IPNext180PTT is a next-generation Hybrid IP-PBX system for interworking with PSTN interface and various IP terminals of AddPac (AP-IP300 IP Phone, AP-IP120 IP Phone, etc)) to provide PTT(Push-to-Talk) server functionas well asmultimedia IP telephony services. This product is built based on the high performance embedded RISC and suitable forthe medium size companies. IPNext180PTT inter-works well with the various VoIP and video products of AddPac to provide a variety of IP application service and PTT service.
AddPac Technology PTT over IP solution is a PTT group call service total solution which can be used under various wire/wireless environments such as 3G smart phone (mVoIP) and existing radio device network.
For small and medium size companies, SIP based AddPacPTT group call solution is composed of IPNext180PTT IP-PBX, LMR(Land-to-Mobile Radio) Gateways, I-Phone & Android based smart phone SIP PTT application, PTT IP phone, and network management system NMS (Network Management System).
AddPac PTT group call solution is designed on the basis of IP network to support 1:1 and group call service in any places. It allows saving communication costs with VoIP technology. The main advantage of this device is that it supports radio device network function so that user can maintain existing radio device network system and merits. It also supports group call service by expanding the radio device network coverage to nationwide 3G network.
It is designed to expand wire/wireless PTT service users by increasing the number of PTT server. Particularly, Android smart application supports 3G (mVoIP), and Wifi dual mode. It uses the verified standard signal system so that it has both excellent compatibility and scalability.
Table 1.IPNext180PTT Usage
Service Category
Service Explanation
Supported IP Terminals
IP-PBX Features
Various Call scenario Ring Tone Service Music On Hold 3 way Conference Voice Mail IVR User Presence Service RTP Proxy Service
AP-IP300 IP Phone AP-IP250 IP Phone AP-IP230 IP Phone AP-IP160 IP Phone AP-IP120 IP Phone AP-IP90 IP Phone AP-SM100 Smart Communicator (Soft Phone) Smart Messenger for UC Smart Window for UC
PSTN Line Interface
FXO VoIP Gateway for PSTN
AP-FXO88-Port PSTN Interface Module installed
PSTN Analog Phone Interface
FXS VoIP Gateway for analog phone connection
AP-FXS88-Port FXS Analog Interface Module installed
Next-Generation Hybrid IP-PBX System
IPNext180PTT Hybrid IP-PBX system consists of CPU part for system control and two(2) VoIP module slots in backside. AddPac IPNext180PTT supports LED displays of device status and has two(2) 10/100 Mbps Fast Ethernet ports, the RS-232C console port for Command Line Interface (CLI) in front side. IPNext180PTT supports various VoIP interfaces such as FXO, FXS, E&M, and Audio Interface depending on module options. IPNext180PTT can support maximum 16 port VoIP interface (two(2) module slots x 8-Port VoIP module). The call scenarios supported by IPNext180PTT provide SIP-based basic calls, color ring services, music on hold, blind transfer, call pickup, group call pickup, consult calls, switching calls, consult transfer, call waiting, call waiting notification, call park, call pickup remote, and hunt group. This product is designed to provide application services that require much memory such as voicemail using internal memory.
Advanced PTT(Push-to-Talk) Service Features
IPNext180PTT provides the PTT server function as well as hybrid IP-PBX service features. It supports Push-to-Talk service function for LMR gateway, PTT IP Phones, WiFi Phone, 3G smart phone appl. IP based Push-to-Talk service is easy to interwork (cable, wireless) compare to pure radio based Push-to-Talk service and it has no local limitation. IPNext180PTT IP-PBX internal PTT server function supports basic Dial-Out based PtP (Point-to-Point) PTT service, Multi-Group, and Multi-Session PTT service. An audio broadcasting function for PTT service is core technologies of PTT server function. It performs broadcasting function of audio information which is transmitted from IP terminal in each group End-Point and each session End-Point and broadcast to the each group or to the entire IP terminal. PTT service causes a lot of load because it has to broadcast real time audio information to the different IP terminal so it requires high performance processing. AddPac IPNext180PTT IP-PBX is designed on the basis of high performance embedded RISC which supports medium scale PTT service. It is an ideal product for medium size corporation and support Push-to-Talk service by interworking with AddPac LMR gateway, WiFi phone, and various PTT IP phone products.
RTP Proxy Service Features for Private IP and IPv6 Address
Since the enterprise network environments configured with a Call Manager like as Hybrid IP-PBX system and IP terminals require larger number IP address, either IPv6 or a private IP address in the NAT environment must be supported to the enterprise networks due to the deficient resources of a public IP address. In such a private IP address environment, the RTP proxy feature is required for reliable multimedia communications between endpoint terminals. The RTP proxy feature of IPNext180PTT is used to make a communication between a private IP terminal and a public IP terminal among endpoint terminals such as IP phones, make a communication between a private IP terminal and a public IP terminal in the NAT environment, make a communication between private IP terminals, make audio/video broadcasting in private and public IP environments, and to enable audio/video conference calls in private and public IP environments. The RTP proxy feature can operate regardless of VoIP signaling protocols such as H.323 and SIP, and supports dual address systems such as IP version 4 and IP version 6addresses.
Intelligent IVR Features
One of the most important IP telephony service features is either ARS or IVR. Different IVR features are required by regular companies, government offices, and call centers respectively. IPNext180PTT Hybrid IP-PBX System provides an IVR tool to meet your requirements. Also, this solution provides an IVR scenario editor to allow you to make and enable a desired call scenario.
User Presence Service for Unified Communication
For enterprise with small number of employee, IPNext180PTT Hybrid IP-PBX system supports the userpresence server function beside call manager function. But, in case of large enterprise, external User Presence Server should be used with IP-PBX due to performance issue independently.In order to support user presence features on real-time basis, such as user busy, on-line, user away,etc, user presence service function should be provided on IP telephony solution.User presence service function in a Call Manager gathers user information (on-line, busy, etc) of each IP terminals, and broadcasts user information to all or group IP terminals with speed-dial keys (built-in presence indication lamp) , AP-PT100 User Presence dedicate terminal, Smart Messenger Program at every second.
Built-in Network based Media Service
AddPac Technology¡¯s IPNext180PTT supports network based media service. Together used with IP terminals, Video Terminals, VoIP gateways, IPNext180PTT supports various media service like as Announcement, Ring Back Tone Service, and Music on Hold. IPNext180PTT supports the G.711, G.726, G.729 voice codec basically. New coming and end enhanced voice and video codec (for example, H.264 video codec, etc) will be added to service profile list according to user requirement and service environment. IPNext180PTTprovide the scheme to enroll personal or group media service file, for example, ring back tone service, to IPNext180PTT. To enroll pre-exist MP3, WMV file for media service, IPNext180PTT provides the conversion software (MP3 to PMA file, etc). MP3 and WMV file can be changed to AddPac PMA file format (G.711 based voice codec) to upload in IPNext180PTTusing SMM.
Unified Messaging Service
IPNext180PTT supports network based unified messaging service like as IP phone voice mailing service, E-mail notification, and web based mailbox browsing service, etc. Basically, this service supports SIP VoIP signaling and IVR scenario managing tool for voice/video message recording/retrieval. Using XML based IVR scenario editor software (AddPac Technology provided), user can make new IVR scenario and edit the pre-existed IVR scenario for new service addition, service modification, etc. IPNext180PTT supports memory (NAND Flash Memory, etc) Quota functions for each user¡¯s voice/video mailbox, etc and e-mail notification function via internet. On business travel, user can monitor IP phone or video phone¡¯s leaving message in office via internet e-mail check and using MS window media player software. Also, user can play voice/video message under various environments such as PSTN phone via VoIP gateway, E-mail, Web, Smart Messenger (AddPac) in addition to IP terminals (IP phone, IP video phone).
Adapt to the Future Circumstances: Firmware upgradeable technology
Because the high-performance RISC CPU of IPNext180PTT is programmable, the service features of IPNext180PTT can continue to be improved, changed, or added. If you download an added or changed feature from the home page directly or set an automatic upgrade option whenever feature addition or change is done, you can use the latest features without further operations.
Reliable IP-PBX Solution with Outstanding Network Service Capability
IPNext180PTT is an integrated network device that supports routing services, NAT/PAT, DHCP Server/Relay, and Quality of Service (QoS). If you want to adapt to a variety of network environments such as metro ethernet, metro ATM, dedicated lines, flexible IP environments, and high-speed private subscriber networks such as xDSL, cable networks, and FTTH, advanced QoS and security features in addition to multiple network services should be supported. In this regard, IPNext180PTT supports two 10/100 Mbps Fast Ethernet interfaces. Based on this feature, IPNext180PTT supports advanced LAN-to-LAN routing and bridge services as well as various network and security services such as NAT/PAT. IPNext180PTTHybrid IP-PBX system is a reliable solution built by using excellent technologies.
AddPac IP-PBX Total Solution
AddPac Technology is not just a vendor of IP-PBX, but provides various product families appropriate for your network environment. To meet your needs, AddPac supports VoIP and media gateways, audio/video terminals, audio/video MCU, IP audio/video broadcast, CCTV VMS solution, audio/video recording solutions, and traffic controller QoS device solutions. In the future all IP-based multimedia telephony environment, various audio/video resources should be shared on an IP network; thus, the integration of the entire solution and that of solutions for each area are very important. AddPac IP-PBX is designed considering the integrated multimedia solution, and can meet your various needs.
The performance and reliability of AddPac VoIP gateway series and multimedia network devices have been recognized in global markets. IPNext180PTT, which is a collection of experiences and know-how accumulated in the enterprise and service provider markets, would meet the needs of customers who ask for a next-generation IP-PBX solution.
Overall Features
High Performance RISC Microprocessor
Two(2) VoIP Module Slots for FXS, FXO,etc.
Á¾·ùº°¸ðµâ»çÁøÃß°¡ (IPNext180)
Network Interface: 2-Port 10/100 Mbps Fast
Ethernet interface
1-Port RS-232C serial console interface
User Terminals : AP-IP300, AP-IP250, AP-IP230, AP-IP160, AP-IP120, AP-IP90,
etc
Supports a scenario editor for IVR services and
support tools.
Supports UMS(unified messaging service) function
(voice mail, etc)
Supports user presence service features
Media Service for Announcement, Ringback tone,
Music on Hold
Web based Smart Multimedia Manager
Fault tolerant and Scalability
Microprocessor
High Performance RISC Integrated Host Processor
Memory
Flash Memory
2Gbyte
Main Memory
128Mbyte
Boot Memory
512Kbyte Flash Memory
Network Interface
Fixed LAN0 Port
One(1) 10/100Mbps Fast Ethernet
Fixed LAN1 Port
One(1) 10/100Mbps Fast Ethernet
Console Port
One(1) RS-232C Interface for CLI
Voice Interface Module
AP-N1-FXS8
8-Port FXS Voice Interface Module (8 x RJ11)
AP-N1-FXO8
8-Port FXO Voice Interface Module (8 x RJ11)
AP-N1-FXS4O4
4-FXO & 4-FXS Voice Interface Module (8 x RJ11)
AP-E&M4
4-Port E&M Voice Interface Module (4 x RJ45)
AP-E1/T1
1-Port E1/T1 Voice Interface Module (1 x RJ45)
Power Requirement
Power VAC 110~220 VAC, 50/60Hz, 40Watt
Operating Temperature
0¡ÆC to + 50¡ÆC (32¡Æ to 122¡ÆF)
Storage Temperature
-40¡ÆC ~ +85¡ÆC (-40¡Æ ~ +185¡ÆF)
Relative Humidity
5% to 95% (Non-condensing)
Dimension(H x W x D)
55mm x 340mm x 267mm - 19" Rack Mountable
Chassis
Weight(kg)
2.5Kg
Number & Call Routing
Trunk Hunting by Preference or Sequential
Calling Hunting by Preference, Simultaneous, Random
Calling Hunting by Chained Hunting Group
Partition for Address Grading
Call Class for Call Access Control
Number Translation Rule for Inbound/Outbound Call
Centrex with Prefix Support
Multiple Shared Devices with One Number
Multiple Numbers on One Device
Individual Call Park within Park Number Pool
Group Call Park within a Group or Other Group
Call Pickup of Ringing Call of Same Group or Other Group
Call Pickup of Parked Call
Call Transfer- Blind, Consult
Call Forwarding – Unconditional, Busy, No Answer, Voice Mail
Call Waiting
Call Swapping
Call Hold
Advanced Features with
AddPac IP phone, Video Phone, etc
Multiple Call Handling with Call Status and Calling Line
Number and Name
Plug and Play with Auto Discovery Function
Softkey Map Download and Control
Telephony and Service
&
Features
Voice Mail List View
Parked Call List View
Call Forwarding Setting
Recent Call List View
Calling Number and Name Identification
Individual Call Park within a Group or Other Group by Softkey
Group Call Park within a Group or Other Group by Softkey
Call Pickup of Ringing Call of Same Group or Other Group by
Softkey
Call Pickup of Parked Call by Softkey
Call Transfer – Blind, Consult by Softkey
Call Waiting Indication
Call Swapping by Softkey
Call Hold by Softkey
Conference Control
Signaling Protocols
SIP Application Server, Proxy, Registrar and Location
Server(RFC3261)
Multiple ITSP Trunk with SIP & H.323 Account Support
- IP UA Client Role for Registering to ITSP SIP Server
- H.323 Gatekeeper Client Role for Registering to ITSP H.323 Gatekeeper
Server
IVR
(Interactive Voice Response)
& Auto Attendant
Default Auto Attendant Support
IVR Function
Provides with GUI-based Smart IVR Scenario Editor
Upload/Download Scenario by Smart IVR Scenario Editor
Supports Multiple Concurrent Scenarios
Support Recordable IVR Prompts
Voice Mail
Support Voice Mail with IVR
Access from Remote Site via Trunk Support
Voice Mail Notification Support
Conference
G.711 u-law, G.711 a-law Internal 3-Party Audio Conference
Support
Ad-Hoc Conference
Dial-Out Conference
Meet-me Conference
Multiple External MCU Support (Video, Audio, etc) :
AP-MC1000, AP-MC3000
Conference Chair and Participants Management
Music & Announcement
Music on Hold
Replaceable Announcements
Dialing Music/Tone Service
IP-PBX User & Device Management
LDAP(Lightweight Directory Access Protocol) Support
- Support Hierarchical Organization
System Performance Analysis for Process, CPU, Connection I/F
Configuration Backup & Restore for APOS Managements
Debugging, System Auditing, and Diagnostics Support
System Booting and Auto-rebooting with Watchdog Feature
System Managements with Data Logging
IP Traffic Statistics with Accounting
Other Scalability Features
DHCP Server & Relay Functions
Network Address Translation (NAT) Function
Port Address Translation (PAT) Function
Transparent Bridging (IEEE Standard) Function
¡æ Spanning Tree Bridging Protocol Support
¡æ Remote Bridging Support
¡æ Concurrent Routing and Bridging Support