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Electronics Suppliers and Manufacturers

Shenzhen Voptel Techonology Co Ltd

Description
VX12 is an all-in-one IP-PBX for small office Appliance, featuring built-in analog phone interfaces, CO line interfaces, SIP trunks and SIP client support. With features such as click-to-call, auto-attendant, on a fly recording, mobile extension, and call back, it greatly improves the communication with customers.


 System Features 
◆Built-in SIP Server
◆SIP Trunk
◆Auto Attendants
◆Distinctive Greetings
◆Greeting Mode Scheduling
◆Remote Extensions
◆Fax Detection
◆Automatic Route Selection
◆Digit Map
◆Configurable Dialing Rules
◆Music on Hold
◆Call Detail Record Storage
◆Call Back
◆Call Queue
◆Automatic Call Distribution
◆Direct Inward Dialing (DID)
◆Call Screening
◆Expandable Local Extensions
◆Calling Security (Encryption)
◆GUI Remote Management


 User Extension Features 
◆Call Forward on All Calls
◆Call Forward on Busy
◆Unanswered Call
◆Call Forward on
◆Unregistered Extension
◆Call Pickup
◆Call Hold
◆Call Transfer (explicit, and blind)
◆Call Forking
◆Simultaneous Ringing
◆Intercom
◆Call Blocking
◆Call Rejection
◆Distinctive Ring
◆Do Not Disturb
◆Hot Line
◆Speed Dialing
◆Call Barring
◆Color Ring Back Tone
◆Caller ID
◆Message Waiting Indicator
◆(FSK/Polarity Reversal)


GUI Remote Management
POTS Line Interface
◆POTS Line Interface
◆Busy Tone Detection
◆Caller ID Detection
◆Prefix Insertion
◆PSTN Failover


 Remote Management 
◆System Configuration
◆Software Upgrading
◆SIP Message Logging
◆SIP Trunks Status
◆Telnet Commands

 
Soft-Console Functions 
◆Click-to-Call
◆Extension Line Status
◆Presence of Users
◆Notification
◆Text Messaging
◆Phone Book Management
◆Call Information
◆Configuration of Extension


 Network Functions 
◆DHCP
◆PPPoE
◆DNS/DDNS
◆LAN Connection for Expansion








IP PBX

Leader Team Investments Limited

 	 Smart hybrid PBX system and medium sized business Expand from 6 analogue trunks and 18 extensions, up to 288 extensions with expansion unit Cost effective installation and operation Built in applications include call centre solution, mobile solution and voicemail Easy evolution from analogue to IP communications  

PABX System

YEASTAR TECHNOLOGY CO., LIMITED

Specification: 	 Users: 100 	 Concurrent Calls: 25 	 Voicemail: 3000min  Interface: 	 Up to 16 Analog Ports (FXO/FXS) 	 Up to 4 GSM Ports (Quad-Band GSM/GPRS850/900/1800/1900MHz) 	 Up to 4 UMTS Ports (UMTS 900/2100MHz or 850/2100MHZ or 850/1900MHZ) 	 Up to 8 BRI Ports 	 LAN: 1 (10/100Mbps) 	 WAN: 1 (10/100Mbps) 	 Size: 280 x 175 x 33 mm 	 Weight: 700g 	 Power Supply: AC 100~240V/50~60Hz (DC 12V, 5A) 	 Protocol: SIP (RFC3261), IAX2 	 Transport Protocol: UDP, TCP, TLS, SRTP 	 Codec: G.711, GSM, SPEEX, G.722, G.726, ADPCM, G.729 A, H261, H263, H263p, H264, MPEG4. 	 DTMF: Inband, RFC2833, SIP INFO 	 Echo Cancellation: Supported 	 LED: Red for FXO/GSM/UMTS, Orange for BRI, Green for FXS 	 Network: Static IP, PPPoE, DHCP, Firewall, VLAN, DDNS, QoS,&nbsp;DHCP Server, OpenVPN  Multiple Languages Support: 	 Chinese, Dutch, English, French, German, Hebrew, Korean, Italian, Polish, Portuguese, Romanian, Russian, Spanish, Thai, Turkish  Features: 	 Attend Transfer 	 Blind Transfer 	 BLF Support 	 Blacklist 	 Callback 	 CDR (Call Detail Records) 	 Call Forward 	 Call Parking 	 Call Pickup 	 Call Routing 	 Call Transfer 	 Call Waiting 	 Caller ID 	 Conference 	 Define Office Time 	 DDNS 	 Dial by Name 	 DISA (Direct Inward System Access) 	 DIDs 	 Distinctive Ringtone 	 DND (Do Not Disturb) 	 FAX (T.38) 	 Firewall 	 Follow Me 	 IVR (Interactive Voice Response) 	 Intercom/Zone Intercom 	 Mobility Extension 	 Multi-language Prompt 	 Music On Hold 	 Music On Transfer 	 MyTwins (Double System) 	 One touch record 	 OpenVPN 	 Phone Provisioning 	 PIN User (PIN Code Control) 	 Paging/Zone Paging 	 PPPoE 	 Queue 	 Ring Group 	 Route by Caller ID 	 Skype Integration (Skype Connect) 	 SMS to Mail/Mail to SMS 	 Speed Dial 	 Static Route 	 Three Way Calling 	 Voicemail 	 Voicemail to Email 	 Voicemail Forwarding 	 Web Based Control Panel 	 Spy functions (Normal Spy, Whisper Spy, Barge Spy)  

IPPBX for SMBs

Rizhao Taitung Electronic Co., Ltd

  Made of Simens 900/1800 MHZ GSM Industrial Moduel  Compatible with billing system & PBX  LED indicator for signal intensity  IP access number programmable  Polarity-reverse signal for billing  

4 Port GSM Adaptor 900/1800 MHz

Excelltel Technology Co Ltd

Capacity:
   Metal cover.   Card expandable structure.   From 8 to 16 CO lines, 32 to 128 extensions.   4 CO line ports in per CO line card.   8 extension line ports in per extension line card.  Main Features:
   Switch power.   Backup battery interface (optional).   Billing software (optional).   Transfer caller ID (optional).   Intercom caller ID (optional).   Auto-attendant (DISA: 20s three paragraphs).   Auto/Operator mode.   Automatic CO line detected.   External music input on holding.   Day/Night mode (optional).   Real time clock (optional).   Doorbell &amp; door open function (optional)   Alarmclock (optional).   Private password.   CO line transfer to CO line.   External broadcasting interface (optional).   Keyphone interface (optional).   Intelligent Routing (LCR) (optional).   Four exts. groups (optional).   Call switch(call pick during conversation) (optional).   Wireless terminal board (optional).  Basic Functions:
   Multiple communication line.   Normal transform power.   Power failure transfer.   External line caller ID.   Internal music on holding.   Remote programming.   Memory protect while power off.   System password.   Multiple dial-out mode.   External line group.   Call duration control.   Call pickup   Outgoing call transfer.   Call transfer.   Call forwarding.   Flexible coding (change ext. No. ).   Do not disturb.   Automatic Ext. reporting.   Flash time choice.   Ringing assignment.   Call restriction.   Three way conference.   Call all extensions.   Specific CO line dialing.   Ring identify.   CO line booking.   CO line reservation.   Operator setting.   CO line monitor.  Packing Details:
   Qty/Ctn: 1pc in one carton.   Weight:7.5~12.5KGS/PC.   Giftbox size: 50x24.5x34CM.  

Metal PBX

Allodotcom Inc.

  The private branch exchange manages calls and voice mails with high efficiency. The light indicators ensure easy monitoring. This advanced private branch exchange is a great choice for your office.  
 
  Programmable design. 
   8  concurrent calls. 
 

Private Branch Exchange

Specification: 	 Users: 32 	 Concurrent Calls (Max): 15 	 Voicemail: 3000min  Interface: 	 Up to 4 Analog Ports (FXO/FXS) 	 Up to 4 BRI Ports 	 LAN: 1 (10/100Mbps) 	 Size: 193 x 153 x 30 mm 	 Weight: 500gPower Supply: AC 100~240V/50~60Hz (DC 12V, 1A) 	 Protocol: SIP (RFC3261), IAX2 	 Transport Protocol: UDP, TCP, TLS 	 Codec: G.711, GSM, SPEEX, G.722, G.726, ADPCM, G.729 A, H261, H263, H263p, H264, MPEG4. 	 DTMF: Inband, RFC2833, SIP INFO 	 Echo Cancellation: Supported 	 LED: Red for FXO, Orange for BRI, Green for FXS. 	 Network: Firewall, VLAN, DDNS, QoS, DHCP Server  Multiple Languages Support: 	 Chinese, Dutch, English, French, German, Hebrew, Korean, Italian,Polish, Portuguese, Romanian, Russian, Spanish, Thai, Turkish  Features: 	 Attend Transfer 	 Blind Transfer 	 BLF Support 	 Blacklist 	 CDR (Call Detail Records) 	 Call Forward 	 Call Parking 	 Call Pickup 	 Call Routing 	 Call Transfer 	 Call Waiting 	 Caller ID 	 Conference 	 Define Office Time 	 Dial by Name 	 DISA (Direct Inward System Access) 	 DIDs 	 Distinctive Ringtone 	 DND (Do Not Disturb) 	 Firewall 	 Follow Me 	 IVR (Interactive Voice Response) 	 Intercom/Zone Intercom 	 Multi-language Prompt 	 Music On Hold 	 Music On Transfer 	 MyTwins (Double System) 	 One touch record 	 Phone Provisioning 	 PIN User (PIN Code Control) 	 Paging/Zone Paging 	 Queue 	 Ring Group 	 Route by Caller ID 	 Skype Integration (Skype Connect) 	 Speed Dial 	 Three Way Calling 	 Voicemail 	 Voicemail to Email 	 Voicemail Forwarding 	 Web Based Control Panel 	 Spy functions (Normal Spy, Whisper Spy, Barge Spy)  

Model:
   CDX-TS+208.   CDX-TS+308.   CDX-TS+416.  Feature:
   Metal cover.   Switch power (optional).   Backup battery interface (optional).   Transfer caller ID (optional).   Intercom caller ID (optional).   Auto-attendant (DISA: 10s one paragraphs).   Auto-attendant (DISA: 20s one paragraphs) (optional).   Auto-attendant (DISA: 60s or 90s four paragraphs)(optional).   Auto/Operator mode.   Auto IP dialing (optional).   External music input on holding.   Doorbell &amp; door open function (optional).   11 chord melody music selectable, auto select daily ((optional).   CO line transfer to CO line (optional).   Wireless terminal board (optional).  Basic Functions:
   Multiple communication line.   Normal transform power.   Power failure transfer.   External line caller ID.   Internal music on holding.   Remote programming.   Memory protect while power off.   System password.   Multiple dial-out mode.   External line group.   Call duration control.   Call pickup.   Outgoing call transfer.   Call transfer.   Call forwarding.   Flexible coding (change ext. No. ).   Do not disturb.   Automatic Ext. reporting.   Flash time choice.   Ringing assignment.   Call restriction.   Three way conference.   Call all extensions.   Specific CO line dialing.   Ring identify.   CO line booking.   CO line reservation.   Operator setting.   CO line monitor.  Packing Details:
   Weight: 1.7~2.0 KGS/PC.   Quantity: 5PCS in one carton.   Giftbox size: 40.7x28.8x8.3cm.   Carton size: 43.5x42x31.5cm.  

GSM PBX Telephone System

  NanoPBX is an IP-based micro pc based on open standards: SIP, Linux.  THE nanoPBX features a voice mail & interactive voice response(IVR) function.  Supports up to 30 users in a single box  4 FXS ports & 2 FXO ports  IVR scheduling  Call detail reports  Voicemail groups  Music on hold  Configure using admin web panel  T.38 fax transmission and reception  One secure digital (SD) card slot  

NanoPBX

ABC Buildtech Pvt. Ltd.

 	 Solutions for PSTN/ISDN/VOIP/GSM/PRI lines calls, digital quality recording , bundled with call analysis software, simultaneous recording of all PBX lines, easy searching and sorting of records using the software, live player for live call listening, GPRS, E-mail, Fax, and SMS logging capabilities, disk usage can be controlled.  

Voice Logger

Kingtel Corporation

   The phone uses SIP protocol.   The IP phone complies with all well-known IP-PBX brands.   The phone supports auto-provision.   The phone is rated as high-quality speakerphone as commercial grade.   The phone supports POE-power over ethernet.   Single line speaker phone.  

IP Phone

 Model: CDX-MD308 PABX System (plastic covers)

Capacity: Basic 3 CO lines, 8 extensions (308)

Main Features:
1. Auto-attendant (DISA: 10s one paragraphs)
2. Auto/Operator mode

Basic Functions:
1. Multiple communication line
2. Normal transform power
3. Power failure transfer
4. External line caller ID
5. Internal music on holding
6. Remote programming
7. Memory protect while power off
8. Multiple dial-out mode
9. Call duration control
10. Call pickup
11. Outgoing call transfer
12. Call transfer
13. Call forwarding
14. Flexible coding (change ext. No. )
15. Do not disturb
16. Automatic Ext. Reporting
17. Flash time choice
18. Ringing assignment
19. Call restriction
20. Three way conference
21. Call all extensions
22. Specific CO line dialing
23. Ring identify
24. CO line booking
25. CO line reservation
26. Operator setting
27. CO line monitor

Packing Details:
Weight: 0.9~1.2 KGS/PC
Quantity: 20PCS in one carton
Giftbox size: 23x19.5x5cm
Carton size: 49x28x41cm 

Analog Telephone Switch PABX System

Function:   Auto-Attendant(DISA: 20s three paragraphs) (CS+ series).   Auto-Attendant(DISA-12s one paragraph) (CS+ series).   Operator setting.   External line caller ID.   Transfer Caller ID (CS+ series).   Intercom Caller ID (CS+ series).   Call transfer.   Outgoing call transfer.   C. O. Line Transfer to C. O. Line.   Call pickup.   Call forwarding.   Call all extensions.   Three way conference.   Auto IP Dialing (CS+ series).   Specific CO line dialing.   CO line booking.   CO line reservation.   Ringing assignment.   Do not disturb.   Multiple dial-out modes.   Capacity: 4 Co lines, 24/32 extensionsperator /Auto-attendant.  Feature:
   Exts. Groups.   External Line Group.   Door Phone Interface (optional for CS+ series).   Door opener (optional for CS+ series).   Internal music on holding.   External Music Input on Holding.   11 Chord Melody Music Selectable (CS+ series).   Call duration control.   Call restriction.   Ring identify.   Outgoing Message Recording.   C. O. Line &amp; Exts Monitor.   Flexible coding (change ext. No. ).   Automatic Ext. Reporting.   System CRBT.   Day/Night Mode (Auto/Manual).   Multiple communication lines.   Flash time choice.   Remote programming.   System Password.   Power failure transfer.   Memory protect while power off.   Lightning Protection.   Backup Battery Interface (optional).   Normal transform power.   Switch Power (optional).  Packing:
   Weight: 2.3~4.2KGS/PC.   Quantity: 5PCS in one carton.   Gift box size: 45&times; 29&times; 11cm.   Carton size: 56&times; 47&times; 31.5cm.  

Central Telephony System

Changzhou Pioneer Electronic Co., Ltd.

 	 PE5000-3-01 12V 48V 80V 250V DC emi noise waves low pass filter 	 What are the Product details? 	 Designed for DC power line 	 Excellent differential and common-mode attenuation 	 High performance, easy to install 	 Custom design acceptable 	 What are the applications of our EMI filter? 	 Available for all kinds of electronic equipment&rsquo;s and measuring instruments(DC side):switching power supply 	 Private Branch Exchange(PBX) 	 Data communication etc 	 What are the specifications EMI filter? 	 Rated Voltage：12/24/48/80/250VDC 	 Rated Current：600A 	 Operating Frequency：50/60Hz 	 Temperature range：HPF 25/085/21 	 Test Voltage(1 min)： 	 Line to Ground :350VDC 	 Line to Line :500VDC  

Noise Waves Low Pass Filter

Uwin Technologies Co., Ltd

This cordless PABX system is designed for small office and residential PABX users. It is very convenient for use due to no wire required for installation. The system can be used in a house for the range up to 25m between the base unit and cordless extensions.
 
  DECT/DECT6.0, 2 line &amp; 4 line cordless PABX system. 
  Good performance. 
 

Cordless PABX System

Xtend Technologies Private Limited

This audio logger is a multi-channel voice logging tool that works in conjunction with audio channels, analog, digital and VoIP telephone lines. This machine helps improving customer service by enabling the upper level management to review the actual telephone conversation with customers.
 
  Reliable performance. 
  Exquisite design. 
 

Audio Logger

NeoGate TG400 is a compact 4 channels VoIP GSM gateway that connects GSM network with VoIP. It is the ideal product for small and medium sized companies with heavy demands of calls to mobile networks. The cost-saving solution makes connection cheaper and efficient

Benefits
1) Cost savings - save phone bills tremendously with mobile-to-mobile calls and LCR
2) Back up - work as a cost-effective backup when the landline network goes down
3) Easy to install - everything can be easily set up in the Web based management interface
4) Easy to integrate - high compatibility with major IP PBX and soft switch brands

Specification   Number of GSM channels (max): 4   Network type: 850/900/1800/1900MHz   Protocol: SIP (RFC3261), IAX2   Transport protocol: UDP, TCP, TLS, SRTP   DTMF: RFC2833, SIP INFO, In-band   Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.   Echo cancellation: ITU-T G.168 LEC   LAN: 1 (10/100Mbps)   Network: static IP, DHCP client, firewall, VLAN, DDNS   QoS, open VPN   NAT traversal: static NAT, STUN   Size: 213x160x44mm   Power supply: AC 100-240V 50/60Hz 0.5A MAX   Operation range: 0&deg;to 40&deg;C, 32&deg; to 104&deg; F   Storage range: -20&deg; to 65&deg;C, -4&deg; to 149&deg; F   Humidity: 10-90% non-condensing  
Features   SIP server and SIP trunk supported   SIP peer mode supported   Calling type: VoIP to GSM, GSM to VoIP   GSM ports group manage, VoIP trunk group   Incoming /outgoing routing rules   SMS sending and receiving   Send bulk SMS   Gain adjustment   USSD   PIN modify   Carrier selection: auto/manual   Balance alarm   Caller ID/CLIR   Black list   Hotline   Call duration limitation   Call transfer   Call back   Call status display   Call detail record (CDR)   Call progress tone generation   Call duration limitation for SIM card/single call   LCR (least cost routing)   Top voice quality (EFR super sound)   SIP response code switch   Open API for SMS and USSD   Real open api protocol (based on asterisk)   IP blacklist   Network attack alert   System logs   Web based configuration  

VoIP-GSM Gateway

CEM Solutions Pvt Ltd.

   The revolutionary FXS/FXO to VOIP card will be detected as a network card in all Windows and Linux systems. The OS will install the driver (network card) then you can use it with your favorite opensource PBX as a regular VOIP channel. The card itself is configured with a script. Hardware DSP for LEC (carrier grade 128ms) and transcoding.  Additional Features: 
   Compatible with 3CX, Elastix, Asterisk, Trixbox, FreePBX.   Windows/Linux compatible (all versions).   On board 128ms Line Echo Cancellation DSP.   On board Transcoding (G711, G729) DSP.   4 analog ports (FXO or FXS).   T.38 fax relay.   Pre configured for almost plug and play experience.   Clean and easy to understand GUI with many options.   Caller ID.   DTMF Digit Detection and Generation.   SIP User Agent IETF RFC3261 compliant.  

FXS/FXO To VOIP Card - Analog Card

OpenVox Communication Co., Ltd.

  Modular Design: Up to 4 FXS, FXO or mixed FXS/FXO ports per card  Scalable: Just add additional cards to extend system  World Wide Usable: Configurable line interface to meet global telephone line interface requirement  High quality with low price  Application ready: Use Asterisk, trixbox, Elastix etc. to build your IP-PBX/Voicemail system  Support Industry Standard: PCI 2.2  Both 3.3 V and 5 V PCI slot can be used for A400P  Support GPL Software driver used with Zaptel  RAM 128 + MB  Linux庐 kernel 2.4.X or 2.6.X  CPU 800+ MHZ  PCI 2.2  

PSTN Communication Adapter Card

  High performance ISDN PC cards  SDN PABX for BRI  VOIP gateways  ISDN LAN routers for BRI  ISDN least cost routers for BRI  ISDN test equipment for BRI  Four integrated S/T interfaces  ITU-T I.430 and TBR 3 certified and S/T ISDN supporting in TE and NT mode  Integrated PCI bus interface (Spec.2.2) for 3.3V and 5V signal environments  DTMF detection on all B-channels  Multiparty audio conferences bridge  Onboard power feeding  PCM bus connectors daisy chaining  Each of the 4 ports can be independently configured for TE or NT mode  Application ready: Use Asterisk to build your IP-PBX/Voicemail system  

ISDN PC Communication Adapter

    NeoGate TE200 is a dual port VoIP E1/ T1 /J1 gateway (VoIP to E1/ T1 /J1, and E1/ T1 /J1 to VoIP) that supports up to 60 concurrent calls. It&rsquo;s designed to bridge the gap between E1/ T1 /J1 and VoIP networks. TE200 offers SMBs cost effective additions to an legacy telephone system to bring the true benefits of VoIP. Integrating TE200 into an existing network will allow for inexpensive communication via SIP trunking. Also, it could connect VoIP systems with E1/T1/J1 service from legacy carriers.   Specification:   E1/T1/J1 Port:&nbsp; 2 (Support PRI, MFC R2, SS7)    Protocol: &nbsp;SIP (RFC3261)    Transport Protocol:&nbsp; UDP, TCP, TLS, SRTP    DTMF:&nbsp; RFC2833, SIP INFO, In-band    Codec:&nbsp; G.711 a/u-law, G.722, G.726, G.729 A, GSM, &nbsp; ADPCM, Speex     LAN:&nbsp; 10/100/1000 Mbps     WAN:&nbsp; 10/100/1000 Mbps    Size: &nbsp;213x160x44mm    Power Supply: &nbsp;AC 100~240V,50~60Hz (DC 12V, 1A)   Features:   Trunk Support   Call Routing Rules   Automatic appending and stripping of digits to dialed numbers   Caller ID name and number support   FAX support   Blacklist   Firewall   DDNS Support   Backup And Restore   Easy to install  Benefits:   Cost-effective Call Routing   Easy-to-navigate GUI   Simple Installation   Easy to integrate   Low power consumption for your green office  Applications:   Connect legacy PBX systems to VoIP services   Connect legacy PBX systems to remote sites over private VoIP links   Connect IP PBX systems to legacy TDM services   Phased transition from Legacy PBX to IP PBX  

VoIP-PRI Gateway (VoIP-E1/T1/J1)

   Use the ALLO GSM card to bridge calls from any Mobile GSM Network to VOIP in order to reduce mobile call cost or create new connections. Our cards are used by any size company who have employees on the road or want to create extensions in remote locations. Your PBX will become mobile and extensions roaming   ALLO drivers are available for Linux 2.4X and 2.6X Kernels for easy installation with any Open Source projects such as Asterisk, Elastix, Trixbox and FreeSwitch. ALLO GSM cards have FRONT PLATE SIM LOAD technology which allows the user to insert/remove the SIM cards without opening the computer chassis  Key Features
   Echo Cancellation for Digital audio quality   User can modify IMEI and PIN   1, 2 or 4 GSM modules   Quad-Band GSM 850/900/1800/1900 MHz   Send/Receive SMS in text mode, AT command ready   Compatible with Asterisk/FreeSwitch/Elastix/TrixBox based applications   Front Plate SIM Load Technology (switch cards in a few seconds)   GSM modules are AT &amp; T certificated   USSD, CLIP, CLIR supported   DTMF detection ready   Up to 4 Simultaneous GSM Calls   Support external antenna   Uses GSM channel driver   Bi-directional calls (Inbound and Outbound calls)   Fast and easy installation   Available in both PCI and PCIe   GPRS data connections   4 LEDs (network state indicators)   Suitable for 3.3 volts and 5.0 volts 32 bit PCI slots   Operation temperature: 0&deg;C to 50&deg;C   Storage temperature: -40&deg;C to 125&deg;C   Includes a 2 year warranty against hardware defects   30 day no hassle money back guarantee   FREE Email support included   Phone Support up to 45 minutes by appointment  

GSM Card for Asterisk PCI and PCIe

A Session Border Controller(SBC) is used to control VoIP signaling and media streams. SBC is responsible for setting up, conducting, and tearing down calls. SBC allows owners to control the types of call that can be placed through the networks and also overcome some of the problems caused by firewalls and NAT for VoIP calls. A common location for a stand-alone SBC is a connection point, called a border, between a private local area network (LAN) and the Internet. SBC polices real-time voice traffic between IP network borders ensuring your private network is robustly secure and fully manageable. 
SBC is enabled with DPI Packet Inspection on VOIP traffic, supporting the Signatures for Key Malwares/Vulnerabilities observed in SIP Deployments like Extensions Enumeration DoS and Password Cracking. Supporting Open Source PBXs like Asterisk, FreeSwitch, TrixBox. 
Handles the SIP-NAT issues observed in the common VOIP deployments. 
Topology-hiding function is to prevent customers or other service providers from learning details about how the internal network is configured, or how calls being placed through the SBC are routed. 

Basic functions   Eliminates bad VoIP signaling and media protocol at the network boundary.   Built-in firewall which can controls IP Addresses/Port based Filtering, DOS/DDOS Attacks, IP Blacklist &amp; NAT. It opens pinhole in the firewall to allow VoIP signaling and media to pass through.   Media bridging, which may include Voice over IP and Fax over IP.   Roaming Extension for Internal SIP PBX.   Support for SIP Outbound/Inbound Trunk and policies to route the calls.   DTMF Support for RFC2833/INBAND/SIP INFO   Can handle simultaneous calls from 10 to 60 channels (Including Media Transcoding and Encryption)   Easy GUI Configuration and call statistics.  
Advanced features   Transcoding SBCs can also allow VoIP calls to be set up between two phones by transcoding of the media stream, when different codecs are in u  

Session Border Controller, SBC

Securinex Inc.

This is a fixed wireless terminal that comes with a robust housing to withstand daily wear and tear. It is well engineered to deliver reliable and stable performance. This wireless terminal offers an extra-long service life.
 
  It is a 4 channels/8 channels GSM gateway that has 4 ports/ 8 ports FXS interfaces. 
  It can connect with ordinary telephone set, PBX or VoIP gateway, call back system, etc. 
  Dialing tone frequency: 450Hz. 
  SIM card supported (1.8V, 3V). 
  Caller ID display. 
  DTMF dialing. 
  Bell signal. 
  Polarity reversal. 
  The RJ-11 socket which connect the phone or billing system for metering purpose. 
  Antenna socket (SMA). 
  Working status indicator. 
  Signal strength indicator 
  Air interface standard: GSM850/1,900MHz, 900/1, 800MHz phase 2+ full band. 
  Frequency ranges: GSM850, GSM1900: transmission: 869.2-893.8MHz I) transmission: 1, 930-1, 990MHz, reception: 824.2-848.2MHz, Ii) reception: 1, 850-1,910MHz; GSM 900: GSM 1800: transmission: 890-915MHz, I) transmission: 1, 710-1,785MHz, reception: 935-960MHz, Ii) reception: 1, 805-1, 880MHz. 
  Phone interface: supply RJ-11 phone interface picking off. 
  Voltage: 30mA/41mA. 
  Antenna interface: antenna amplifying1.5dB. 
  Transmitting power: 3W. 
  Adapter interface: input: AC 175-260V. 
 

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