[asterisk-users] Sharing my Asterisk + SPA3102/PAP2 setup: What I've learned in 1 week.
Rodolfo Alcazar Portillo
rodolfo.alcazar at padep.org.bo
Thu Oct 16 07:31:24 CDT 2008
(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: rodolfo.alcazar at padep.org.bo
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