Is there an example config for asterisk voip to pstn ?
I have the following scenario
TDM400P with 2xFXO connected to the phone lines + asterisk ----> internet
-----> TDM400P 2x FXS + asterisk !
So far i have managed to Pick -up a call incoming from pstn to the fxs , but
it's not working in the other way !
Configs :
---> sip.conf
Asterisk 1 (FXO)
[general]
context=from-sip ; Default context for incoming calls
port=5060 ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
;srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the
Internet
;tos=184 ; Set IP QoS to either a keyword or numeric
val
;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
;maxexpirey=3600 ; Max length of incoming registration we allow
;defaultexpirey=120 ; Default length of incoming/outoing
registration
notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;videosupport=yes ; Turn on support for SIP video
allow=all
allow=gsm
allow=g723.1
allow=g729
allow=ulaw
musicclass=default ; Sets the default music on hold class for all
SIP calls
; This may also be set for individual
users/peers
language=en ; Default language setting for all users/peers
; This may also be set for individual
users/peers
relaxdtmf=yes ; Relax dtmf handling
; when we're on hold (must be >
rtptimeout)
;trustrpid = no ; If Remote-Party-ID should be trusted
;progressinband=no ; If we should generate in-band ringing always
useragent=Asterisk PBX ; Allows you to change the user agent string
nat=no ; NAT settings
; yes = Always ignore info and assume NAT
; no = Use NAT mode only according to RFC3581
promiscredir=yes
register => asterisk@ip_of_the_other_machine
[21]
type=friend
user=21
fromuser=21
secret=1234
host=dynamic
nat=0
allow=all
[22]
type=friend
user=22
fromuser=22
secret=1234
host=dynamic
nat=0
allow=all
----------------------------------------------------------------------------------------------------------------
extensions.conf
[globals]
[general]
;
static=yes
writeprotect=yes
;
[extensions]
;
;ton de test
;
exten => 11,1,Milliwatt()
exten => 11,2,Hangup
;
; Data si Timp
;
exten => 13,1,DateTime()
exten => 13,2,Wait(1)
exten => 13,3,DateTime()
exten => 13,4,Hangup
;
exten => 21,1,Dial(SIP/21,20)
exten => 21,2,Voicemail(u21)
exten => 21,3,Hangup
;
exten => 22,1,Dial(SIP/22,20) ;
exten => 22,2,Voicemail(u22)
exten => 22,3,Hangup
[linia1-centrala]
exten => s,1,Dial(SIP/21,20)
exten => s,2,Hangup
[linia2-centrala]
exten => s,1,Dial(SIP/22|20)
exten => s,2,Hangup
[default]
include => linia1-centrala
include => linia2-centrala
;
[from-sip]
include => extensions
exten => 0,1,Dial(Zap/g1/${EXTEN},70)
exten => _XX,1,Dial(Zap/g1/${EXTEN},70)
;
--------------------------------------------
the other machine
sip.conf
[general]
context=from-sip ; Default context for incoming calls
port=5060 ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
;srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the
Internet
;tos=184 ; Set IP QoS to either a keyword or numeric
val
;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
;maxexpirey=3600 ; Max length of incoming registration we allow
;defaultexpirey=120 ; Default length of incoming/outoing
registration
notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;videosupport=yes ; Turn on support for SIP video
allow=all
allow=gsm
allow=g723.1
allow=g729
allow=ulaw
musicclass=default ; Sets the default music on hold class for all
SIP calls
; This may also be set for individual
users/peers
language=en ; Default language setting for all users/peers
; This may also be set for individual
users/peers
relaxdtmf=yes ; Relax dtmf handling
; when we're on hold (must be >
rtptimeout)
;trustrpid = no ; If Remote-Party-ID should be trusted
;progressinband=no ; If we should generate in-band ringing always
useragent=Asterisk PBX ; Allows you to change the user agent string
nat=no ; NAT settings
; yes = Always ignore info and assume NAT
; no = Use NAT mode only according to RFC3581
promiscredir=yes
register => 21:xxxx@xxx.xxx.xxx.xxx
register => 22:xxxx@xxx.xxx.xxx.xxx
[sip1-out]
type=friend
secret=1234
username=21
fromuser=21
host=xxxxxxxxxxx
nat=no
canreinvite=no
allow=all
[sip2-out]
type=friend
secret=1234
username=22
fromuser=22
host=xxxxxxxxxxxxx
canreinvite=no
nat=no
allow=all
[asterisk]
user=asterisk
type=friend
host=dynamic
-------------------------------------------------------------------------
extensions.conf
[incoming1]
exten => _2XXXXXXXXXXXX,1,Dial,SIP/${EXTEN:1}@sip1-out|20
exten => _1XX,1,Dial,SIP/${EXTEN:1}@sip1-out,20
exten => _1XX,2,Voicemail,u21
[incoming2]
exten => _1XX,1,Dial,SIP/${EXTEN:1}@sip1-out,20
exten => _1XX,2,Voicemail,u22
[from-sip]
include => tel1
include => tel2
[tel1]
exten => 21,1,Dial(Zap/1|25)
[tel2]
exten => 22,1,Dial(Zap/2|25)
Sincerly yours,
Mike