263+,H.264
Call Hold / Forward / Transfer / Waiting / Parking
Call Paging and Intercom
Caller ID
Music On Hold
DISA(Direct Inward System Access)
Flexible Dial Plan
Ring Group
Conference Room
Black List
Phone Book
Call Queue
BLF(Busy Lamp Field)
Built-in SIP/IAX2 server
Build-in voice mail server
Echo Cancelation/VAD
Configuration By web
Support Static/DHCP/PPPOE
VPN Client(Support N2N)
DDNS Client(Support Dyndns.org)
Support NTP(Network Time Protocol)
Conectivité

1 × RJ11 port
1 × RJ45 port
1 × Power port 
32 bit embedded RISC DSP
256M Onboard Nand Flash
64M SDRAM

Environmental requirements

Temperature: -10°C-45°C
Storage temperature:-30°C-65°C
Humidity: 10-80% no dew
Power: AC100~240V

FICHE TECHNIQUE DU TELEPHONE IP C56P:

Support SIP 2.0 (RFC3261) and correlative RFCs
Support IAX2,iax2 line callbr> SIP support 2 SIP lines. Can connect to SIP1 and SIP2 server at the same time and can incoming and outgoing via any SIP server.
Support multiple road call waiting in line
Codec:G.711A/u, G.7231 high/low, G.729a/b, G.722.1,G.726,AMR Support HD voice Echo cancellation: Support G.168, and Hands-free can support 96ms, Hand free Speaker Phone
Support Voice Gain Setting, VAD, CNG
Full duplex hands-free speakerphone
SIP support SIP domain, SIP authentication (none, basic, MD5),
DNS name of server, Peer to Peer/ IP call
DTMF Relay:support inband, SIP info, RFC2833
PoE (Power over Ethernet)
9 kind of ring type and 5 user-defined music ring (optional)
Large dot matrix LCD display and softkeys make user easier to use Support headset jack- RJ9
4 DSS Key
Soft keys programmable; function keys programmable
Multilanguage realizes localization
SIP application: support call forward/transfer (blind/attended)/ holding/waiting/ 3 way talking/ conference/ paging and intercom/ call park/call pickup/ join call/ redial and unredial/ click to dial/ secondary dialing automatically
Call control features: flexible dial map, hotline, empty calling no. reject service, black list for reject authenticated call, white list, limit call, no disturb, caller ID, CLIR (reject the anonymous call), CLIP (make a call with anonymous), Dial without register
Incoming calls /outgoing calls / missing calls. Each support 100 records.
Support Phonebook 500 records
Support SMS and Speed Dial
Support XML phonebook/browser
Support SRTP and Bf
Code synchronization via IP PBX/IMS
Support click to dial via web phone book
Voice codec setting for each SIP line Hands-free headset ringing choice
Network Features
WAN/LAN: 10/100M Ethernet ports, supports Route and Bridge mode.
Support bridge working as hub(optional) Support PPPoE for xDSL and PoE(optional)
Support VLAN(voice vlan/data vlan)
Support 802.1
Support basic NAT and NAPT
NAT transverse:support STUN client
Support DHCP client on WAN
Support DHCP server on LAN
Support main DNS and secondary DNS server.
Support DNS Relay, SNTP Client, Firewall, openVPN
Support VPN (L2TP) and DMZ
Network tools in telnet server: including ping, trace route, telnet client
SOFTWARE FEATURES
Support 2.0 (RFC3261) and correlative RFCs
Support 2 SIP lines 1 Button
Codec:G.711A/u, G.723.1 high/low,G.729 a/b, G.722.1,G.726 
Support HD voice
Soft keys programmable
Hard function key reprogrammable
SIP application:support call forward/transfer(blind/attended)/holding/waiting/3 way talking/conference/pickup/join call/redial and unredial/ Click to dial/secondary dialing automatically
Support SRTP
Support BLF
HARDWARE FEATURES
Two 10/100 BaseT Ethernet Interface
128*48 FSTN graphic LCD
Support Multi-line
Support IAX2 line key
4 DSS keys
PoE(optional)
TR-069(optional)
Maintenance and Management
Web ,telnet and keypad management
Management with different account right
Upgrade firmware through POST mode and HTTP, FTP or TFTP
Support dhcp option66 auto provisioning
Telnet remote management/upload/download setting file
Safe mode provide reliability
Support Auto Provisioning to upgrade
firmware or configuration
file(HTTPS/FTPS optional)
Support TR-069(optional) and Syslog

FICHE TECHNIQUE DU TELEPHONE IP C62:

Compatible avec:
Cisco Call Manager, Avaya Office 500, Mitel, Nortel, Alcatel Lucent, Bradsoft, Asterisk 
Features

Support SIP 2.0 (RFC3261) and correlative RFCs
Support 3 SIP lines. Can connect to 3 SIP servers at the same time.
Support multi line/HD voice
Support iax2 line key/ IAX2
Codec:G.711A/u, G.723.1 high/low, G.729a/b, G.722.1,G.726
Echo cancellation: G.168 Compliance in LEC, additional acoustic echo cancellation(AEC) can reach 96ms max filter length in hands-free mode 
Support Voice Gain Setting, VAD, CNG
Full duplex hands-free speakerphone
SIP support SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call
DTMF: Support SIP info, DTMF Relay, RFC2833
9 kind of ring type and 5 user-defined music ring 
Large dot matrix LCD display and softkeys make user easier to use
Friendly graphic menu
Support headset jack- RJ9
4 line keys defined as multi line with screen display, or used as SIP line keys
8 DSS keys
Soft keys programmable; function keys programmable
SIP application: support call forward/ transfer (blind/attended)/ holding/ waiting/ 3 way talking/ conference/pickup/ join call/ redial and unredial/ click to dial
Call control features: flexible dial map, hotline, empty calling no. reject service, black list for reject authenticated call, white list, limit call, no disturb, caller ID, CLIR (reject the anonymous call), CLIP (make a call with anonymous), Dial without register
Incoming calls /outgoing calls / missing calls. Each support 100 records.
Support Phonebook 500 records、SMS、XML、Speed dial、 SRTP、Blf
Code synchronization via IP PBX/IMS
Support EXT DSS consoles with 5max.
Support click to dial via web phone book/ Group listening
Voice codec setting for each SIP line
Support keypad lock, and imergency call during the keypad lock.
Customized lcd logo
Ring play via headset or speaker setting
Signal tone parameters customized

Network Features

WAN/LAN: 10/100M Ethernet ports, supports Route and Bridge mode. 
Support bridge working as hub(optional)
Support PPPoE for xDSL
Support PoE (optional) 
Support VLAN(voice vlan/data vlan)
Support 802.1
Support basic NAT and NAPT
NAT transverse:support STUN client
Support DHCP client on WAN
Support DHCP server on LAN
Support main DNS and secondary DNS server.
Support DNS Relay, SNTP Client, Firewall and openVPN
Support VPN (L2TP) and DMZ
QoS with DiffServ
Network tools in telnet server: including ping, trace route, telnet client

Maintenance and Management

Web ,telnet and keypad management
Management with different account right
Upgrade firmware through HTTP, FTP or TFTP
Support dhcp option66 auto provisioning
Telnet remote management/upload/ download setting file
Safe mode provide reliability
Support Auto Provisioning to upgrade firmware or configuration file(HTTPS/FTPS optional)
Support TR-069(optional)/ Syslog

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