Chapter 1 Introduction

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Transcription:

Etross-400E Manual Page 1 of 23 Version 1.0

Contents Chapter 1 Introduction... 3 Chapter 2 Hardware introduction... 6 Chapter 3 Intallation and configuration... 8 Chapter 4 Test example... 19 Chapter 5 Reference... 23 Page 2 of 23 Version 1.0

1. What s Etross-400E? Chapter 1 Introduction Etross-400E is a 4 port FXS/FXO analog Asterisk PCI card which is mainly applied to PSTN environment. The PCI plug is based on PCI2.2, It supports all the functions of Asterisk PBX systems needed. User could choose any of combinations of FXO_100 or FXS_100 modules according to their requirements. Both software and hardware compatible with Digium s TDM400P. It is mixable with X100M/S100M of Digium. 2. What s Asterisk? Asterisk is the world's most popular open source telephony project. Under development since 1999, Asterisk is free, open source software that turns an ordinary computer into a feature-rich voice communications server. Asterisk makes it simple to create and deploy a wide range of telephony applications and services. Asterisk is released as open source under the GNU General Public License (GPL), and it is available for download free of charge. Asterisk is the leading open source telephony project and the Asterisk community has been ranked as a key factor in the growth of VoIP. 3. What Does Asterisk Do? Asterisk is like an erector set or a box of Legos for people who want to create communications applications. That's why we refer to it as a "tool-kit" or "development platform". Asterisk includes all the building blocks needed to create a PBX system, an IVR system or virtually any other kind of communications solution. The "blocks" in the kit include: Drivers for various VoIP protocols. Drivers for PSTN interface cards and devices. Routing and call handling for incoming calls. Outbound call generation and routing. Media management functions (record, play, generate tone, etc.). Call detail recording for accounting and billing. Transcoding (conversion from one media format to another). Protocol conversion (conversion from one protocol to another). Database integration for accessing information on relational databases. Page 3 of 23 Version 1.0

Web services integration for accessing data using standard internet protocols. LDAP integration for accessing corporate directory systems. Single and mult-party call bridging. Call recording and monitoring functions. Integrated "Dialplan" scripting language for call processing. External call management in any programming or scripting language through Asterisk Gateway Interface (AGI) Event notification and CTI integration via the Asterisk Manager Interface (AMI). Speech synthesis (aka "text-to-speech") in various languages and dialects using third party engines. Speech recognition in various languages using third party recognition engines. This combination of components allows an integrator or developer to quickly create voice-enabled applications. The open nature of Asterisk means that there is no fixed limit on what it can be made to do. Asterisk integrators have built everything from very small IP PBX systems to massive carrier media servers. 4. Asterisk In Action: Key Applications Asterisk As A PBX Asterisk can be configured as the core of an IP or hybrid PBX, switching calls, managing routes, enabling features, and connecting callers with the outside world over IP, analog (POTS), and digital (T1/E1) connections. Asterisk runs on a wide variety of operating systems including Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris and provides all of the features you would expect from a PBX including many advanced features that are often associated with high end (and high cost) proprietary PBXs. Asterisk's architecture is designed for maximum flexibility and supports Voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk As A Gateway It can also be built out as the heart of a media gateway, bridging the legacy PSTN to the expanding world of IP telephony. Asterisk s modular architecture allows it to convert between a wide range of communications protocols and media codecs. Asterisk as a feature/media server. Page 4 of 23 Version 1.0

Need an IVR? Asterisk s got you covered. How about a conference bridge? Yep. It s in there. What about an automated attendant? Asterisk does that too. How about a replacement for your aging legacy voicemail system? Can do. Unified messaging? No problem. Need a telephony interface for your web site? Ok. Asterisk In The Call Center Asterisk has been adopted by call centers around the world based on its flexibility. Call center and contact center developers have built complete ACD systems based on Asterisk. Asterisk has also added new life to existing call center solutions by adding remote IP agent capabilities, advanced skills-based routing, predictive and bulk dialing, and more. Asterisk In The Public Network Internet Telephony Service Providers (ITSPs), competitive local exchange carriers (CLECS) and even first-tier incumbents have discovered the power of open source communications with Asterisk. Feature servers, hosted services clusters, voicemail systems, pre-paid calling solutions, all based on Asterisk have helped reduce costs and enabled flexibility. Asterisk Everywhere Asterisk has become the basis for thousands of communications solutions. If you need to communicate, Asterisk is your answer. Page 5 of 23 Version 1.0

Chapter 2 Hardware introduction 1. Operation Environment CentOS 5.0 Dahdi-linux-xxx Dahdi-tools-xxx Asterisk-1.6 2. Etross-400E with 2FXS+ 2FXO Etross-400E Picture as following: The green light turn on mean the channel work normally Red module is FXO Green module is FXS RJ11 slot Page 6 of 23 Version 1.0

Clock output plug Etross-400E has a clock output plug to realize the colck synchronization. Through the clock line connect to the plug, Etross-400E can be connected to other cards, like GSM400P. With the colck synchronization, user can plug some cards in one computer. Page 7 of 23 Version 1.0

Chapter 3 Intallation and configuration 1. Plug Etross-400E A. Shut down the PC and remove the power cable. B. Insert Etross-400E card in PCI slot, and fix with the screw. C. Connect the power cable and turn on the PC. 2. Start the CentOs5 system, input lspci -v and click enter. It can diagnose whether Etross-400P is found in the operating system. If the operating system show the information like the red frame in following pictures, that s mean the Etross-400E was found. 3. Check the software packages which Asterisk needed: Input following command to check whether you have installed Asterisk auxiliary software package. rpm -q bison rpm -q bison-devel rpm -q ncurses rpm -q ncurses-devel rpm -q zlib rpm -q zlib-devel rpm -q openssl rpm -q openssl-devel Page 8 of 23 Version 1.0

rpm -q gnutls-devel rpm -q gcc rpm -q gcc-c++ It means that you have not installed Asterisk auxiliary software package if the computer shows the information as following picture. Please input yum install xxx to install the Asterisk auxiliary software package. xxx mean the relation Asterisk auxiliary software package, for example, bison-devel as following picture. 4. Install Zaptel and Asterisk software package Download same version Zaptel and Asterisk software package, propose using version 1.4. A. Download way 1: Download it from website: http://downloads.asterisk.org/pub/telephony/ Find the version of the Zaptel and Asterisk you need. As following picture showed. Page 9 of 23 Version 1.0

The package downloaded is as following: B. Download way 2: Input cd /usr/src in the command line to enter the src directory. Input wget + xxx xxx mean files address Page 10 of 23 Version 1.0

5. Compile Zaptel and Asterisk, follow is the step: A. Decompress the Zaptel and Asterisk software package: cd /usr/src tar xvzf zaptel-xxxx tar xvzf asterisk-xxxx xxxx means the version xxxx means the version For example: B. Compile Zaptel cd zaptel-xxxx./configure Page 11 of 23 Version 1.0

The following picture showes that./configure runs correctly. Input make as following picture The computer will show the information like following pictures if the make command run correctly. Input make install as following picture: Page 12 of 23 Version 1.0

The computer will show the information like following pictures if the make install command run correctly. Input make config as following picture: The computer will show the information like following pictures if the make config command run correctly. C. Compile Asterisk cd /usr/src/asterisk-xxxx xxxx means version./configure Page 13 of 23 Version 1.0

The computer will show the information like following pictures if the./configure command run correctly. Page 14 of 23 Version 1.0

Input make The computer will show the information like following pictures if the make command run correctly. Input make install The computer will show the information like following pictures if the make install command run correctly. Page 15 of 23 Version 1.0

Input make samples The computer will show the information like following pictures if the make samples command run correctly. D. Run two commands to produce automatically two files: cd /usr/src/zaptel-xxxx/kernel/xpps/utils xxxx means version./genzaptelconf sdvm Runs the command as above will produce two files: /etc/zaptel.conf and /etc/asterisk/zapata-channels.conf. zaptel.conf detail information as following: # Span 1: WCTDM/0 "Wildcard S400P Prototype Board 1" (MASTER) fxsks=1 //FXO 口 fxsks=2 //FXO 口 fxoks=3 //FXS 口 fxoks=4 //FXS 口 # Global data loadzone = us defaultzone = us /etc/asterisk/zapata-channels.conf detail information as following: ; Span 1: WCTDM/0 "Wildcard S400P Prototype Board 1" (MASTER) ;;; line="1 WCTDM/0/0" signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel => 1 context=default ;;; line="2 WCTDM/0/1" signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel => 2 context=default Page 16 of 23 Version 1.0

;;; line="3 WCTDM/0/2" signalling=fxo_ks callerid="channel 3" <6003> mailbox=6003 group=5 context=from-internal channel => 3 callerid= mailbox= group= context=default ;;; line="4 WCTDM/0/3" signalling=fxo_ks callerid="channel 4" <6004> mailbox=6004 group=5 context=from-internal channel => 4 callerid= mailbox= group= context=default E. Check whether the two files produced automatically is disposed identically with your Etross-400E. If not identical, please adapt it. Attention: FXS uses FXO signalling, FXO uses FXS signalling. Page 17 of 23 Version 1.0

The software version this manual used will have a problem, but other version will not have. The problem is that /etc/asterisk/zapata-channels.conf doesn t include in /etc/asterisk/zapata.conf. Our solution is to add #include zapata-channels.conf to the last line of zapata.conf F. Run following command: modprobe zaptel modprobe wctdm G. Start Asterisk, and run following command: asterisk vvvvvvvgc zap show channels Channel 1 and channel 2 are FXO, channel 3 and channel 4 are FXS. Page 18 of 23 Version 1.0

1. Test FXS module Chapter 4 Example of texting A. Connect two analog phones to the card by RJ11 slot point out in red in following picture. B. In the command line, run two commands: cd /etc/asterisk vi /extensions.conf Then turn to the last line of extensions.conf, add codes as following: [from-internal] exten => 6005,1,Dial(zap/1) exten => 6005,2,Hangup() exten => 6000,1,Dial(zap/2) exten => 6000,2,Hangup() Page 19 of 23 Version 1.0

The codes above define the dialing rules: Dial 6005 from FXS channel in addition to channel 1 can make the phone which connected to channel 1 ringing, dial 6000 from FXS channel in addition to channel 2 can make the phone which connected to channel 2 ringing. Right now run command asterisk r to enter CLI directory Run reload in CLI directory NOTED: Please reload the asterisk if you adapt the extensions.conf C. Check whether the dialing rules works by using phone dial the number 6005 and 6000. If the dialing is success, the system will show following informations in the CLI directory. And that s mean FXS works normally. *CLI> -- Starting simple switch on 'Zap/2-1' -- Executing [6005@from-internal:1] Dial("Zap/2-1", "zap/1") in new stack -- Called 1 -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Hungup 'Zap/1-1' == Spawn extension (from-internal, 6005, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' Page 20 of 23 Version 1.0

2. Test FXO module: A. Connect PSTN phone line to the card by RJ11 slot point out in red in following picture. Connect a phone to channel 2. B. Run some commands to define the dialing rules: #cd /etc/asterisk #vi extensions.conf Then turn to the last line of extensions.conf, add codes as following: [from-internal] exten => 6005,1,Dial(zap/1) exten => 6005,2,Hangup() exten => 6000,1,Dial(zap/2) exten => 6000,2,Hangup() [from-pstn] exten => s,1,answer() exten => s,2,dial(zap/2) exten => s,3,hangup() The codes above define the dialing rules: The incoming call from PSTN will make the phone connect to channel 1 ringing. Right now run command asterisk r to enter CLI directory Run reload in CLI directory Page 21 of 23 Version 1.0

NOTED: Please reload the asterisk if you adapt the extensions.conf C. Check whether the dialing rules works by using phone or cell phone dial the number of the PSTN which we connect to the Etross-400E. If dial successfully, the phone connect to channel 2 will ringing. -- Starting simple switch on 'Zap/3-1' -- Executing [s@from-pstn:1] Answer("Zap/3-1", "") in new stack -- Executing [s@from-pstn:2] Dial("Zap/3-1", "zap/2") in new stack -- Called 2 -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/3-1 -- Native bridging Zap/3-1 and Zap/2-1 -- Hungup 'Zap/2-1' == Spawn extension (from-pstn, s, 2) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' D. Same way to test other FXO module. Page 22 of 23 Version 1.0

Chapter 5 Reference www.etross.com www.asterisk.org www.digium.com www.voip-info.org www.ctiforum.com Disclaimers: Digium is registered trademark of Digium, Inc Asterisk is registered trademark of Digium, Inc Page 23 of 23 Version 1.0