|
|
Telecoms / Business / SIP Business Phones
|
SIPTRONIC ST-500 SIP/IAX Phone
The ST-500 is a powerful yet affordable SIP/IAX2 IP phone. Superb voice quality along with the ability to connect two SIP servers simultaneously, built in firewall and many other features including built in answer phone service and supports side tone.
- Supports a maximum of two SIP servers running at the same time.
- Redundant SIP service capable - can automatically switch to secondary service if the primary one fails.
- Answer phone - Can record three messages each at 90 seconds in length
- NAT, Firewall, WAN and LAN ports can operate as two independent networks, or in bridge mode (both network cards on the same LAN), DHCP client and server, Supports PPPoE, (used for ADSL, cable modem connecting).
- Support major G7.xxx CODEC, VAD, CNG, G.165 compliant 16ms echo cancellation, E.164 dial plan and customised dial rules, Hotline (if configured will auto dial a predetermined number when off hook state is determined), Do not disturb, Auto answer, Ban all outgoing calls, Call Forward.
- Customisable message on LCD.
- 8 selectable ringtones.
- Attended and Unattended call transfer - both run natively when using SIP protocol. For IAX, the phone uses Asterisk attended transfer and native (or Asterisk) unattended transfer. Three-way conference calls, Call ID display, DND (Do Not Disturb), Black List (Block specific incoming numbers from calling), Limit List (Block specific outgoing numbers from being dialed), Mute, Handsfree, Speed dial (9 keys on phone), VPN using L2TP and UDP, Sidetone support, SMTP time server support.
- Static/Dynamic WAN-IP-Addressing, PPPoE.
- Two RJ45 ports, one for WAN, one LAN (bridge mode available to ensure both NICs can be on the same LAN), Power port
You can buy now the ST-500 for €60.
SIPTRONIC ST-530 SIP/IAX Phone
 The ST-530 is a powerful yet affordableSIP/IAX2 IP phone. Superb voice quality along with the ability to connect two SIP servers simultaneously, built in firewall and many other features including built in answer phone service and supports side tone.
- Supports a maximum of two SIP servers running at the same time.
- Redundant SIP service capable - can automatically switch to secondary service if the primary one fails.
- Answer phone - Can record three messages each at 90 seconds in length
- NAT, Firewall, WAN and LAN ports can operate as two independent networks, or in bridge mode (both network cards on the same LAN), DHCP client and server, Supports PPPoE, (used for ADSL, cable modem connecting).
- Support major G7.xxx CODEC,VAD,CNG, G.165 compliant 16ms echo cancellation, E.164 dial plan and customised dial rules, Hotline (if configured will auto dial a predetermined number when off hook state is determined), Do not disturb, Auto answer, Ban all outgoing calls, Call Forward.
- Customisable message on LCD.
- 8 selectable ringtones.
- Attended and Unattended call transfer - both run natively when using SIP protocol. For IAX, the phone uses Asterisk attended transfer and native (or Asterisk) unattended transfer. Three-way conference calls, Call ID display, DND (Do Not Disturb), Black List (Block specific incoming numbers from calling), Limit List (Block specific outgoing numbers from being dialed), Mute, Handsfree, Speed dial (9 keys on phone), VPN using L2TP and UDP, Sidetone support, SMTP time server support.
- Static/Dynamic WAN-IP-Addressing, PPPoE.
- Two RJ45 ports, one for WAN, one LAN (bridge mode available to ensure both NICs can be on the same LAN), Power port
You can buy now the ST-530 for €70.
Grandstream GXP2000 SIP IP Enterprise Phone
Grandstream GXP-2000 is a next generation enterprise IP telephone based on open industry standards. Built on innovative technologies, GXP-2000 features market leading superb audio quality, rich functionalities, and excellent manageability at affordable prices.
- Support SIP (RFC3261), TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP, DNS (A record and SRV), DHCP (both client and server), PPPoE, TFTP, NTP
- Support NAT traversal (STUN, etc), server fail-over, SIP presence (SIMPLE), and more
- Multiline support of up to 11 lines indicators (expandable to a few dozen more through expansion key-module)
- Graphical LCD to display up to 8 lines and 22 characters per line
- Dual 10/100Mbps Ethernet ports
- Headset jack
- Support Caller ID display or block, per call or permanent
- Call waiting, Hold, Mute, Transfer (blind or attended), Forward, and more
- Multi-party conferencing
- Integrated Power-over-Ether (802.3af)
You can buy now the GXP2000 for €102.
|
Technical Characteristics
|
ST-500 SIP/AIX
|
ST-530 SIP/AIX
|
GXP2000 SIP
|
|
SIP Servers (Lines)
|
2
|
2
|
2 |
|
Firewall
|
x
|
x
|
x |
|
Answer Phone
|
x
|
x |
x |
|
Nat
|
x |
x |
x |
|
Wan
|
x |
x |
x |
|
Lan
|
x |
x |
x |
|
DHCP Client And Server
|
x |
x |
x |
|
G7.Xxx CODEC, VAD, CNG
|
x |
x |
x |
|
G.165 Compliant
(Echo Cancellation)
|
x |
x |
x |
|
Dial Plan E.164
|
x |
x |
x |
|
Customised Dial Rules
|
x |
x |
|
|
Hotline
|
x |
x |
x |
|
Do Not Disturb (DND)
|
x |
x |
x |
|
Call Waiting
|
|
|
x |
|
Auto Answer
|
x |
x |
x |
|
Ban All Outgoing Calls
|
x |
x |
|
|
Call Forward
|
x |
x |
x |
|
Customisable Message On LCD
|
x |
x |
8 lines - 22 char. |
|
Ring Tone
|
8 |
8 |
x |
|
Attended And Unattended
Call Transfer
|
x |
x |
x |
|
Conference Calls
|
3 |
3 |
Multi |
|
Caller ID
|
x |
x |
x |
|
Black List, Limit List
|
x |
x |
|
|
Mute
|
x |
x |
x |
|
Handsfree
|
x |
x |
x |
|
Speed Dial
|
x |
x |
x |
|
Headset Jack
|
|
|
x |
|
VPN Using L2TP And UDP
|
x |
x |
|
|
Sidetone
|
x |
x |
x |
|
Supports SMTP Time Server
|
x |
x |
x |
|
|
x |
x |
x |
|
PPPoE
|
x |
x |
x |
|
Rj45
|
2 |
2 |
2 |
|
Power Over Ethernet
|
|
x |
x |
For more information, contact the Sales Teams in
Greece at +30 210 6441354 ext. 313 or send your enquiry via e-mail to sales@first.gr
Serbia at +381 11 241 5866 ext. 311 or send your enquiry via e-mail to prodaja@first.gr
|
|
|
