タイトルまたは説明

1）SIP support 2 SIP lines. Can connect to SIP1 and SIP2 server at the same time
2）SIP application: support Call forward/ transfer/ holding/ waiting / 3 way talking
3）Call control features: Flexible dial map, Hotline, Empty calling reject, Black list for reject authenticated call, limit call, No disturb, Caller ID
Support Phonebook **0 records
4）Incoming calls / Outgoing calls / Missing calls. Each support **0 records
5）Support conference and voice record on SIP server
6)Support *0 kind of ring type and 1 user ring
7)Support IAX2，
8)Support MWI，SMS，Speed Dial, Memo, Multilanguage realizes localization
9)Support SIP 2.0 (RFC***1) and correlative RFCs
*0)Codec:G.**1 A/U Law, G.**3.1, G.**9a/b, G.**2,G.**2.1, G.**6
*1)Echo cancellation: Support G.**8, and Hands-free can support *6ms, Hand free Speaker Phone
*2)Support Voice Gain Setting, VAD, CNG
*3)Full duplex hands-free speakerphone
*4)NAT transverse:support STUN client
*5)DTMF:Support SIP info, DTMF Relay, RFC***3
*6)Support DMZ,VPN (L2TP)