Asterisk Brochure Locanet

8/8/2019 Asterisk Brochure Locanet

1/24


2/24

Contents

Asterisk - the Open Source PBX! ...............................................................3Anatomy of an Asterisk PBX.......................................................................4

Supported VoIP Protocols ....................................................................6Important advantages of Asterisk .........................................................7Myths Versus Reality ............................................................................8

Asterisk™ Features ....................................................................................9Asterisk™ Architecture ............................................................................. 11

Loadable Module APIs .......................................................................12Asterisk™ Hardware.................................................................................14

Digium TDM400P ...............................................................................14Digium TDM2400P .............................................................................15

Digium Wildcard TE110P ....................................................................16Digium Wildcard TE210P ...................................................................17Digium Wildcard TE410P ...................................................................18Digium Wildcard TE411P ....................................................................19BeroNet BN4S0 ISDN Board ..............................................................20BeroNet BN8S0 ISDN Board ..............................................................21Sirrix.PCI4S0 ......................................................................................22Digium S101I („IAXy“) ........................................................................23


3/24

3

Asterisk - the Open Source PBX!

Asterisk is a complete PBX in software. It runs on Linux1 and provides allof the features you would expect from a PBX and more. Asterisk does voice

over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

Asterisk provides Voicemail services with Directory, Call Conferencing, In-teractive Voice Response and Call Queuing. It has support for three-waycalling, caller ID services, ADSI, SIP and H.323 (as both client and gate-way). Check the Features section later in this document for a more com-plete list.

Asterisk needs no additional hardware for Voice over IP. For interconnec-tion with digital and analogue telephony equipment, Asterisk supports anumber of hardware devices, most notably allof the hardware manufactured by Asterisk‘ssponsors, Digium™. Digium has single andquad span T1 and E1 interfaces for intercon-nection to PRI lines and channel banks as wellas a single port FXO card and a one to four-port modular FXS and FXO card.

Asterisk supports a wide range of TDM protocols for the handling and trans-mission of voice over traditional telephony interfaces. Asterisk supports USand European standard signalling types used in standard business phonesystems, allowing it to bridge between next generation voice-data integra-ted networks and existing infrastructure. Asterisk not only supports traditio-nal phone equipment, it enhances them with additional capabilities.

Using the Inter-Asterisk eXchange (IAX™) Voice over IP protocol, Asterisk

merges voice and data traf fic seamlessly across disparate networks. Whileusing Packet Voice, it is possible to send data such as URL information andimages in-line with voice traf fic, allowing advanced integration of informa-tion.

1 Asterisk is primarily developed on GNU/Linux for x/86. It is known to compile and run on GNU/Linux for PPC along with

OpenBSD, FreeBSD, and Mac OS X Jaguar.

Easily build your

own multiprotocol

PBX on Linux!

Source: www.asterisk.org


4/24

4

Asterisk provides a central switching core, with four APIs for modular loa-ding of telephony applications, hardware interfaces, file format handling,and codecs. It allows for transparent switching between all supported inter-faces, allowing it to tie together a diverse mixture of telephony systems intoa single switching network.

Anatomy of an Asterisk PBX

The transition from circuit-switchedPBX systems to server-based sy-stems is possible because indivi-dual components are now modular,conformable and faster. Computer-

telephony integration means thatthe same network of servers canprocess both data and voice – and the nature of Asterisk‘s modular designallows for extensive customization. For instance, dial plans for extensionscan be configured to route traf fic to either digital or analogue endpoints,including phones.

Wildcard PCI hardware from Digium provides access to the telephone com-pany and to analogue endpoints such as traditional phones and networked

devices such as the facsimile machine or printer. To protect customer in-vestment and allow incremental migration, Digium provides PCI cards thatinterface with both analogue and digital phones.

Asterisk also complements existing PBX systems by avoiding costly re-placement of existing PBX investment while eliminating further vendor lock-in. In this fashion, Asterisk delivers hybrid PBX capability for purposes fromsimple additional lines to special features like branch-to-branch calling. Ful-ly compatible with a wide range of IP and analogue protocols and codecs,Asterisk translates between them on the fly.

An Asterisk-based solution can support larger numbers of endpoints andservices with little or no expense of adding additional PBX hardware. Thekey is having a Linux-capable administrator or service provider available tomanage the computer telephony needs of the business. But a competentadministrator or service provider working on the LAN or business applica-tions has the same set of skills.

Source: www.riseforth.com


5/24

5

There are also companies (e.g. LocaNet oHG) that offer Asterisk system in-tegration, administration consoles, and software packaging, support plans,and training.

A possible Asterisk PBX is illustrated below. The Asterisk application runson the Linux server. IP phones connect directly to the LAN router, both in-ternally and over the internet. Normal analogue or digital phones connect tointerface cards in the Linux server. Asterisk can provide an outside line or aline to a PBX via different interfaces, both digitally or analogue.

The PBX components shown can typically be purchased for about € 2500,which is probably one-tenth the cost of other commercial alternatives. Ac-tual performance of an Asterisk-based solution depends on traf fic. CPUutilization will quickly increase when multiple calls or codec transcodingsare in progress, placing larger demands on the server.



6/24

6

Asterisk offers a large set of capabilities. In addition to traditional PBX fea-tures such as voice mail, call conferencing, automatic call distribution andinteractive voice response, Asterisk delivers caller ID, call queuing, bridgingand toll-bypass. For example, intelligent routing features can automaticallyredirect calls, making service representatives more accessible, responsiveand productive – a win for both the company and the caller.

Asterisk provides video call support, message-waiting indicators, call par-king and transfer, paging, and intercom, among others features. It includesover 500 recorded professional voice prompts. Incoming voice mails arerecorded in Wave format for storage or forwarding to a user mailbox, orforwarding by email for playback on a personal computer.

Supported VoIP Protocols

Asterisk supports three VoIP protocols, two industry standards and one ori-ginally developed specifically for Asterisk and adopted by number of otherhardware and software devices.

Inter-Asterisk Exchange (IAX): IAX is the de facto standard VoIP protocolfor Asterisk networking. IAX differentiates itself through transparent inter-operation with NAT and PAT (IP masquerade) firewalls. This allows plug-

and-play portability of PBXs and phones. IAX is extremely low-overhead(four bytes of header, as compared to at least 12 bytes of header for RTPbased protocols like SIP and H.323).

Session Initiation Protocol (SIP): SIP is the IETF standard for VoIP. Itscall control syntax resembles SMTP, HTTP, FTP and other IETF proto-cols. SIP is widely regarded as the replacement standard for H.323 VoIPdue to its relative simplicity and human-readability.

H.323: H.323 is the ITU standard for VoIP.

*

*

*



7/24

7

Important advantages of Asterisk

The Asterisk software package delivers a number of unparalleled advan-tages:

Substantial cost reduction - Combined with low-cost PCI-Card telephonyhardware and a Linux PC server, Asterisk can be used to create a PBX ata fraction of the price of traditional PBX and key systems, while providing alevel of functionality exceeding that of many of the most expensive systemsavailable.

Control - Asterisk allows the user to take control of their phone system.Once a call is in a Linux server with Asterisk, anything can be done to it. In

the same way that Apache gives the userfi

ne-grained control over virtuallyevery aspect of its operation (and configuration), the same applies to As-terisk.

Rapid deployment and development - Asterisk allows PBX‘s and interactivevoice response (IVR) applications to be rapidly created and deployed. Itspowerful command line interface (CLI) and text configuration files facilitateboth rapid configuration and real-time diagnostics.

Customization - Through its support for internationalization, configurationfiles, and open source code, every aspect of Asterisk can be configured ormodified. For example, codes for call features can be modified in order toaccommodate proprietary protocols.

Dynamic content deployment - As web servers like Apache allow a devel-oper to deploy dynamic content on the web, such as account information,Asterisk permits such dynamic content deployment over the telephone,with programming ease similar to the Common Gateway Interface (CGI), a

vitally important web technology.

Flexible dial plan - Asterisk‘s unusually flexible dial plan allows seamless in-tegration of IVR and PBX functionality. Many of Asterisk’s existing featurescan be implemented simply with extension logic.



8/24

8

Myths Versus Reality

The transition to IP-based systems like Asterisk is unstoppable, but somemyths persist. In reality, enterprise adoption is pushing aside the myths,technology is outpacing concerns, and first movers are demonstrating solidbenefits.

Myth Reality Internal Use Strategy

Existing infrastructurecannot support the QoSstandards necessary forreal-time voice commu-nications.

Ethernet switches sup-port far more ef ficientvoice traf fic than traditi-onal dedicated telepho-ny, reducing operationalcost.

Server and LAN infrastructure invest-ment at most companies has largelyoutpaced traditional circuit switchedvoice communications systems, ma-king this much newer than existingPBX systems. Opportunity exists for

IP telephony to leverage these plat-forms.

The fundamental con-cern for IP telephony isvoice quality.

Conversation degradesonly when one-way de-lay exceeds 150 ms

About half of normal voice interactionis silence. This means that 50% of thecapacity of the traditional TDM net-work remains unused due to silencealone.

The traditional PBX hashigh reliability throughstable components and

built in manufacturer re-dundancy.

The traditional PBXdeliver zero vendor re-dundancy and is infle-

xible and prohibitive toextend.

Eliminating the separate telecommu-nications network, and instead mana-ging PBX servers in the same LAN

cabinets and on the same operatingsystems as the data network, createssignificant advantages in reliability,maintenance, and ease of admini-stration.



9/24

9

Asterisk™ Features

Asterisk-based telephony solutions offer a rich and flexible feature set. As-terisk offers both classical PBX functionality and advanced features, and in-

teroperates with traditional standards-based telephony systems and Voiceover IP systems.

Call featuresADSI On-Screen Menu SystemAlarm Receiver Append MessageAuthenticationAutomated AttendantBlacklistsBlind Transfer

Call Detail RecordsCall Forward on BusyCall Forward on No Answer Call Forward VariableCall MonitoringCall ParkingCall QueuingCall RecordingCall RetrievalCall Routing (DID & ANI)Call Snooping

Call Transfer Call WaitingCaller IDCaller ID BlockingCaller ID on Call WaitingCalling CardsConference BridgingDatabase Store / RetrieveDatabase IntegrationDial by NameDirect Inward System Access

Distinctive RingDistributed Universal Number Discovery(DUNDi™)Do Not DisturbE911ENUMFax Transmit and Receive (3rd Party OSSPackage)Flexible Extension LogicInteractive Directory ListingInteractive Voice Response (IVR)

*******

***********

***********

**

****

***

Local and Remote Call AgentsMacrosMusic On HoldMusic On Transfer

Flexible Mp3-based SystemRandom or Linear PlayVolume Control

Predictive Dialler PrivacyOpen Settlement Protocol (OSP)Overhead PagingProtocol ConversionRemote Call PickupRemote Of fice SupportRoaming ExtensionsRoute by Caller IDSMS MessagingSpell / Say

Streaming Media AccessSupervised Transfer Talk DetectionText-to-Speech (via Festival)Three-way CallingTime and DateTranscodingTrunkingVoIP GatewaysVoicemail

Visual Indicator for Message Waiting

Stutter Dial tone for Message WaitingVoicemail to emailVoicemail GroupsWeb Voicemail Interface

Zapateller

****◦

◦

◦

***********

**********◦

◦◦

◦

◦

*



10/24

10

Computer-Telephony IntegrationAGI (Asterisk Gateway Interface)Graphical Call Manager Outbound Call SpoolingPredictive Dialler TCP/IP Management Interface

ScalabilityTDMoE (Time Division Multiplex overEthernet)

Allows direct connection of Asterisk PBXZero latencyUses commodity Ethernet hardware

Voice-over IPAllows for integration of physicallyseparate installations

Uses commonly deployed dataconnectionsAllows a unified dialplan across multipleof fices

CodecsADPCMG.711 (A-Law & μ-Law)G.723.1 (pass through)G.726G.729 (through purchase of commerciallicense through Digium)GSMiLBCLinear LPC-10Speex

*****

*

◦

◦

◦

*◦

◦

◦

*****

*****

ProtocolsIAX™ (Inter-Asterisk Exchange)H.323SIP (Session Initiation Protocol)MGCP (Media Gateway Control Protocol)SCCP (Cisco® Skinny®)

Traditional Telephony

InteroperabilityE&ME&M WinkFeature Group DFXSFXOGR-303Loopstart

GroundstartKewlstartMF and DTMF supportRobbed-bit Signaling (RBS) Types

PRI Protocols4ESSBRI (ISDN4Linux)DMS100EuroISDN

Lucent 5ENational ISDN2NFAS

*****

*******

****

****

***



11/24

11

Asterisk™ Architecture

Asterisk is carefully designed for maximum flexibility. Specific APIs are de-fined around a central PBX core system. This advanced core handles the

internal interconnection of the PBX, cleanly abstracted from the specifi

cprotocols, codecs, and hardware interfaces from the telephony applica-tions. This allows Asterisk to use any suitable hardware and technologyavailable now or in the future to perform its essential functions, connectinghardware and applications.

The Asterisk core handles these items internally:

PBX Switching - The essence of Asterisk, of course, is a Private Branch

Exchange Switching system, connecting calls together between varioususers and automated tasks. The Switching Core transparently connectscallers arriving on various hardware and software interfaces.

Applicat ion Launcher - launches applications which perform servicesfor uses, such as voicemail, file playback, and directory listing.

Codec Translator - uses codec modules for the encoding and decodingof various audio compression formats used in the telephony industry. A

number of codecs are available to suit diverse needs and arrive at thebest balance between audio quality and bandwidth usage.

Scheduler and I/O Manager - handles low-level task scheduling andsystem management for optimal performance under all load conditions.

*

*

*

*



12/24

12

Loadable Module APIs

Four APIs are defined for loadable modules, facilitating hardware and pro-tocol abstraction. Using this loadable module system, the Asterisk coredoes not have to worry about details of how a caller is connecting, whatcodecs are in use, etc.

Channel API - the channel API handles the type of connection a caller isarriving on, be it a VoIP connection, ISDN, PRI, Robbed bit signalling, orsome other technology. Dynamic modules are loaded to handle the lowerlayer details of these connections.

Applicat ion API - the application API allows for various task modules

to be run to perform various functions. Conferencing, Paging, DirectoryListing. Voicemail, In-line data transmission, and any other task whicha PBX system might perform now or in the future are handled by theseseparate modules.

Codec Translator API - loads codec modules to support various audioencoding and decoding formats such as GSM, Mu-Law, A-law, and evenMP3.

*

*

*



13/24

13

File Format API - handles the reading and writing of various file formatsfor the storage of data in the file system.

Using these APIs Asterisk achieves a complete abstraction between itscore functions as a PBX server system and the varied technologies existing(or in development) in the telephony arena. The modular form is what al-lows Asterisk to seamlessly integrate both currently implemented telephonyswitching hardware and the growing Packet Voice technologies emergingtoday. The ability to load codec modules allows Asterisk to support both theextremely compact codecs necessary for Packet Voice over slow connec-tions such as a telephone modem while still providing high audio qualityover less constricted connections.

The application API provides for flexible use of application modules to per-form any function flexibly on demand, and allows for open development ofnew applications to suit unique needs and situations. In addition, loadingall applications as modules allows for a flexible system, allowing the admi-nistrator to design the best suited path for callers on the PBX system andmodify call paths to suit the changing communication needs of a goingconcern.

*



14/24

14

Asterisk™ Hardware

The Wildcard TDM400P is a half-length PCI 2.2-compliant card that supports FXS and FXO station

interfaces for connecting analogue telephones and

analogue POTS lines through a PC. Using Digium’sTDM hardware, Open Source Asterisk PBX software,and a standard PC, users can create a Small Of ficeHome Of fice (SOHO) telephony environment whichincludes all the sophisticated features of a high-endPBX/Voicemail platform.

The TDM400P takes the place of an expensive channel

bank and brings the system price point to the lowest inthe industry. The FXO and FXS modules are interchan-geable to create various combinations of interfaces. Toscale this solution, just add additional TDM400P cards.This revolutionary solution has an unprecedented pricepoint in the industry.

Target ApplicationsSmall Of fice Home Of fice (SOHO)applicationsGateway Termination to AnalogTelephonesAdd Inexpensive Analog Phones toExisting PBXsWireless Point-to-Point Applicationsbetween Asterisk Servers

Services & FeaturesCaller ID and Call Waiting Caller ID

ADSI TelephonesPCI Half-length SlotRJ-11C Connector

Standard Confi gurationsTDM10B: 1-port FXS bundleTDM40B: 4-port FXS bundleTDM01B: 1-port FXO bundleTDM04B: 4-port FXO bundleTDM11B: 1-port FXS &1-port FXObundleTDM22B: 2-port FXS &2-port FXO

bundleTDM31B: 3-port FXS &1-port FXObundleOther configurations available onrequest

•

•

•

•

•

•••

•••••

•

•

•

Digium TDM400PScalable and Effective SOHO Solution

Source: www.digium.com


15/24

15

The Wildcard TDM2400P is a full-length 32-bit

33MHz PCI 2.2-compliant card that supports quad

FXS station and quad FXO of fi ce interfaces (for a

total of 24 maximum channels on a single PCI slot )for connecting analog telephones and lines through

a PC. The TDM2400P can be used to connect standardanalog telephone lines to a PC and Asterisk.

Using Digium’s TDM hardware, Asterisk PBX software,and standard PCs, users can create an SME or SOHOtelephony environment that includes all the sophistica-ted features of a high-end PBX/Voicemail platform.

Using an industry-standard bursting, bus-masteringPCI interface chip, and Digium’s patent-pending Voi-ceBus technology, the TDM2400P eliminates the re-quirement for separate channel bank and T1 interfacecards, at an industry-leading price. The quad-FXO andquad-FXS modules are interchangeable, allowing thecreation of any combination of interfaces. The optionalhardware echo cancellation module provides 256 tapsof echo cancellation for superior echo cancellation onboth FXO and FXS interfaces. Scaling of this solution isaccomplished by adding additional TDM2400P cards.

Target ApplicationsChannel Bank Replacement/AlternativeSmall Of fice Home Of fice (SOHO)

applicationsSmall and Medium Enterprise (SME)applicationsGateway Termination to analog tele-phones and lines

Services and FeaturesCaller ID and Call Waiting Caller IDADSI TelephonesPCI Full-length SlotRJ-21 Connector

Standard Confi gurationsSix slots for 4-port FXS or FXOmodulesHardware Echo cancellation moduleavailable

ModulesS400M: Quad FXS ModuleX400M: Quad FXO ModuleVPM100MG.168 Echo Can. Module, 32ms Tail /256 Taps, 1 per TDM2400P

Hardware and Software Re-

quirements1.4 GHz Pentium 4 or better with 64MBRAMAvailable Full-length PCI Slot

••

•

•

••••

•

•

••••

•

•

Digium TDM2400PScalable and Effective SME Solution



16/24

16

The Wildcard TE110P brings a high-performance,

cost-effective, and fl exible single span togglable

T1/PRI or E1/PRI interface to the Digium line-up

of telephony interface devices. The TE110P is acompact and powerful interface supporting voice anddata transmission over T1, E1, and Primary Rate ISDN(PRI) connections.

The TE110P supports industry-standard telephony anddata protocols, including Robbed Bit Signalling (RBS),GR-303, and PRI protocols including NFAS (Non-Faci-lity Association Signalling) for voice and Cisco HDLC,PPP and Frame Relay for data transmission. A low

profi

le, half-length PCI form factor allows this device tofit within a 2U rack mount case or equivalent chassis,offer excellent density for call center, service provider,and other space-sensitive applications.

Of course, the TE110P is fully supported by Digium’sopen source Asterisk PBX software. Used in conjunc-tion with Asterisk, the TE110P offers the power to cre-ate a seamless network, interconnecting traditionaltelephony systems with the emerging Voice-over-IPtechnologies. The TE110P can be used to deliver a

wide range of PBX and IVR services to the network orhandset including Voicemail, Call Conferencing, Three-Way Calling, and VoIP Gateways.

Target ApplicationsPacket Voice Gateways and SwitchesCalling Card Services

One Number ServicesMessage ServicesConferencingCustomized and Web TelephonyVoice/Data IntegrationFutureproof PBXISDN Remote Access Servers

PRI Switch CapabilityAT&T 4ESSDMS 100Lucent 5E

National ISDN 2Euro ISDNNetwork or CPENFAS

RBS Voice ModesGR-303A-Law, μ-Law, and Linear ModesSupportedE&ME&M WinkFeature Group D

Groundstart (FXO and FXS)Loopstart (FXO and FXS) withOptional Disconnect Supervision

Data ModesSyncPPP (both Fixed and Dialup)Frame RelayCisco HDLC

••

•••••••

•••

••••

••

•••

•••

•••

Digium Wildcard TE110PHigh-performance and Cost-effect ive T1 and E1



17/24

17

The TE210P is the next generation of Digium hard-

ware that improves performance and scalability

through bus mastering architecture. The TE210P

supports both E1 and T1 environments and is selecta-ble on a per-card or per-port basis. This feature enablessignaling translation between E1 and T1 equipment andallows inexpensive T1 channel banks to connect withE1 circuits. Because the TE210P improves I/O speedover slave-only architecture, the result is reduced CPUusage and increased card density per server.

Digium has designed the TE210P to be fully compatiblewith existing software applications and it is fully integra-

ted with the Asterisk Open Source PBX/IVR platform.Also, the open source driver supports an API interfacefor custom application development. With the combi-nation of Digium Hardware and Asterisk software, nu-merous combinations of telephony configurations arepossible. From the traditional PBX to VoIP Gateways,Digium solutions are paving the way for a new genera-tion of worldwide communications.

The TE210P supports industry standard telephony anddata protocols, including Primary Rate ISDN (both N.

American and StandardEuro) protocol families forvoice, PPP, Cisco, HDLC,and Frame Relay datamodes. Both line-side andtrunk-side interfaces aresup-ported, also includedare advanced call features.

Target ApplicationsLegacy PBX/IVR servicesVoice-over Internet Protocol (VoIP)

servicesComplex IVR Trees“Meet-me-Bridge” conferencingVoIP Gateways (supports SIP,H.323 and IAX)Calling Card PlatformsVoice/Data Router (replace expensiverouters)

PRI Switch CompatibilityEuroISDN (PRI or PRA) - Q.931/Q.921AT&T 4ESS

DMS 100Lucent 5ENetwork or CPENational ISDN 2

CAS Voice ModesFeature Group DE&M WinkA-Law, Mu-Law, and Linear ModesSupported

Data Modes

SyncPPP (both Fixed and Dialup)Frame RelayCisco HDLCMulti-link PPP

The TE210P is for use only with a 3.3 VoltPCI slot. For use with 5.0 Volt PCI slots,the otherwise feature-identical TE205Pshould be considered.

••

••••••

••

••••

•••

••••

Digium Wildcard TE210PHigh Density and Performance in both T1 and E1



18/24

18


ware that improves performance and scalability

through bus mastering architecture. The TE410P

supports both E1 and T1 environments and is selecta-ble on a per-card or per-port basis. This feature enablessignaling translation between E1 and T1 equipmentand allows inexpensive T1 channel banks to connectwith E1 circuits. Because the TE410P improves I/Ospeed by up to 10 times, the result is reduced CPUusage and increased card density per server.

Digium has designed the TE410P to be fully compatiblewith existing software applications and it is fully integra-

ted with the Asterisk Open Source PBX/IVR platform.Also, the open source driver supports an API interfacefor custom application development. With the combi-nation of Digium Hardware and Asterisk software, nu-merous combinations of telephony configurations arepossible. From the traditional PBX to VoIP Gateways,Digium solutions are paving the way for a new genera-tion of worldwide communications.

The TE410P supports industry standard telephony anddata protocols, including Primary Rate ISDN (both N.

American and Standard Euro)protocol families for voice, PPP,Cisco, HDLC, and Frame Relaydata modes. Both line-side andtrunk-side interfaces are sup-ported, also included are advan-ced call features.


servicesComplex IVR Trees„Meet-me-Bridge“ conferencingVoIP Gateways (supports SIP, H.323and IAX)Calling Card PlatformsVoice/Data Router (replace expensiverouters)

PRI Switch CapabilityEuroISDN (PRI or PRA) - Q.931/Q.921AT&T 4ESS



Data Modes

SyncPPP (both Fixed and Dialup)Frame RelayCisco HDLCMulti-Link PPP


••

•••

••

••

••••

•••

••••

Digium Wildcard TE410PUltimate Densi ty and Performance in both T1 and E1



19/24

19


ware that offers an onboard echo cancellation mo-

dule. It supports E1, T1 and J1 environments and is

selectable on a per-card or per-port basis.

The echo cancellation module supports all T1, E1 andJ1 channels and improves voice quality in environmentswhere software echo cancellation is not suf ficient. TheTE411P reduces CPU overhead required for softwareecho cancellation thereby, freeing resources for otherprocesses, such as codec translation. By supporting16ms with 128 channels or 64ms on 32 channels thiscard will perform in the most dif ficult of environments.

Digium has designed the TE411P to be fully compatiblewith existing software applications and it is fully integra-ted with the Asterisk Open Source PBX/IVR platform.Also, the open source driver supports an API interfacefor custom application development. With the combi-nation of Digium Hardware and Asterisk software, nu-merous combinations of telephony configurations arepossible. From the traditional PBX to VoIP Gateways,Digium solutions are paving the way for a new genera-tion of worldwide communications.

The TE411P supports industry standard telephonyand data protocols, including Primary Rate ISDN (both

N. American and StandardEuro) protocol families forvoice, PPP, Cisco, HDLC,and Frame Relay datamodes. Both line-side andtrunkside interfaces are sup-ported, also included are ad-vanced call features.


servicesComplex IVR Trees„Meet-me-Bridge“ conferencingVoIP Gateways (supports SIP, H.323and IAX)Calling Card PlatformsVoice/Data Router (replace expensiverouters)

PRI Switch CapabilityEuroISDN (PRI or PRA) - Q.931/Q.921AT&T 4ESS



Data Modes

SyncPPP (both Fixed and Dialup)Frame RelayCisco HDLCMulti-Link PPP


••

•••

••

••

••••

•••

••••

Digium Wildcard TE411PUltimate Density and Performance with on-board echo cancellation



20/24

20

The BN4S0 is an ISDN card wi th four S/T interfaces

for Basic Rate Interface (BRI) based on Cologne ChipHFC-4S. It is designed for all kind of data applications

and voice applications such as PC based PABX, VoIPgateways and ISDN monitoring/recording (possiblewith another card version with a Cologne Chip HFC-8Sand different component insertion).

ISDN ConformityThe ISDN BN4S0 board family is compliant to thehardware specifications of the ISDN standards (I.430,CTR3). The BN4S0 board is certified by an authorizedtest laboratory (TÜV Rheinland Product Safety). A copy

of the Layer 1 Measurement Report can be obtainedupon request.

S/T Interface4 BRI portseach port can be configured separately to TE/NTmodeTE/NT mode setting can be detected by software foreach portall ports of the card can be fed independently byone external power supply unit (permitted in NT

mode only; unit optional available)short circuit protection for the ISDN cabling by fuses(no blowing, auto-recovery)line termination (100 Ω) is independently selectablefor each port by DIP switch

PCI InterfacePCI interface is suitable for 3.3V as well as 5V PCI2.2 slots (5V to 3.3V regulator on board)

••

•

•

•

•

•

PCM Busdaisy chaining possible by twoconnectors (2x10 pins, 2.54cm pinpitch) on the cardflat ribbon cable for connection ofseveral cards optional available2/4/8 Mbit/s data transfer rateReport can be obtained uponrequest.

ChipsetCologne Chip HFC-4S ISDNhigh precision 49.152 MHz quartzoscillator 512 bit x 8 serial EEPROM forthe storing of PCI configurationinformation (e.g. PCI vendor ordevice ID)512k x 8 SRAM for enlarging FIFObuffer (soldering option)

General Purpose I/O4 dual LEDs at the slot bracketthat can be controlled by software

with possible states green, red,off for each LED (controllable viaGPIO – general purpose input-output)3-DIP switch can be checked bysoftware for e.g. identification ofthe card (controllable via GeneralPurpose Input - GPI)

•

•

••

••

•

•

•

•

BeroNet BN4S0 ISDN Boardwith 4 S/T Interfaces

Source: www.beronet.com


21/24

21

The BN8S0 is an ISDN card with eight S/T inter-

faces for Basic Rate Interface (BRI) based on Co-logne Chip HFC-8S. It is designed for all kind of data

applications and voice applications such as PC basedPABX or VoIP gateways. The RJ45 jacks have a non-standard pinning (2 ISDN ports per connector). Thereare 4 transformers on the backside of the PCB so thatthe standard PCI dimensions are exceeded.

ISDN ConformityThe ISDN board family is compliant to the hardwarespecifications of the ISDN standards (I.430, CTR3).

S/T Interface8 S/T interfaces: 2 ISDN ports per RJ45 jack. Theinner 4 pins belong to an ISDN port (accessible witha standard ISDN cable) while the outer pins buildan additional ISDN port (special cable or adapterrequired). So the cabling has to be split by anadapter to give access to all 8 S/T interfaces.each port can be configured separately to TE/NTmodeall ports of the card can be fed independently byone external power supply unit (permitted in NT

mode only; unit optional available)short circuit protection for the ISDN cabling by fuses(nonblowing, auto-recovery)line termination (100 Ω) is independently selectablefor each port by DIP switch

PCI InterfacePCI interface is suitable for 3.3V as well as 5V PCI2.2 slots (5V to 3.3V regulator on board)

•

•

•

•

•

•

PCM Busdaisy chaining possible by twoconnectors (2x10 pins, 2.54cm pinpitch) on the cardflat ribbon cable for connection ofseveral cards optional available2/4/8 Mbit/s data transfer rate

ChipsetCologne Chip HFC-8S ISDNhigh precision 49.152 MHz quartzoscillator 512 bit x 8 serial EEPROM forthe storing of PCI configurationinformation (e.g. PCI vendor ordevice ID)

•

•

•

••

•

BeroNet BN8S0 ISDN Boardwith 8 S/T Interfaces

Source: www.beronet.com


22/24

22

OverviewSirrix.PCI4S0 is a PCI-based ISDN interface card. It provides 4 BRI (Basic Rate Interface)ports that can be connected directly to both the public ISDN network as well as to ISDN end-user devices like telephones and fax machines. Sirrix.PCI4S0 is full ISDN compatible andthus, it can be employed for voice applications as well as for data communications.

Hardware StructureThe BRI ports are provided by four Infineon PSB 21150 (ISAC-X) interface circuits. There is a32 KByte SRAM on-card that can be used for buffering the B-Channel data between hardwareand the software application. The communication with the software is realized by a FPGAusing the PCI bus.

The optionally available crypto mechanisms are realized in hardware and included within theFPGA. They allow a zero-delay encryption and decryption of the B-Channels.

BRI Interface LinesAll 4 BRI lines of Sirrix.PCI4S0 can be configurated independently to work in NT mode (Net-work Termination) or TE mode (Terminal Equipment). In both modes they can be operatedas PtMP (Point-to-MultiPoint) or PtP (Point-to-Point). Moreover, they support MSNs (MultipleSubscriber Number) and DDI (Direct Dialing In) in all configurations.

Integrated PCM BusUsing the Sirrix.PCM bus one can link up several Sirrix.PCI4S0 cards, building a fully hard-ware-based ISDN switching network. This provides a zero-delay perfect-quality voice and datatransmission.

Power SupplySirrix.PCI4S0 can be connected with an internal 40V power supply (Sirrix.PS40V-A). The po-wer is transformed from the computer‘s internal 12V power supply. Every line can be selectedto be supplied by jumper configuration. Moreover, every line can be terminated (100 Ohmresistor for BRI line termination) also by using jumpers.

Optical DisplayThe activation of the BRI lines are indicated on 4 LEDs. An additional LED indicates the statusof the FPGA while another LED is free programmable.

ISDN Facil ities SupportAlmost all ISDN facilities are supported by Sirrix.PCI4S0. This encloses e.g. CLIP (CallingLine Identification), CF (Call Forwarding), CW (Call Waiting), Call Hold and Transfer, Adviceof Charge (AOC) and AOC During Call (AOCD), 3PTY (ThreeParty Conference) etc. These facilities are supportedby our comprehensive ISDN stack imple-mentation. They can be used in PBXapplications to provide the correspon-ding facilities.

Sirrix.PCI4S0PCI-based ISDN interface card with 4 BRI Ports

Source: www.sirrix-ag.de


23/24

23

The S101I, affectionately known as the IAXy, takes

Aster isk from the PC to the CPE. The IAXy providesa single, fully featured FXS interface, with an Ethernet

back-end, speaking the Asterisk-native IAX protocol, ata highly competitive volume price. The IAXy is aimedat Voice-Over Broadband and Internet Telephone Ser-vice Providers. IAXy provisioning, firmware updating,and feature activation can be performed entirely withinAsterisk.

The IAX protocol provides complete NAT transparency,enabling full operation behind NAT and PAT firewalls.This includes the ability to robustly transfer calls bet-

ween endpoints, allowing on-net calls to be moved offof a service provider’s network for better quality andlower cost.

Target ApplicationsInternet Telephony ServiceProvider Remote PBX ExtensionsWireless Phone Service withExternal Bridge

FeaturesAuto UpgradeRemote ReprovisioningCaller IDCall WaitingCancel Call Waiting (*70)Caller ID on Call WaitingCaller ID Disable, Enable (*67,*82)Three-way CallingCall Transfer Blind Transfer Call ParkingVMWI (Voice Mail WaitingIndicator)Mute Rx on-HookPulse Dial

Call HoldVoIP Codecslaw (G.711)ADPCM

VoIP ProtocolInter-Asterisk eXchange (IAX)

TelephoneConnector: RJ11

Ringer Equivalence Number(REN): 5 at 1500 ft.

•

••

•••••••

•••••

••

•

••

•

•

•

Digium S101I („ IAXy“ )Taking Asterisk to the Customer Premises



24/24

LocaNet oHG

Lindemannstr. 8144137 Dortmund

Tel. +49 231 91596-66Fax +49 231 91596-55

http://[email protected]

Names and brands may be claimed as the property of others. Information regarding products is provided solely for educatio-nal purposes. LocaNet oHG is not responsible for the performance or support of any products and does not make any repre-sentations or warranties whatsoever regarding quality, reliability, functionality, or compatibility of these devices or products.

Date post:	01-Jun-2018
Category:	Documents
Upload:	anthony-cascante
View:	219 times
Download:	0 times

Asterisk Brochure Locanet

Documents