Virtual call centrer services

– IP phone used for 2 SIP accounts.
- Digital display, 2 RJ45 jacks, RJ9 headphone jack, voicemail indicator light.
- 2 direct phone numbers, allowing to receive 2 incoming calls at the same time.
- Function to display person's name and phone number when calling (Name is saved in customer contacts corresponding to phone number).
- Allows customizing ringtones.
- Easy to use and manage
- Use with all Grandstream IP PBX, Panasonic IP call centrer, Siemens call centrer, Asterisk call centrer...
- Network support: 2 10/100 Mbps auto-sensing Ethernet RJ45 ports (1 LAN, 1 PC).
- Support 3-way conversation and SRTP. - Headset jack RJ9.
- Function keys: Transfer, Conference, Voicemail, Record, Paging, Speakerphone, Contacts, Mute, Volume...
- Voicemail indicator light indicates a message.

Grandstream GXP1628 IP Phone
- Premium IP phone for small business users, easy to use, modern design.
- Gigabit port feature maximizes connection speed and call quality.
- Used for 2 SIP accounts, 3 XML programmable soft keys, 8 status keys
- Color LCD display 132 x 48.
- 2 direct lines.
- Simultaneous 3-party conference calls.
- Fully compatible with open standards.
- HD Audio sound quality for both speakers and headphones.
- Rich functions (display incoming calls, address book, ...)
- Allow ringtone option.
- Compatible and interoperable with SIP platforms.
- 2 10/100/1000 Mbps auto-sensing Ethernet ports.
- Full duplex Speaker Phone with AEC function.
- Standard telephony functions and features.
- Headset jack RJ9.
- Advanced 3-party conference calling function.
- Safe and automatic provisioning: Using TFTP / HTTP / HTTPS / multiple SIP accounts, SIP over TCP / TLS, SRTP.
- Support PoE network power supply function.
- Multi-language support: English, German, Italian, French, Spanish, Portuguese, Russian, Croatian, Chinese, Korean, Japanese.
- Dimensions: 220.5 x 192.5 x 76.0 mm.
- Weight: 1.2 kg.
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Grandstream GXP2130 . IP Phone
– IP phone used for 3 SIP accounts
– 2.8 inch LCD color screen, Bluetooth port, USB port
- 3 direct lines
– Concurrent 5-party conference calls
– Fully compatible with open standards
– HD Audio sound quality
– Rich functions (display incoming calls, address book, ...)
– Allow ringtone option
– Compatible and interoperable with SIP platforms
– 2 10/100/1000 Mbps auto-sensing Ethernet RJ45 ports (1 LAN, 1 PC)
– Full duplex Speaker Phone with AEC function, support PoE power supply function, 8 status keys.
– Standard phone features and functions
– Headset jack RJ9
- Advanced 5-party conference call function
– Voicemail indicator

Analog VoIP Gateway Dinstar DAG1000-4S4O
- UI: 4 FXS và 4 FXO, T.38 FAX.
- Support: G.711, G.729 A/ B, G.168.
- Network communication: 10/ 100 Base-TX.
- 1 WAN port and 3 LAN ports.
- Electricity supply: 220V AC.
- Wattage: 15W.

Analog VoIP Gateway Dinstar DAG1000-8O Tính năng Gateway DAG1000-8O-SIP
- Support 8 ports FXO
– Standard voice: G.711,G723.1,G.729 A/B
– FAX signal connection standard: T.38 hoặc G.711
– Compatible with PBX Asterisk, Elastic, Avaya, Siemen, 3CX, vv…
– Simple configuration via Web interface
– Support features: Call hold, Call waiting, Call forward, Do not disturb, Direct IP calling, Blind transfer, – – – Attended transfer
– Built-in echo cancellation module (G.168 Echo Cancellation)
- Firmware upgrade using TFTP
– Automatic time synchronization

UI: 24 FXS, T.38 FAX.
- Network communication: 10/ 100 Base-TX.
- Standard support: G.711, G723.1, G.729 A/B, G.168.
- 1 WAN port and 1 Console port.
- flexible port setting group
- Support Sip Trunk
- online and flexible operation
- Data / Voice / VLAN Management
- Voice / Fax / Modem - SNMP
- Automatic Firmware Update / Configuration Update

The dinstar GSM Gateway device UC2000-VE-8G-B is a multi-function product used to perform switching between mobile networks with Voip. Fully compatible with Dinstar SIMCloud / SIMbank solutions and easy to use.
- Connect using 8 Simultaneous mobile sims of all carriers in Vietnam
- Use for any IP PBX like Freepbx, Elastix, Trixbox, .. Switchboards that support Siptrunk
- Allows direct connection to Softphone software or IP phone for conversation
- Signal signal strong or weak,
- Standard RJ45 network interface, allowing device connection from anywhere,
- Allows to display or not display incoming calls,
- Allows to block incoming and outgoing calls,
- Support for sending SMS messages, Support for SDK(API) that allows developers to write software to send automated messages,
- Config configure device via Web interface, or IVR (Voice guide)
- SIP standards;
- Quad-band GSM: 850/ 900/ 1800/ 1900 MHz.
- LED indicator: Power, run, signal strength display.
- Point-to-point protocol over Ethernet (PPPoE);
- Hypertext Transfer Protocol (HTTP);
- Electricity supply
Input: 100-240V, 50-60Hz
Output: DC12V 4.0A
- Power consumption:
UC2000-VE-4G: 20w
- Active humidity: 10% -90% No condensation
- Dimensions (W/D/H): 250 * 156 * 32.5mm
- Weight: 1.069kg

Yealink YHS32 headset specifications:
- 1xRJ9 headset jack
- Cable length 1 m.
- Quick disconnect (QD) cord
- Transmit sensitivity: -38.0 dB re. 1 V/Pa
- Receive Sensitivity: 95dB SPL +/-3 dB SPL
- Impedance: 150 ohm

- Support SIP-T41S / T42S / T46S / T48S / T53 / T53W / T54W / T57W / T58A / VP59
- For T41S / T42S / T46S / T48S: upgrade to version 82 or higher
- USB and 3.5mm and smart devices
- Simple call management via controller
- Built-in warning and warning lights
- ActiveProtection technology protects users from sonic injury
- HD speaker performance / Wideband speaker and noise cancellation
- 330° bendable arm for easy adjustment without breaking

Headphone Yealink YHS33
- One-ear headphones line
- QD connector support
- Ready cable to connect to Yealink phone
- Compatible with Computers and all phones
- Polycom, cisco, Yealink, Grandstream,vv…