asterisk-howto.pdf

(114 KB) Pobierz

howto: isdn phone

<–>

sip routing (sipgate / freenet

iphone)

(software: asterisk, mISDN and festival)

Steﬀen Buehl

asterisk@sbuehl.com

v0.82 08.09.2004

the oﬃcal version of this document can be obtained from

<http://www.sbuehl.com/projects/asterisk/>

- pdf version is available

here

<http://www.sbuehl.com/projects/asterisk/asterisk-howto.pdf>

introduction

during a trip to lloret de mar (in april 2004) i read some stuﬀ about ”pbx4linux” in the german computer

magazine c’t. they explained how to connect a standard isdn phone to a linux box (acting as pbx) running

pbx4linux. unfortunately pbx4linux supports only h.323 connections at this time so i decided to use asterisk

to connect my isdn phone to a sip (german provider ”sipgate”) network.

asterisk is a linux pbx software with a nice plugin interface that allow to use a wide spectrum of

endpoints. e.g. you can route between h.323, sipgate, capi (isdn), mISDN (new isdn interface for kernel 2.6)

etc.. there are also some application plugins for asterisk: festival - a linux text2speech engine - makes your

pbx speaking english mp3player - playback of mp3s stored on your harddisk a cgi-like extensions - called

agi (asterisk gateway interface) - allows you to control asterisk with your favourite scripting language (perl,

php, python, ...). i wrote a small phonebook script to lookup the callerid in a mysql table and display a

custom name on my isdn phone.

my conﬁguration routes between mISDN (HFC-S card with attached isdn phone) and sipgate (multi-

ple accounts). by using diﬀerent preﬁxes on the phone i can choose which sipgate account to use, e.g. preﬁx

1 for hamburg (040), preﬁx 2 for berlin etc.

to plug a standard isdn phone in a ISDN card you need a 100ohm resistor, a power source and a

crossed isdn cable (standard crossover ethernet doesn’t work). (see

<http://isdn.jolly.de>

). the easiest

solution for this is to use an NT1 (also called NTBA). it contains the needed resistors and supplies power

to your attached isdn phones, additionally you can wire your cable directly to internal pins - so it’s kinda

perfect for this job.

you can conﬁgure multiple MSNs on your isdn phone for each sip account (i used the sip-id as MSN) and

therefore a ringtone for each id.

Okay - i got a really cheap fritz!pci isdn card (5 eur) to continue this project... The installation was

pretty easy - just plugged that card into my box and connected it to my isdn s0 bus. I conﬁgured asterisk

to dial-out with preﬁx 0, see example conﬁg for more details.

festival isn’t used so far - but it’s easy to extend your conﬁguration with the following command:

<http://www.voip-info.org/wiki-Asterisk+cmd+Festival>

2. installation

okay, enough theory, let’s get started...

2.1

2.1.1

installation

requirements

hardware

NT1 (NTBA)

HFC-S card (see

<http://isdn.jolly.de/cards.html>

ISDN phone

ISDN card (capi/mISDN)

2.1.2

other requirements

- linux (2.6 works, don’t know about 2.4)

2.2

fetching software

asterisk:

<http://www.asterisk.org/>

festival:

<http://www.cstr.ed.ac.uk/projects/festival>

chan misdn:

<http://www.beronet.com/?PageID=3018>

chan capi:

<http://ns1.jnetdns.de/jn/relaunch/asterisk/page1.html>

2.3

compiling

should be easy - just read the docs - maybe i write some documentation for this laster...

2.3.1

2.3.2

2.3.3

2.3.4

2.3.5

asterisk

mISDN

chan misdn

chan capi

festival (text2speech)

<http://www.voip-info.org/wiki-Asterisk+Festival+installation>

with gcc 3.2.2 you have to patch festival and speech tools:

3. conﬁguration

3.1

3.2

3.2.1

conﬁguration

mISDN

asterisk

capi.conf

[general]

nationalprefix=

internationalprefix=0

rxgain=0.8

txgain=0.8

[interfaces]

msn=5

incomingmsn=5

controller=1

softdtmf=0

context=in

devices=2

3.2.2

misdn.conf

[general]

context=misdn

; 0 - No Debug

; 1 - mISDN Messages and * - Messages, and * - State changes

; 2 - Messages + Message specific Informations (e.g. bearer capability)

; 3 - very Verbose, the above + lots of Driver specific infos

; 4 - even more Verbose than 3

debug=1

[intern]

ports=2

context=out

3.2.3

extensions.conf

[general]

static=yes

writeprotect=no

[globals]

PHONE=mISDN/2

ISDNOUT=CAPI/5:b

NULL=

#include "macro_dial-result.inc"

[macro-process-call]

3. conﬁguration

; ${ARG1} Device

; ${ARG1} MSN

exten => s,1,Ringing

exten => s,2,Wait,2

exten => s,3,agi,directory.agi

exten => s,4,Dial(${ARG1}/${ARG2})

exten => s,5,Hangup

[macro-dialout]

; ${ARG1} CIDNAME

; ${ARG2} Device

; ${ARG3} Num

; ${ARG4} SIP EXT

exten => s,1,SetCIDName(${ARG1})

exten => s,2,Dial(${ARG2}${ARG3}${ARG4},,t)

exten => s,3,Playback(invalid)

exten => s,4,Hangup

[sipgate-in]

exten => <sip-id-1>,1,Macro(process-call,${PHONE},<sip-id-1>)

exten => <sip-id-2>,1,Macro(process-call,${PHONE},<sip-id-2>)

exten => <freenet-id>,1,Macro(process-call,${PHONE},1)

[isdn-in]

exten => 5,1,Macro(process-call,${PHONE},5)

[sipgate-out]

exten => _1.,1,Macro(dialout,<Name ID 1>,SIP/,${EXTEN:1},@<sip-extension-1>)

exten => _2.,1,Macro(dialout,<Name ID 2>,SIP/,${EXTEN:1},@<sip-extension-2>)

exten => _3.,1,Macro(dialout,<Name ID 3>,SIP/,${EXTEN:1},@<sip-freenet-extension>)

[isdn-out]

;ISDN OUT

exten => _0.,1,Macro(dialout,<Name>,${ISDNOUT},${EXTEN},${NULL})

;ISDN-INTERN

exten => _8.,1,Macro(dialout,<Name>,${ISDNOUT},${EXTEN:1},${NULL})

[ip-out]

;IP DIAL

exten =>

_9.,1,SetCIDName(<Name>)

_9.,2,AGI(parseExtension.agi)

_9.,3,Dial(SIP/${CNUM}@${CIP},30,tr)

_9.,4,Playback(invalid)

_9.,5,Hangup

[in]

include => sipgate-in

include => isdn-in

exten => t,1,Macro(dial-result)

3. conﬁguration

[out]

include => sipgate-out

include => isdn-out

include => ip-out

exten => s,1,Answer

exten => s,2,DigitTimeout,5

exten => s,3,ResponseTimeout,30

exten => i,1,Macro(dial-result,5)

exten => t,1,Macro(dial-result)

exten => T,2,PlayTones(congestion)

exten => T,3,Wait(5)

exten => T,4,Hangup

3.2.4

modules.conf

[modules]

autoload=yes

load => res_musiconhold.so

load => chan_misdn.so

load => chan_capi.so

noload => pbx_gtkconsole.so

noload => pbx_kdeconsole.so

noload => app_intercom.so

noload => chan_modem_i4l.so

noload => chan_modem_bestdata.so

noload => chan_modem_aopen.so

noload => chan_modem.so;

noload => chan_zap.so

noload => chan_alsa.so

noload => chan_oss.so

noload => app_groupcount.so

noload => chan_agent.so

noload => chan_iax.so

noload => app_txtcidname.so

noload => app_controlplayback.so

[global]

chan_capi.so=yes

3.2.5

sip.conf

[general]

port = 5060

bindaddr = 0.0.0.0

localnet = 192.168.0.0

localmask = 255.255.255.0

maxexpirey=3600

defaultexpirey=300

nat=no

disallow=all

allow=alaw

asterisk-howto.pdf

Plik z chomika:

Inne pliki z tego folderu:

Inne foldery tego chomika: