<?xml version="1.0" encoding="utf-8"?>
<rss version="2.0">
<channel>
<title>Voip Telephone</title>
<link>http://www.blunet.net.cn/networking/voip-telephone/</link>
<description>Products /  / Networking  /  / Voip Telephone</description>
<language>zh-cn</language>
<generator>DSW Fire  safety &amp; Security </generator>
<webmaster>sales@dswbrand.com</webmaster>
<item>
    <title>VP-2005</title>
    <link>http://www.blunet.net.cn/networking/voip-telephone/voip-telephone-1924.html</link>
    <description>Support Three sip servers running at  the same time.
Redundancy sip server capable.
NAT,Firewall.
Auto Call Routing Selection VOIP and  PSTN(choice)
DHCP client and server.
Support PPPoE,(used for ADSL,cable  modem connecting).
Support major G7.xxx CODEC.
G.165 compliant 16ms echo  cancellation
E.164 dial plan and customized dial  rules
Hotline.
Call Forward,Call Transfer,3-way  conference calls
Call ID display
DND(Do Not Disturb),Black List,Limit List</description>
    <pubDate>2008-06-17</pubDate>
    <category>Voip Telephone</category>
    <author>DavidJon</author>
    <comments>DSW safety &amp; Security </comments>
</item>
<item>
    <title>VP-2004</title>
    <link>http://www.blunet.net.cn/networking/voip-telephone/voip-telephone-1923.html</link>
    <description>Support Three sip servers running at the same time.
Redundancy sip server capable,NAT,Firewall.
Auto Call Routing Selection VOIP and PSTN(choice)
DHCP client and server.
Support PPPoE,(used for ADSL,cable modem connecting).
Support major G7.xxx CODEC.
G.165 compliant 16ms echo cancellation
E.164 dial plan and customized dial rules
Hotline.
Call Forward,Call Transfer,3-way conference calls
Call ID display DND(Do Not Disturb),Black List,Limit List
CPU:32Bit 125Mhz Flash:16m Ram:16m FCC/CE
Size:200 x 170 x 65 mm (L x W x H)</description>
    <pubDate>2008-06-17</pubDate>
    <category>Voip Telephone</category>
    <author>DavidJon</author>
    <comments>DSW safety &amp; Security </comments>
</item>
<item>
    <title>VP-2003</title>
    <link>http://www.blunet.net.cn/networking/voip-telephone/voip-telephone-1922.html</link>
    <description>Support Three sip servers running at the same time.
Redundancy sip server capable,NAT,Firewall.
Auto Call Routing Selection VOIP and PSTN
DHCP client and server.
Support PPPoE,(used for ADSL,cable modem connecting).
Support major G7.xxx CODEC.
G.165 compliant 16ms echo cancellation
E.164 dial plan and customized dial rules
Hotline.
Call Forward,Call Transfer,3-way conference calls
Call ID display
DND(Do Not Disturb),Black List,Limit List
CPU:32Bit 125Mhz Flash:16m Ram:16m FCC/CE
Size:200 x 170 x 65 mm (L x W x H)</description>
    <pubDate>2008-06-17</pubDate>
    <category>Voip Telephone</category>
    <author>DavidJon</author>
    <comments>DSW safety &amp; Security </comments>
</item>
<item>
    <title>VP-2002</title>
    <link>http://www.blunet.net.cn/networking/voip-telephone/voip-telephone-1921.html</link>
    <description>Support Three sip servers running at the same time.
Redundancy sip server capable,NAT,Firewall.
Auto Call Routing Selection VOIP and PSTN(choice)
DHCP client and server.
Support PPPoE,(used for ADSL,cable modem connecting).
Support major G7.xxx CODEC.
G.165 compliant 16ms echo cancellation
E.164 dial plan and customized dial rules
Hotline.
Call Forward,Call Transfer,3-way conference calls
Call ID display
DND (Do Not Disturb),Black List,Limit List
CPU 32Bit 125Mhz Flash:16m Ram:16m FCC/CE
Size:220 x 170 x 65 mm (L x W x H)</description>
    <pubDate>2008-06-17</pubDate>
    <category>Voip Telephone</category>
    <author>DavidJon</author>
    <comments>DSW safety &amp; Security </comments>
</item>
<item>
    <title>VP-2001</title>
    <link>http://www.blunet.net.cn/networking/voip-telephone/voip-telephone-1920.html</link>
    <description>Support SIP REC2543,H.323V4,MGCP RFC2705and dynamic configures IP address via DHCP and static IP configuration. 
Set phone via standard web browser (e.g. IE) and standard TENET method. 
Upgrade the program on line via FTP method. 
Support main G.723.1 (5.3K/6.3K),G.729,G.711A-Law,and u-Lawand gsm610 voice parser code algorithm. 
Set phone via web browser,standard TELNET program and specific tool ?V IPPHONETOOL. 
Connect with phone in the PSTN via gateway. 
Dynamic voice0-detecting,nice voice-generating,dynamic voice cache technology. 
Support multinnetwork construction. 
Support full duplex hand-free,2*16LCD 
Adjust the volume of the conversation and speaker respectively. 
Record the missed call received call and dialed call (80 groups respectively). 
Match the G.165 16ms echo counteracting,offer the nice timbre as the additional phone. 
Match the ITU-T standard phone voice and DTMF-generating and detecting. 
Store 100 groups' phonebook numbers. 
Voice indication function</description>
    <pubDate>2008-06-17</pubDate>
    <category>Voip Telephone</category>
    <author>DavidJon</author>
    <comments>DSW safety &amp; Security </comments>
</item>
<item>
    <title>VP-850</title>
    <link>http://www.blunet.net.cn/networking/voip-telephone/voip-telephone-1919.html</link>
    <description>Support 5 SIP lines
Two 10/100M Ethernet Interfaces
Suport registering two SIP accounts from two SIP Servers
Codec:G.711 / G723/G.729a
Support both router and bridge mode
Support PPPoE for XDSL
Support Voice Mail,Call Waiting/Transfer/Holding/Conference.
Support Voice Gain Setting,Jitter Buffer,VAD and CNG.
Web configuration through Built-in web server.
Upgrading firmware and configurations by HTTP,FTP or TFTP.
DHCP Client on WAN and DHCP Server on LAN.</description>
    <pubDate>2008-06-17</pubDate>
    <category>Voip Telephone</category>
    <author>DavidJon</author>
    <comments>DSW safety &amp; Security </comments>
</item>
<item>
    <title>VP-840</title>
    <link>http://www.blunet.net.cn/networking/voip-telephone/voip-telephone-1918.html</link>
    <description>Support 4 SIP lines
PoE (Power over Ethernet);very good handfree voice quality
Two 10/100M Ethernet Interfaces
Suport registering two SIP accounts from two SIP Servers
Codec;G.711 / G723/G.729a
Support both router and bridge mode
Support PPPoE for XDSL
Support Voice Mail,Call Waiting/Transfer/Holding/Conference.
Support Voice Gain Setting,Jitter Buffer,VAD and CNG.
Web configuration through Built-in web server.
Upgrading firmware and configurations by HTTP,FTP or TFTP.
DHCP Client on WAN and DHCP Server on LAN.</description>
    <pubDate>2008-06-17</pubDate>
    <category>Voip Telephone</category>
    <author>DavidJon</author>
    <comments>DSW safety &amp; Security </comments>
</item>
<item>
    <title>VP-820</title>
    <link>http://www.blunet.net.cn/networking/voip-telephone/voip-telephone-1917.html</link>
    <description>Support SIP 2.0 (RFC3261)and correlative RFCs
Supprt IAX2
Codec:G.711A/u,G.7231 high/low,G.729,G.722
Echo cancellation:Support G.168,and Hands-free can support 96ms,Hand freeSpeaker Phone
Support Voice Gain Setting,VAD,CNG
Full duplex hands-free speakerphone
NAT transverse:support STUN client
SIP support SIP domain,SIP authentication (none,basic,MD5),DNS name ofserver,Peer to Peer/ IP callSIP support Pubic &amp;amp;Private server. Can connect to Public SIP and Private SIP server at the sametime
DTMF:Support SIP info,DTMF Relay,RFC2833
SIP application:support Call forward/transfer/holding/waiting
Call control features:Flexible dial map,Hotline,Empty calling reject,Blacklist for reject authenticated call,limit call,No disturb,Caller ID.
Support three way conference call
Support voice record,240 seconds max or 3 record max,user-defined recordprompt with 1 minute max 
Support Phonebook 500 records
Incoming calls/Outgoing calls/Missing calls. Each support 100 records.
Support con</description>
    <pubDate>2008-06-17</pubDate>
    <category>Voip Telephone</category>
    <author>DavidJon</author>
    <comments>DSW safety &amp; Security </comments>
</item>
<item>
    <title>VP-810</title>
    <link>http://www.blunet.net.cn/networking/voip-telephone/voip-telephone-1916.html</link>
    <description>Support SIP 2.0 (RFC3261)and correlative RFCs
Supprt IAX2
Codec:G.711A/u,G.7231 high/low,G.729,G.722
Echo cancellation:Support G.168,and Hands-free can support 96ms,Hand freeSpeaker Phone
Support Voice Gain Setting,VAD,CNG
Full duplex hands-free speakerphone
NAT transverse:support STUN client
SIP support SIP domain,SIP authentication (none,basic,MD5),DNS name ofserver,Peer to Peer/ IP call
SIP support Pubic &amp;amp;Private server. Can connect to Public SIP and PrivateSIP server at the same time
DTMF:Support SIP info,DTMF Relay,RFC2833SIP application:supportCall forward/transfer/holding/waiting
Call control features:Flexible dial map,Hotline,Empty calling reject,Blacklist for reject authenticated call,limit call,No disturb,Caller ID.
Support three way conference call
Support voice record,240 seconds max or 3 record max,user-defined recordprompt with 1 minute max 
Support Phonebook 500 records
Incoming calls/Outgoing calls/Missing calls. Each support 100 records.
Support conf</description>
    <pubDate>2008-06-17</pubDate>
    <category>Voip Telephone</category>
    <author>DavidJon</author>
    <comments>DSW safety &amp; Security </comments>
</item>
<item>
    <title>VP-900</title>
    <link>http://www.blunet.net.cn/networking/voip-telephone/voip-telephone-1915.html</link>
    <description>Solution:Infineon,support SIP/IAX2 protocols
Support auto provision by WAN Port MAC address
Two 10/100M Ethernet Interfaces
Suport registering two SIP Servers and one IAX Server simultaneously
Codec:G.711 / G723/G.729a
Support both router and bridge mode
Support PPPoE for XDSL
Support Voice Mail,Call Waiting/Transfer/Holding/Conference.
Support Voice Gain Setting,Jitter Buffer,VAD and CNG.
Web configuration through Built-in web server.
Upgrading firmware and configurations by HTTP,FTP or TFTP.
DHCP Client on WAN and DHCP Server on LAN.</description>
    <pubDate>2008-06-17</pubDate>
    <category>Voip Telephone</category>
    <author>DavidJon</author>
    <comments>DSW safety &amp; Security </comments>
</item>

</channel>
</rss>
