Jin310 IAX/SIP IP Phone

Jin310 IAX/SIP phone-H is a kind of voice communication terminal, which is based on wide band, and is independent of PC (personal computer).

Voice Features

*Support IAX2 and SIP RFC3261 synchronously

*Codec: G.711A/u, G.7231 high/low, G.729

*Echo cancellation: Support G.168, and Hands-free can support 96ms

*Support Voice Gain Setting, Jitter Buffer, VAD, CNG, dual GK, call forward, and Peer to Peer

*NAT transverse: support STUN client, Citron, AVS

*SIP support SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to peer

*SIP support Pubic & Private server. Can connect to ISP and Private SIP server at the same time

*SIP support dual public server

*DTMF: Support SIP info, DTMF Relay, RFC2833

*SIP application: support Call forward/transfer/holding/waiting

*Call control features: Flexible dial map, support Hotline, Empty calling No. Reject service,

Black list for reject authenticated call, No disturb, Caller ID.

*Support conference call and voice record

*English, Spanish, Czechoslovak alternative

Details:https://telecomc.ipower.com/osCommerce/catalog/product_info.php/products_id/28?osCsid=b5a2a2512d6a6b7f36e5844d71596301

Contact us:
Web: http://telecomchinasourcing.com
Email: grace@telecomchinasourcing.com
MSN; sale2@telecomchinasourcing.com
Skype: telecomchina
Tel: +86-1082901131

[quote=“telecomchina”]Jin310 IAX/SIP phone-H is a kind of voice communication terminal, which is based on wide band, and is independent of PC (personal computer).

Voice Features

*Support IAX2 and SIP RFC3261 synchronously

*Codec: G.711A/u, G.7231 high/low, G.729

*Echo cancellation: Support G.168, and Hands-free can support 96ms

*Support Voice Gain Setting, Jitter Buffer, VAD, CNG, dual GK, call forward, and Peer to Peer

*NAT transverse: support STUN client, Citron, AVS

*SIP support SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to peer

*SIP support Pubic & Private server. Can connect to ISP and Private SIP server at the same time

*SIP support dual public server

*DTMF: Support SIP info, DTMF Relay, RFC2833

*SIP application: support Call forward/transfer/holding/waiting

*Call control features: Flexible dial map, support Hotline, Empty calling No. Reject service,

Black list for reject authenticated call, No disturb, Caller ID.

*Support conference call and voice record

*English, Spanish, Czechoslovak alternative

Details:https://telecomc.ipower.com/osCommerce/catalog/product_info.php/products_id/28?osCsid=b5a2a2512d6a6b7f36e5844d71596301

Contact us:
Web: http://telecomchinasourcing.com
Email: grace@telecomchinasourcing.com
MSN; sale2@telecomchinasourcing.com
Skype: telecomchina
Tel: +86-1082901131
[/quote]

is the software supported Linux OS???