(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become).
Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to > provide > ASTRISK software... In terms of additional hardware, what do I buy ? Im' a linux geek (many years), but an asterisk newbie (less than a week). Anyway, this is what I've get and done (my setup is fully experimental, just to learn, then scale to a full 35 users/ 8 PSTN lines setup for starting). All of this are my notes. Previous: Definition of FXS, FXO: http://www.3cx.com/PBX/FXS-FXO.html NOTE: I assume no responsability, if you damage one equipment or person or cat. You are working with voltages which get sometimes higher as 100 volts. Dogs know nothing about telephones. I have: * studied a lot. DO THAT. if you find an error, you will not find the cause if you do not know where to look, where to change something, where to disable something. * Someone ponted me to this document, which I started with. Nice to start. http://www.viagenie.ca/publications/2007-03-apricot-asterisk-primer-blanchet.pdf * Fedora 9 (my desktop and the same at home). IP: 192.168.1.141. IPTables disabled. * installed asterisk as a fedora root user: # yum install asterisk asterisk-sounds asterisk-voicemail I bought: * 1 Linksys SPA3102 (1 FXS, 1 FXO) * 1 Linksys PAP2 (2 FXS) * 3 old panasonic analog telephones * 1 telephone line (also called POTS, PSTN) from my city provider. * Connected between them (easier as connecting your microwave oven). Phones go to the FXS/PHONE ports. POTS/PSTN go to the WALL/LINE/FXO port. Ethernet ports go all to a hub. SPA3122 yellow port keeps empty. * Programmed them first with a telephone connected (both Linksys features a menu with voices): Enter voice menu **** Factory Reset 73738# 1 IP Data CHECK SET ----------------------------------- STATICIP 100# 101# 1# 1 WAN IP 110# 111# 192*168*1*223# 1 WAN MASK 120# 121# 255*255*255*0# 1 WEB SERVER 7932# 1# 1 (Hangup: ATA will reboot) (192.168.1.223 is the SPA3102, 192.168.1.222 is the PAP's. All IPs must match your office's) * Once configured, programmed them with a web browser: http://192.168.1.223 WAN Eth PORT ============ ROUTER WAN SETUP Gateway: 192.168.1.1 Primary DNS: 192.168.1.1 FXS Ports: (Example for SPA3102 FXS Line) ========================================= SIP Port: 5060 Proxy: 192.168.1.141 (my own computer) Display Name: 103 User ID: 103 Password: green FXO (Example for SPA3102 FXS PSTN) ================================== SIP Port: 5061 Proxy: 192.168.1.141 Outbound Proxy: 192.168.1.141 Use Outbound Proxy: yes Display Name: 201 User ID: 201 Password: green Dial Plan 8: (S0<:192.168.1.141>) VoIP-To-PSTN Gateway Setup -------------------------- VoIP-To-PSTN Gateway Enable: Yes VoIP Caller Auth Method: HTTP Digest VoIP Users and Passwords (HTTP Authentication) ---------------------------------------------- VoIP User 1 Auth ID: 201 VoIP User 1 Password: green PSTN-To-VoIP Gateway Setup -------------------------- PSTN-To-VoIP Gateway Enable: yes PSTN Ring Thru Line 1: no PSTN Caller Default DP: 8 FXO Timer Values (sec) ---------------------- PSTN Answer Delay: 0 International Control --------------------- Line-In-Use Voltage: 30 (lowered to 25 for testing behind my current Panasonic) * Installed twinkle, a softphone: Konto: Asterisk Registrar: 192.168.1.141:5061 Benutzer: 104 Passwort: green Auth-name: 104 Zeitlimit: 3600 * Ok, now to configure asterisk in the fedora BOX: # vi /etc/asterisk/sip.conf [101]; port 1 FXS on PAP2 type=friend secret=green regexten=101 qualify=1000 nat=no host=dynamic context=padep registertrying=yes mailbox=101 [102]; port 2 FXS on PAP2 type=friend secret=green regexten=102 qualify=1000 nat=no host=dynamic context=padep registertrying=yes mailbox=102 [103]; port 1 FXS on SPA3102 type=friend secret=green regexten=103 qualify=1000 nat=no host=dynamic context=padep registertrying=yes mailbox=103 [104]; my computer's softphone (ekiga, twinkle) type=friend secret=green regexten=104 qualify=1000 nat=no host=dynamic context=padep registertrying=yes mailbox=104 [201]; call from FXS extension --> PSTN type=peer host=dynamic port=5061 secret=green context=padep dtmfmode=rfc2833 canreinvite=no [201]; call from PSTN --> 101 type=user host=dynamic port=5061 secret=green context=padep dtmfmode=rfc2833 * extensions.conf, commented, contains: [padep] ; Everyone of this three groups is for one extension. Means: ; - Connect the call to the extension ; - wait 12 seconds ; - go to voicemail ; - Hangup exten => 101,1,Dial(SIP/101,12,rt) exten => 101,2,Voicemail(101) exten => 101,3,Hangup exten => 102,1,Dial(SIP/102,12,rt) exten => 102,2,Voicemail(102) exten => 102,3,Hangup exten => 103,1,Dial(SIP/103,12,rt) exten => 103,2,Voicemail(103) exten => 103,3,Hangup exten => 104,1,Dial(SIP/104,12,rt) exten => 104,2,Voicemail(104) exten => 104,3,Hangup ;Voicemail ; When the user dials 500, has his voicemail menu exten => 500,1,VoiceMailMain() ; When an incoming call from PSTN-FXO-SPA3102 arrives, redirect to extension 101 ; From PSTN to 101: exten => s,1,Transfer(101) ; When user dials 7#, starts an echo test exten => 7,1,Answer exten => 7,2,Echo() exten => 7,3,Hangup ; to get external PSTN line Dial 9#, authenticate with 1111# exten => 9,1,Answer exten => 9,2,Authenticate(1111) exten => 9,3,Dial(SIP/201) ; A simple test: dial 2# and get a "hello world" answer. I recorded my own voice, and put ; hello-world.wav in the sounds dir exten => 2,1,BackGround(hello-world) exten => 2,2,Hangup ; Some notes ;exten => _91XX!,11,Authenticate(1111); How to authenticate ;exten => _91XX!,20,AGI(agi.bash); How to run an AGI script ;exten => _91XX!,30,GotoIf($["${numero}" = "22"]?4:5); How to return the number from AGI script ;exten => _91XX!,40,Dial(SIP/201/${EXTEN:${GLOBAL(TRUNKMSD)}}); Dial a number like 9125, behind my panasonixPBX ;exten => _91XX!,50,Hangup; this is really complicated. * Example for my AGI script: # cat /usr/share/asterisk/agi-bin/agi.bash #!/bin/bash checkresults() { while read line do case ${line:0:4} in "200 " ) echo "$line" >> /tmp/agi.log return;; * ) echo $line >&2;; esac done } RESULT='SET VARIABLE numero 22' echo "$RESULT" > /tmp/agi.log # former line is my log, to verify AGI runs.... echo "$RESULT" checkresults --------------------------------------------------------- That's all. I studied a lot, since saturday. You must do that, if you find an error, I'm sure you will not solve it, if you didn't studied. > Where do I connect the phones? to PC ? How to connect Linksys with PC ? When you buy the ATAs (linksys), you'll find that easy as connecting your TV. That's simple. Both include full-colored diagrams. > Do you have a visio diagram or powerpoint that you can share with me No. I prefer reading and writing. Read above. Read your equipment manuals. Look for google images of the ATAs. Is really simple. You are worrying without having read anything... Read, google what you don't understand from my mail. Good luck. Ask, if you need something more. -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft für Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gestión Pública Descentralizada y Lucha Contra La Pobreza - PADEP Av. Sánchez Lima 2226 La Paz, Bolivia Tel: +591 22417628 (121) Fax: +591 22417628 (126) Web: www.padep.org.bo Email: [EMAIL PROTECTED]
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