Professional IP Phone
- 2 lines gray graphic(128×32)
- 4 programmable feature keys with 3 color LED
- Headset / Speaker phone
- PoE(802.3af)
- LLDP-MED / 802.1x security support
- Open VPN support
Equipment
Equipment
มีอุปกรณ์ด้าน ICT อย่างครบวงจรสามารถให้บริการได้อย่างมีคุณภาพ โดยทางบริษัทเป็นตัวแทนจำหนายอุปกรณ์ชั้นนำ ดังนี้ CISCO , WatchGuard , Paloalto , ericsson LG, Sonus, Audiocode, dinstart, Uniview
IP Phones and SIP Phones
Enjoy a wide selection of easy to use system IP phones to enrich your communications experience with a powerful speaker phone and wideband voice support. There are 5 types of phones from Entry to Advanced level, designed to best fit the users’ business needs. Ericsson-LG Enterprise proprietary protocol provides fully integrated system features just like the conventional features in rich PBX solution and SIP protocol is supported as well.
Products
Simple functionality
for a basic-level IP Phone
- 3 lines gray graphic with backlit(320×48)
- 5 programmable feature keys with 3 color LED
- WB voice for Handset / Speaker phone
- PoE(802.3af)
- LLDP-MED / 802.1x security support
- Open VPN support
Standard Gigabit IP Phone
- 5 lines gray graphic with backlit(320×80)
- 10 programmable feature keys with 3 color LED
- WB voice for Handset / Speaker phone
- PoE(802.3af)
- LLDP-MED / 802.1x security support
- Open VPN support
Standard Gigabit IP Phone
- 5 lines gray graphic with backlit(320×80)
- 10 programmable feature keys with 3 color LED
- WB voice for Handset / Speaker phone
- PoE(802.3af)
- LLDP-MED / 802.1x security support
- Open VPN support
Mid range Gigabit IP Phone
- 7 lines gray graphic with backlit(320×112)
- 24(8x3page) programmable feature keys with 3 color LED
- WB voice for Handset / Speaker phone
- PoE(802.3af)
- LLDP-MED / 802.1x security support
- Open VPN support
- UC enabled (IM Presence of UC client with UCP)
Professional Gigabit IP Phone
- 9 lines gray graphic with backlit(320×144)
- 36(12x3page) programmable feature keys with 3 color LED
- WB voice for Handset / Speaker phone
- PoE(802.3af)
- LLDP-MED / 802.1x security support
- Open VPN support
- UC enabled (IM Presence of UC client with UCP)
High-end IP video Phone
with diverse UC features
- Dialing(DTMF digits)
- Adjustable ringing,Handset, Headset, Speaker Phone volume levels
- Adjustable display brightness
- Auto-answer
- Call back
- Call forward and forward notification
- Call history lists(Up to 500)
- Call auto dial(From Recent call list)
- Call park / pickup / timer / waiting
- Caller ID •Conference(Ad-hoc)
- Direct transfer
- Do Not Disturb(DND)
- Group call pickup
- Hold
- Intercom
- Last-number redial(LNR)
- Message Waiting Indicator(MWI)
- 3-way audio conference
- Mute(Audio and Video)
- Distinctive ringing
- Personal directory(Up to 2,000)
- Predialing before sending
- Self view(Video call)
- Shared lines – BLA buttons(Up to 48)
- BLF/DSS buttons(Up to 48)
- Voicemail
- Station SMS(Between iPECS Stations only)
- 17 useful built-in applications
ACCESSORY
- 12 keys with triple color LED
- Paper underlay
- Power via keyset
SONUS
- 12 keys and 2 pages button with triple color LED
- LCD underlay
- Power via keyset
- 24 keys with triple color LED
- Paper underlay
- Power via keyset
- 48 keys with triple color LED
- Paper underlay
- Power via keyset
SBC
A Session Border Controller (SBC)
is a device regularly deployed in Voice over Internet Protocol (VoIP) networks to exert control over the signaling and usually also the media streams involved in setting up, conducting, and tearing down telephone calls or other interactive media communications.
Scalable SIP Security for Small Business and Enterprise
NAONWORKS’s VIPER-N is a powerful, flexible and cost-effective E-SBC for SIP connectivity, security and interoperability, such as connecting PBXs and UC solutions to SIP trunking service providers. Unlike Data firewall, NAONWORKS’s VIPER-N can open/close rtp ports dynamically by parsing SIP messages and so it is the optimal firewall for VoIP
The NAONWORKS’s VIPER-N offers
- Industry-standard session border control functionality for Enterprise
- support up to 10,000 sessions just with one system
- Flexible software based architecture enables easy and quick accommodation of service provider’s specific requirements.
- Roles as VoIP Firewalls and E-SBC
- Common Criteria Certified EAL3
- TTA Ver3 and IPv6 Verified
SONUS
Voice Gateway
ทำหน้าที่แปลงสัญญาณเสียงแบบแอนะล็อก (Analog) ให้อยู่ในรูปของข้อมูลดิจิทัล (Digital)
Mediant 800 E-SBC and Media Gateway
The AudioCodes Mediant 800 Enterprise Session Border Controller (E-SBC) and Media Gateway offers a complete connectivity solution for small-to-medium sized enterprises. Supporting up to 60 voice channels in a 1U platform, the Mediant 800 provides versatile connectivity between TDM and VoIP networks.
Benefits
- Supports a wide variety of qualified SIP Trunks, SIP Platforms and IP Cloud services
- The “all-in-one” VoIP Media Gateway and E-SBC lowers CAPEX and reduces space and power footprints
- Simplified integrated management reduces OPEX
- A highly integrated device for secured SIP Trunking and PSTN access, forming a single and managed point of demarcation for VoIP networks
- Optional Open Solution Network (OSN) Platform for hosting value-added applications
Features
- Rich and powerful SIP normalization and routing mechanism for seamless interoperability
- Support for analog and digital TDM interfaces
- Hybrid SBC enables seamless migration and PSTN fallback
- Enhanced perimeter defense against DoS attacks
- Advanced monitoring tools help analyze and optimize VoIP service quality-of-experience
- Embedded DSP performs media handling (codec transcoding, DTMF, fax, call progress announcements, etc.)
- High Availability using two box redundancy
Mediant 1000 E-SBC and Media Gateway
The AudioCodes Mediant 1000 Enterprise Session Border Controller (E-SBC) and Media Gateway is designed to provide a complete connectivity solution for small-to-medium sized enterprises. Supporting up to 192 concurrent voice sessions in a 1U modular platform, the Mediant 1000 provides versatile connectivity between TDM and VoIP networks.
Benefits
- Support a wide variety of qualified SIP Trunks, SIP Platforms and IP Cloud services
- The “all-in-one” VoIP Media Gateway and E-SBC lowers CAPEX and reduces space and power footprints
- Simplified integrated management reduces OPEX
- A highly integrated device for secured SIP Trunking and PSTN access, forming a single and managed point of demarcation for VoIP networks
- Scalable “pay-as-you-grow” modular architecture
Features
- Rich and powerful SIP normalization and routing mechanism for seamless interoperability
- Support for analog and digital TDM interfaces
- Hybrid SBC enables seamless migration and PSTN fallback
- Enhanced perimeter defense against DoS attacks
- Advanced monitoring tools help analyze and optimize VoIP service quality-of-experience
- Embedded DSP performs media handling (codec transcoding, DTMF, fax, call progress announcements, etc.)
- E1/T1/J1 bypass and dual power supplies for high-availability
Mediant 2000 Media Gateway
The AudioCodes Mediant 2000 Media Gateway is designed to provide optimal connectivity between small-to-medium sized enterprises and service providers’ networks. Supporting up to 480 voice channels in a 1U platform, the Mediant 2000 offers superior performance in connecting legacy telephone and PBX systems to IP networks, as well as seamless connection of IP-PBXs to the PSTN.
Benefits
- A highly integrated device for secured SIP Trunking and PSTN access
- Increase flexibility with support for a wide variety of call control protocols
- Scalable to meet the demand of enterprises and small carrier POPs
GSM VoIP Gateway
DWG2000G-32G with 128 SIM Slots GSM VoIP Gateway
- 32 GSM Channels with 128 SIM Slots, Easy to Conduct SIM Rotation Locally
- GSM: 850/900/1800/1900MHz; CDMA: 800MHz; WCDMA: 850/900/1900/2100MHz
- Built-in Bandwidth Saving, Signaling & RTP Encryption
- HTTP API for Bulk SMS
- Auto Call/SMS Generation
- Auto Update SIM Number and Balance Check
- No Balance Alarm and Call Minutes Limitation
DWG2000E-4G GSM/CDMA/WCDAM VoIP Gateway
- 4 GSM/CDMA/WCDMA Channels
- GSM: Quad-band 850/900/1800/1900MHz
- CDMA: 800MHz
- PSTN and VoIP Hotline
- Up to 8 Concurrent Calls
- Flexible Dial Rules and Manipulation Rules
- SMS Sending and Receiving
- USSD, Open API for SMS/USS
- BCCH
- Carrier Selection
- PIN Code Management
- Remote SIM Card Management (SIMCloud)
DWG2000F-16G with 64 SIM Slots GSM VoIP Gateway
- 16 GSM Channels with 64 SIM Slots, Easy to Conduct SIM Rotation Locally
- GSM: 850/900/1800/1900MHz; CDMA: 800MHz; WCDMA: 850/900/1900/2100MHz
- Built-in Bandwidth Saving, Signaling & RTP Encryption
- HTTP API for Bulk SMS
- Auto Call/SMS Generation
- Auto Update SIM Number and Balance Check
- No Balance Alarm and Call Minutes Limitation
DWG2000E-8G GSM/CDMA/WCDMA VoIP Gateway
- 8 GSM/CDMA/WCDMA Channels
- GSM: Quad-band 850/900/1800/1900MHz
- CDMA: 800MHz
- PSTN and VoIP Hotline
- Up to 8 Concurrent Calls
- Flexible Dial Rules and Manipulation Rules
- SMS Sending and Receiving
- USSD, Open API for SMS/USSD
- BCCH
- Carrier Selection
- PIN Code Management
- Remote SIM Card Management (SIMCloud)
DWG2000-1G GSM VoIP Gateway
- 1 GSM ports
- GSM: quad-band 850/900/1800/1900MHz
- Web-based SMS/SMSC and USSD, open API
- Support digit map, flexible dial rules
- Manual /automatic selection operators
- Syslog way output tracking information
- IP/Password based white/blacklist
- Voice prompts, HTTP Web support for configuration and updates
- BCCH management
DWG2000F-16G GSM/CDMA/WCDMA VoIP Gateway
- 8/16 GSM/CDMA Channels
- GSM: Quad-band 850/900/1800/1900MHz
- CDMA: 800MHz
- PSTN and VoIP Hotline
- Up to 16 Concurrent Calls
- Flexible Dial Rules and Manipulation Rules
- SMS Sending and Receiving
- USSD, Open API for SMS/USSD
- BCCH
- Carrier Selection
- PIN Code Management
- Remote SIM Card Management (SIMCloud)
DWG2000G-32G GSM/CDMA/WCDMA VoIP Gateway
- 16/32 GSM/CDMA/WCDMA Channels SMS Sending and Receiving
- GSM: Quad-band 850/900/1800/1900MHz USSD, Open API for SMS/USSD
- CDMA: 800MHz(Optional) PSTN and VoIP Hotline
- WCDMA: 900/2100MHz(Optional) BCCH
- Up to 32 Concurrent Calls PIN Code Management
- Flexible Dial Rules and Manipulation Rules Remote SIM Card Management(SIMCloud)
Trunk Gateway
MTG200
- Cost-effective VoIP trunk gateway
- Support echo cancellation, DJB, VNG, VAD and Qos
- Voice codec option: G.711A/U,G.723.1 (option), G.729A/B(option), iLBC(option)
- Simple Configuration, friendly GUI, strong handling capacity,stability system
- Good compatibility, support Asterisk, Elastix, Freeswitchand Small and medium IPPBX platform
MTG600
- The strong performance of hardware platform, energy efficiency concurrent processing
- Service oriented architecture, rich services, in addition to support voice and IP fax outside, still can be expanded to fax server
- Flexible dial-up rules and operation, in order to adapt to the different work scene.
- Voice codec: G.711A/U,G.723.1,G.729A/B, iLBC
- Good compatibility, support Asterisk, Elastix, Freeswitch and Small and medium IPPBX
MTG1000
- Energy efficiency concurrent processing, up to 240
- Service oriented architecture, rich services , support voice, IP fax and Modem/POS service
- Flexible dial-up rules and operation, in order to adapt to the different work scene.
- Voice codec: G.711A/U,G.723.1,G.729A/B, iLBC
- Good compatibility, support Asterisk, Elastix, Freeswitch and Small and medium IPPBX platform
PBX
FEATURES
- Integrated Auto Attendant / Voice Mail
- IP-Attendant
- Centralized Attendant
- E-mail Notification
- Centralized Voice Mail
- Personal Groups
- Mobile Extension
- Automatic Call Distribution (ACD)
- Click to Call
- Green Power Save
FEATURES
- Embedded UC
- Embedded AA/VM
- VM to Email forwarding
- Embedded Audio conference
- Embedded ACD
- One number service
- Mobile extension service
- Hot desking
- Teleworking
- Click to Call
- Emergency paging
- Emergency call monitoring and recording
- Multiple call handling
FEATURES
- iPECS UCS Client Support
- Mobile Extension
- Auto Attendant/Voice Mail Group
- E-Mail Notification
- Centralized Voice Mail
- Embedded Hotel Features
- Embedded SIP
- Automatic Call Distribution (ACD)
- One Number Service
- Web Administration
- Click to Call
- Centralized Control T-NET
- System Geographic Redundancy
Snack+
SNACK+FMC เหมาะกับลูกค้าที่มีความต้องการดังนี้
- ต้องการใช้ Mobile Application สำหรับการโทร ส่งข้อความ รูปภาพและวิดีโอ ภายในบริษัท
- มีคุณภาพเสียงระดับ HD Voice
- ต้องการเชื่อมต่อกับตู้โทรศัพท์สาขาเดิมที่ใช้อยู่ เพื่อเพิ่มความสามารถในการติดต่อสื่อการผ่าน Mobile Application
- มีระบบแจ้งประกาศภายในบริษัท พร้อมแยกประกาศในแต่ล่ะแผนกได้
- มีระบบ Directory ที่ง่ายในการติดต่อประสานงานในแต่ล่ะส่วนของบริษัท
- สามารถแสดงสถานะของพนักงานในบริษัทได้
- มี emoticon sticker และสามารถออกแบบฉพาะให้บริษัทได้
- มี api สำหรับเชื่อมต่อและพัฒนา mobile application เฉพาะเพื่อธุรกิจของบริษัทได้อย่างมีประสิทธิภาพ