Digital Media Gateway AP-MG3000, Up to 4E1, 4T1, Digital E1/T1 VoIP Gateway (ISDN PRI, MFC-R2, etc)

AP-MG3000

AP-MG3000 Front

AP-MG3000 Rear

VoiceFinder AP-MG3000 is a high performance Media Gateway providing high scalability, cost-effective service in a small footprint. Moreover, AP-MG3000 supports major VoIP call control protocols including H.323, SIP, and MGCP concurrently on one embedded OS, and maintains state-of-art voice quality on any types of conventional IP networks by utilizing new level of QoS capability. 

Meet the needs for VoIP Trunking, Security, and IP Centrex 

VoiceFinder AP-MG3000 offers 120 channels of unprecedented quality of digital VoIP com munication with the density requirement. Not only general QoS algorithm, AP-MG3000 supports the advanced QoS algorithms to maintain state-of-art voice quality on any types of conventional IP networks. Also, the front panel of AP-MG3000 incorporates 120 LEDs clearly indicating operational status of VoIP channels such as channel occupation status, security enable/disable status. 

High-Capacity Media Gateway Solution for Enterprises

AddPac VoiceFinder VoIP gateways have been recognized with their unmated level of performance and reliability in the IP telephony market. With the years of experience and the know-how at the enterprise and carrier-class markets, AP-MG3000 Media Gateway delivers true voice over IP service to businesses worldwide. Moreover, AP-MG3000 supports multi-stack operation of H.323, SIP and MGCP control protocols which well suitable for any commercial VoIP deployment. 

Best Choice for Secure VoIP Communication

In Internet environment, security is the issue that needs to be confronted. AP-MG3000 takes upon this complicated task with its built-in security processor, the first-of-its-kind in commercial Media Gateway market. Especially, choosing AP-MG3000 enables service providers to rapidly deploy future secured service. Including AP-MG3000, AddPac Technology carries SRTP protocol based secure VoIP Gateways such as VoiceFinder AP500S, AP2520S. AP-MG3000 is the right choice for secured, next generation communication.

Network Diagram

AP1820 Dual Power Analog/Digital VoIP Gateway (FXS, FXO, E&M, E1/T1)

VoiceFinder AP1820 is a new cutting edge VoIP gateway supporting maximum 16 ports of analog voice interface and two(2) digital E1/T1 voice interface. Analog and digital interface of AP1820 provide an optimized call scenario when it interoperates with conventional PBX. Compact cost effective design and system architecture of AP1820 provides customer satisfaction in high quality, performance and system reliance.

Supports analog and digital interface concurrently

VoIP gateway combining IP-PBX is now suggesting a new model for a main voice communication solution. In order to be a part of advanced VoIP communications naturally in the future, making an excellent choice of choosing VoIP gateway is essential. VoiceFinder AP1820 is a mid-range VoIP gateway providing total 16 analog interface ports or two(2) digital E1/T1 interfaces. It is suitable for general enterprises, small and public offices. It has functions as a media gateway which interoperates with IP-PBX and it enables to replace small and mid size analog PBX. Especially AP1820 provides an optimal solution of VoIP communications in telephony environments that using telephone lines and PBX to get connect with a head office.

Powerful Network·CPU Interface

AP1820 is designed to provide high-performance service as a media gateway. It provides LEDs on the front panel indicating operational status, line status, and LAN Interface status of gateway. Two(2) 10/100Mbps fast ethernet port and a RS232C console interface are the base for delivering various network services such as LAN-to-LAN routing, Bridge and NAT/PAT service etc

High quality of communication and additional services

Various VoIP gateway series of AddPac is approved for its high performance and reliability in world wide markets. AddPac provides highly advanced VoIP services in order to meet demanding and evolving business requirements of customers. Especially AP1820 has been developed through years of experiences and accumulated know-how from existing enterprises and communication markets. AP1820 provides concurrent triple stack such as H323, SIP and QoS (Quality of Service) for high quality of communication service. This will satisfy customers with high expectation. 

Network Diagram

VoiceFinder AP1820 Dual Power VoIP Gateway Release, Dual Power Digital Analog Hybrid VoIP Gateway

VoiceFinder AP1820 is a new cutting edge dual power supply VoIP gateway supporting maximum 16 ports of analog voice interface and two(2) digital E1/T1 voice interface. Analog and digital interface of AP1820 provide an optimized call scenario when it interoperates with conventional PBX. Compact design and system architecture of AP1820 provides customer satisfaction in high quality, performance and system reliance. 

VoiceFinder AP1820 is a mid-range VoIP gateway providing total 16 analog interface ports or two(2) digital E1/T1 interfaces. It is suitable for general enterprises, small and public offices. It has functions as a media gateway which interoperates with IP-PBX and it enables to replace small and mid size analog PBX. Especially AP1820 provides an optimal solution of VoIP communications in telephony environments that using telephone lines and PBX to get connect with a head office.

AP1820 is designed to provide high-performance service as a media gateway. It provides LEDs on the front panel indicating operational status, line status, and LAN Interface status of gateway. Two(2) 10/100Mbps fast ethernet port and a RS232C console interface are the base for delivering various network services such as LAN-to-LAN routing, Bridge and NAT/PAT service etc

Various VoIP gateway series of AddPac is approved for its high performance and reliability in world wide markets. AddPac provides highly advanced VoIP services in order to meet demanding and evolving business requirements of customers. Especially AP1820 has been developed through years of experiences and accumulated know-how from existing enterprises and communication markets. AP1820 provides concurrent triple stack such as H323, SIP and QoS (Quality of Service) for high quality of communication service. This will satisfy customers with high expectation. 

Network Diagram

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Small, but Powerful SIP VoIP Paging Terminal, SIP Public Announcement Terminal AP602 Release

Main Features

- Standard SIP Protocol and AddPac Proprietary Protocol Support 
- RTP/RTSP Transport Protocol Support 
- RISC + Voice DSP Embedded H/W Architecture
- 40Watt Digital AMP. (Option)
- Two(2) 10/100Mbps Fast Ethernet 
- Various High Quality VoIP Codec Support : G.722(16Hz), G.711,G.726,G.729A,G.723,1 etc
- I/O Interface S : Alarm Input 1-Port, Relay Output 2-Port, I2C Interface
- Volume Up/Down Button

Voicefinder AP602 SIP paging terminal provides high-performance SIP signaling based VoIP paging service for bank, securities, military, large enterprise, and government. 

AP602 SIP paging terminal should be used together with SIP paging server for dedicated IP paging service, or can be connected to legacy SIP call manager like as IP phones for SIP paging service. Generally, SIP call manger supports the SIP paging service for simple paging service via IP phones. Using SIP VoIP signaling and internal 40watt digital amplifier, AP602 performs SIP paging service for overall room paging announcement. At front side, this device provides UP, DOWN volume button for easy speaker volume control. 

AP602 SIP paging terminal provides excellent performance and satisfaction for integrated security management environment like as intrusion detection application (government oil station, military, national power plant, etc). Installation and management is very simple. Beginners can handle the device easily. 

AddPac SIP paging service solution consists of SIP paging server, SIP paging management system and AP602, AP1605 SIP paging terminals. Designed on the foundation of high performance embedded RISC CPU + Voice DSP, AP602 SIP paging terminal supports two(2) 10/100Mbps fast ethernet interface, one(1) port RS232C console port. Providing built-in 40watt internal digital amplifier, AP602 SIP paging terminal is designed to support VoIP (Voice over IP) paging service without external amplifier via directly connected to SPEKER. And, this paging terminal supports high quality 16KHz G.722 voice codec beside traditional 8KHz G.711, G.726, G729A, G.7231.1,etc. At rear side, this device supports I2C interface and Alarm Input 1-Port, Relay Output 2-Port for various application services.

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SIP Paging Terminal AP1605, SIP Voice Broadcasting Terminal

AP1605

AP1605 Front

AP1605 Rear

VoiceFinder AP1605 SIP paging terminal is a SIP based VoIP broadcasting terminal which supports audio, voice broadcasting service through internet IP protocol in headquarter and branch. It provides high definition audio quality (up to 16KHz) by using high quality audio compressed codec such as other than existing 8KHz based PSTN. At front, it provides volume adjust rotary switch so that a user may adjust volume in accordance with the needs of user. Also, it provides built-in digital amp broadcasting module with direct connection of external speaker. AP1605 SIP paging terminal supports an excellent solution for audio broadcasting through internet in enterprise environment to maximize the full satisfaction. AP1605 SIP paging terminal terminal is designed on module type to support audio broadcasting interface module such as built-in amp. Installation and management are easy for the beginner.

Hardware Type High Performance Video Broadcasting Terminal

AddPac SIP paging service solution is composed of AP-SPS10000 SIP paging server, AP-BCR5000 broadcasting router and AP1602, AP1605, AP2620 SIP paging terminals. AP1605 HD audio broadcasting terminal is designed to install 1 module slot to support high quality audio broadcasting. AP1605 audio broadcasting module is designed to support high quality audio codec basically. In addition to, this terminal supports PSTN voice quality codec such as G.711, G.726, etc for backward compatibility. AP1605 SIP paging terminal combines with AddPac DSP hardware and high performance processor for real time decoding/play of HD audio signal.

Real Time HD Live Audio

AddPac HD IP audio broadcasting terminal opens a whole new world of real time audio broadcasting system based on high performance and stability. Compressed audio signal such as support stable high quality audio in data convergence network through QoS. It is designed to decode the encoded data in an appropriate bit rate from broadcasting server in accordance with network status.

Support Unicasting/Multicasting Protocol

AP1605 SIP paging terminal supports both multiple point-to-point method unicasting IP broadcasting scheme and IGMP based multicasting routing scheme

High Performance Network CPU Interface

AP1605 SIP paging terminal is designed to support high quality/high performance service. It supports status check LED to display the device status as well as blue lamp to check On-Air broadcasting status at a glance. It also supports one 10/100Mbps fast ethernet port and RS232C console interface for various network services.

Audio Broadcasting Feature and Supplementary Service

AddPac IP broadcasting series have been fully recognized in terms of its performance and stability throughout the world. With our accumulated experience in enterprise market and service provider market, AP1605 IP broadcasting terminal will satisfy the needs of customer along with AddPac SIP paging solution. It provides stable service by using QoS algorithm feature.

Network Diagram

AP2650 High Performance Dual Power 32 Port VoIP Gateway (Up to 2 Port Digital E1/T1, 32 Port FXS, 32 Port FXO, 32 Port E&M, etc)

AP2650 VoIP Gateway

VoiceFinder AP2650 is a cutting edge VoIP gateway supporting maximum 32 ports of analog voice interface and digital E1/T1 voice modules. Analog and digital interface of AP2650 provide an optimized call scenario when it interoperates with analog PBX. In the aspects of adoptability and scalability, AP2650 delivers outstanding functions and performance with power redundancy and changeable CPU/network modules etc.

Supports 32 ports of analog and digital interface concurrently

VoIP gateway combining IP-PBX is now suggesting a new model for a main voice communication solution. In order to be a part of advanced VoIP communications naturally in the future, making an excellent choice of choosing VoIP gateway is essential. AP2650 of AddPac is a mid-range VoIP gateway providing total 32 analog interface ports with 2 of digital E1/T1 interfaces. It is suitable for general enterprises, small and public offices. It has functions as a media gateway which interoperates with IP-PBX and it enables to replace small and mid size analog PBX. Especially AP2650 provides an optimal solution of VoIP communications in telephony environments that using telephone lines and PBX to get connect with a head office.

Network·CPU module structure, Power redundancy

AP2650 is designed to provide high-performance service as a media gateway. It has changeable CPU and network interface modules which provide LEDs on the front panel indicating operational status of gateway. The main feature of AP2650 is that it increases reliability in the aspects of hardware by providing power redundancy and changeable modules.
Two of 10/100Mbps fast Ethernet port and RS232C console interface on module-type CPU cards are the base for delivering various network services such as LAN-to-LAN routing, Bridge and NAT/PAT service etc.

High quality of communication and additional services

Various VoIP gateway series of AddPac is approved for its high performance and reliability in world wide markets. AddPac provides highly advanced VoIP services in order to meet demanding and evolving business requirements of customers. Especially AP2650 has been developed through years of experiences and accumulated know-how from existing enterprises and communication markets. AP2650 provides concurrent triple stack such as H323, SIP, MGCP and QoS (Quality of Service) for high quality of communication service. This will satisfy customers with high expectation.

Flexible for environmental changes

AP2650 is designed to be flexible for environmental changes such as increasing flash, main memory size using SODIM type socket memory slot. So when configuring IPv6, IVR or VoIP based recoding features on AP2650, it is easy to access new features.

Network Diagram

AP3110 IP Broadcasting Server

VoiceFinder AP3110 small size IP voice broadcasting server is installed in enterprise, government office, turner, and school network. It is a fast internet access small size IP broadcasting device which provides voice broadcasting transmission function through internet network using IP protocol in headquarter, branch and site. It also provides previously existing 8KHz based PSTN voice quality broadcasting server function.

VoiceFinder AP3110 IP voice broadcasting server supports an excellent solution for voice broadcasting task help through enterprise environment internet network. With an excellent performance, it is designed to maximize the full satisfaction of user. Also, VoiceFinder AP3110 IP voice broadcasting server supports two stabilized fast Ethernet as well as 2 maximum audio broadcasting codec module. Installation and operation is easy with beginner. 

Hardware Type High Performance Voice Broadcasting Server 

AddPac IP voice broadcasting solution is composed of AP3110 IP voice broadcasting serve, AP1601 IP voice broadcasting terminal, and AP-EMBMS broadcasting management system. AP3110 broadcasting server is designed to install 2 module slots and provide maximum of 4 voice broadcasting. AP3110 broadcasting module is design to support PSTN voice quality codec such as G.711 and G.729, G.723 to accept under inferior internet environment. AddPac AP3110 IP broadcasting server is combined with high performance DSP hardware and high performance processor to encode/transmit high quality voice signal in real time. 

Real Time Live IP Voice Broadcasting 

AddPac AP3110 IP voice broadcasting server opens a whole new world of high quality real time voice broadcasting based on an excellent performance and stability. A video image with compressed G.711 supports stabilized high quality audio through QoS algorithm in Broadband convergence Network. Especially, user may able to use voice codec to setup an appropriate bandwidth in accordance with network status so that it would accept from low bandwidth to high bandwidth network environment.

Unicasting/Multicasting Protocol 

AP3110 IP voice broadcasting server provides Multiple Point-to-Point Unicasting IP audio broadcasting function as well as IGMP based Multicasting Routing. It can be used or setup properly in accordance with user network operation as well as to provide economical/efficient solution as a fast Ethernet access IP broadcasting system. 

Powerful Network, CPU Interface

AP3110 IP voice broadcasting server is designed to support high performance service in terms of hardware architecture aspect as IP multichannel broadcasting server. With status check LED, user may able to verify line status, LAN interface status, and CPU status at a glance. Two fast Ethernet 10/100Mbps ports and RS232C console interface is a base of various network services such as LAN-to-LAN routing service, bridge service, and NAT/PAT service. 

High Satisfaction Audio Broadcasting and Additional Service 

AddPac’s various IP broadcasting device series are well recognized along with VoIP products domestically and internationally. With our accumulated experience in enterprise market and communication market, AP3110 small capacity IP voice broadcasting server will satisfy the needs of customer in terms of various services. It supports AddPac QoS algorithm function to support stabilized service function in internet network

Network Diagram

H.323 VoIP Gatekeeper AP-GK3000

AP-GK3000

AP-GK3000 Front

AP-GK3000 Rear

VoiceFinder AP-GK3000 is embedded Gatekeeper exclusive system offering called party information for call routing and terminal management to H.323 Client system such as VoIP gateways and various VoIP end points on enterprise VoIP network.

VoiceFinder AP-GK3000 Gatekeeper system is focused on enterprise customers offering basic function of call routing and various additional functions such as primary and secondary Gatekeeper configuration for fault tolerance. AP-GK3000 Gatekeeper system can be the ultimate solution for customers requesting expensive Gatekeepers for limited functions of call routing and dynamic IP supports due to reliability issues.

AP-GK3000 is aim to provide reliable VoIP telephony service even without network managers by adopting embedded OS based exclusive system, unlike commonly used OS platform. It focuses on Gatekeeper functionality realizing the best price vs. performance ratio. Also, it offers secure service along with H.245 crypto-token based security function. Moreover, AP-GK3000 offers web-based management tools and end-point operation monitoring, so the users can form effective VoIP communication environment with less efforts.

AP-GK3000 is the ultimate solution for customers considering Gatekeepers but reluctant due to its expenses and management issues.

Network Diagram

AP2370 56 Port Large Capacity Analog VoIP Gateway (Analog/Digital 7 Module Slot)