[Asterisk-Users] does aanyone have an example of how to dial outwith a sip phone on a pstn line?

James Bean james at hdcs.com.au
Sat Dec 11 16:17:13 MST 2004


Charles S. Antrim wrote:
 > I am using a card that has an fxo and fxs module. 
 
 I am no where near an expert but I have my sip phone working through my
pstn line and this is my config.
 
/etc/asterisk/sip.conf
[general]
port = 5060
bindaddr = 192.168.69.1
context = sip
disallow = gsm
allow = alaw
disallow = ulaw
nat=disable
srvlookup=no
localnet=192.168.69.0/255.255.255.0
subscribecontext = sip
 
[snom-james]
type=friend
secret=<password>
host=dynamic
callerid="James Bean" <690>
defaultip=192.168.69.250
dtmfmode=rfc2833
mailbox=690
 
[bt-karen]
type=friend
secret=<password>
host=dynamic
callerid="Karen Colomb" <691>
defaultip=192.168.69.251
dtmfmode=info
mailbox=691
 
/etc/asterisk/extensions.conf
[pstn]
 
exten => s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a
comment in the CLI for info.
exten => s,2,SetMusicOnHold(random)
exten => s,3,Dial(SIP/snom-james&SIP/bt-karen,45,t) 
exten => s,4,Hangup
;exten => s,5,VoiceMail(u100)    ;Whatever box you want.
 
[internal]
 
exten => i,1,Playback(invalid)
exten => i,2,Hangup
exten => t,1,Hangup
 
exten => 099,1,Echo     ;simple echo test when you dial 099 on your
phone
 
include => outgoing
include => sip
 
[outgoing]
 
exten => _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten => _9X.,2,Congestion()
exten => _9X.,3,Hangup
 
include => sip
 
[sip]
 
exten => 690,1,SetMusicOnHold(random)
exten => 690,2,Dial(SIP/snom-james,30,tr)
exten => 690,3,voicemail2,u690
exten => 690,102,voicemail2,b690
 
exten => 691,1,SetMusicOnHold(random)
exten => 691,2,Dial(SIP/bt-karen,30,tr)
exten => 691,3,voicemail2,u691
exten => 691,102,voicemail,b691
 
include => internal
include => outgoing

 
[from-sip]
 
include => internal
 
This isn't the best example of how to do it but it works.
 
I hope it helps.
 
James


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