Voip ip phone cisco

Multi-user (4 SIP account

Auto Provisioning Server General compatible TFTP, FTP , HTTP & HTTPs Server Software. Voice Codec G.711a/u (64k bps) G.729A/B (8k bps) G.723.1 high/low G.726 32 G.722 Advance Voice Quality Feature Silence Suppression Acoustic Echo Cancellation (G.167) Voice Active Detection (VAD) Comfort Noise Generation Jitter Buffer DTMF Transmitter (SIP info, Transparent, RFC 2833) Packet Lost Concealment (PLC) HD Voice handset Signal, Media & Network Protocols SIP RFC 3261 & the related RFC standard in Appendix A SDP RFC 2327 RTP RFC 1889 IP assignment: Static IP, DHCP and PPPo.

DPH-215SE compact IP Phon

SPECIFICATIONS: Audio Features Voice: handset, speaker, headset Smart Noise Filtering Wideband codec: Opus, G.722 Narrowband codec : G.711(A/µ), G.729, G.729A, i. LBC DTMF : RFC 2833,SIP INFO, Inband Full duplex hands free speakerphone with AEC VAD, CNG, AEC, PLC, AJB, AGC Phone Feature 4 SIP accounts Call hold, DND, Speed dial Call forward, call waiting, call transfer Redial, call return, auto answer Local 5 way conferencing Direct IP call, Time/Date Management Directory Local phonebook up to 1000 entries Blacklist XML/LDAP remote phonebook Smart dialing Phonebook search/import/export Call history : dialed/received/missed Display & Indicator 2.4” 320*240 –pixel color graphical LCD display LED for call and message waiting indication illuminated LED for line status information Intuitive user interface with icons and soft keys Multilingual user interface Caller ID with name and number Function Keys 4 line keys， 5 navigation keys 6 features keys: message, headset, redial, mute,hands free speakerphone Volume control keys Network Security SIP v1(RF2543), v2(RFC3261) NAT traversal: STUN Proxy and peer to peer SIP IP assignment: static/DHCP HTTP/HTTPS web server SNTP UDP/TCP/DNS SRV (RFC 3263) SRTP/ SIP TLS HTTPS certificate manager IPv6 DHCP VLAN/LLDP Management Configure via browser/phone/auto provision Auto provision via FTP/TFTP/HTTP/HTTPS TR 069, System log Reset to factory, reboot IP PBX Features BLF, Anonymous call Anonymous call rejection Call park, call pickup Other Physical Features 2 * RJ 45 10/100Mbps 1 * RJ9 (4P4C) Handset 1 * RJ9 (4P4C) Headset Wall Mountable Power Adapter(Optional) 100 240VAC Input, DC 5V/1A Output Power Consumption :2~3W Dimension(W x D x H x T): 190mm x 153mm x 162mm x 42mm Operating humidity: 10~95% Operating temperature: 10~50°C (+14~122°F) Package Package content: DPH 130SE IP phone Handset with handset cord Ethernet Cable (1.5m CAT5E UTP Cable) Stand Quick Start Guide QIG

AUTO PROVISIONING SERVER General compatible TFTP, FTP , HTTP & HTTPs Server Software. PHONE FEATURES 2 SIP accounts, MWI(Line1) Call hold, Call waiting, Call forward, Call return; Call transfer (blind/busy/ask) Caller ID display, Redial, Mute, DND Auto answer, 3 way conferencing Speed dial, SMS, Voicemail Tone scheme, Volume control Direct IP call without SIP proxy Ring tone selection/import/delete Black list Call history: dialed/received/missed/call forwarding Soft keys programmable, Supports PC control Multi Language: Chinese, English, French, Russian, Italian, Portuguesa, Portuguesa Brasil, Spanish, Turkish ADVANCED FEATURES 9 programmable keys XML Phone Book search and input & output Enterprise phone book (2000 entries) Personal Phonebook (1000 entries) LDAP phone book IP PBX SYSTEM INTEGRATION Busy lamp field (BLF), BLF list Bridged line appearance (BLA) Intercom, Paging, Music on hold Call pickup, Dial Plan, Call recording Anonymous call, Anonymous call rejection Network conference, Distinctive ring tone CODES AND VOICE FEATURES Wideband Codec: G.722 Narrowband codec: G.711µ/A, G.726, G.729AB,i.

E HTTP/HTTPS web server Time and date synchronization using SNTP UDP/TCP/DNS SRV(RFC 3263) Qo. S: 802.1p/Q tagging (VLAN),Layer 3 To.

Part of the GRP series of

Auto Provisioning Server General compatible TFTP, FTP , HTTP & HTTPs Server Software. PHONE FEATURES 2 SIP accounts, MWI (Line1) Call hold, Call waiting, Call forward, Call return; Call transfer (blind/busy/ask) Caller ID display, Redial, Mute, DND Auto answer, 3 way conferencing Speed dial, SMS, Voicemail Tone scheme, Volume control Direct IP call without SIP proxy Ring tone selection/import/delete Black list Call history: dialed/received/missed/call forwarding Soft keys programmable, Supports PC control Multi Language: Chinese, English, French, Russian, Italian, Portuguesa, Portuguesa Brasil, Spanish, Turkish.

Audio. HD voice: HD handset Codec: G.711A/U, G.723.1 high/low, G.729AB, G.722,G.726 32 DTMF: in band, RFC2833 and SIP INFO Full duplex Acoustic Echo Canceller (AEC) –Hands free Mode, 96ms tail length Voice Activity Detection (VAD) / Comfort Noise Generation (CNG) / Background Noise Estimation (BNE)/ Noise Reduction (NR) Packet Loss Concealment (PLC) Dynamic Adaptive Jitter Buffer up to 300ms DTMF: In band, Out of Band – DTMF Relay (RFC2833)/ SIP INFO Advanced Features Call transfer (unattended/ attended/ semi attended).