David Uzzell
2004-Dec-12  08:39 UTC
[Asterisk-Users] Totally LOST with dialplan and Extensions.
Ok I have spent the last week working on getting my small PBX to work.
I will in the end only have 4 SIP extensions being either softphones of 
IP phones. Currently only 1 SIP config for testing.
And at the this point it should be all fairly easy with all inbound and 
outbound to PSTN will be going Via Firefly/Freshtel.net in Australia via 
IAX. Inbound does work in it's current basic state.
There will be NO ZAP devices, so I have ztdummy running.
I would say that for the outbound dialing I have either missed out 
something plainly obvious or a simple typo which would be the challange.
I would think that all the problems are in the extensions.conf file 
which really has me confused and totally lost.
I don't expect answers just pointers in the correct direction so that I 
can get it to work for the outbound calling to work, I have the inbound 
working which was a task but I was able with some pointers to get it 
working.
I would like to thank you all for your casting experianced eyes to look 
over this. What ever is worked out I will make sure the info gets onto 
the Wiki for Freshtel and for a SIP to IAX to PSTN config so that others 
can look up the basic configs to do this type of setup. There does not 
seem to be from what I can find this basic configs for IAX without FXS & 
FXO devices.
cheers
David
SIP.CONF
[general]
context=default
realm=monitor.diversified.com.au
bindaddr=203.29.98.221
srvlookup=yes
maxexpirey=180
defaultexpirey=160
disallow=all
allow=speex
allow=gsm
allow=ilbc
allow=ulaw
allow=ilbc
[801]
type=friend
regexten=801
username=801
secret=password
callerid=801
host=dynamic
nat=yes
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=speex
allow=ulaw
allow=alaw
IAX.CONF
[general]
tos=lowdelay
jitterbuffer=no
disallow=all
allow=speex
allow=ilbc
allow=gsm
allow=adpcm
allow=alaw
register => 89280250:password@cts-au.freshtel.net
register => 89280250:password@gateway.freshtel.net
[guest]
type=user
context=default
auth=none
;inbound
[firefly]
type=friend
host=cts-au.freshtel.net
context=default
; outbound
; Firefly (Freshtel)
[89280250] ; Firefly
context=89280250
qualify=no
username=89280250
secret=password
auth=md5
type=friend
host=gateway.freshtel.net
EXTENSIONS.CONF
[general]
static=yes
writeprotect=no
[globals]
SpeakingClock=123
[default]
exten => s,1,Wait,1
exten => s,n,Answer
exten => s,n,DigitTimeout,5
exten => s,n,ResponseTimeout,10
exten => s,n,WaitExten
exten => s,n,Dial(SIP/801)
exten => 13,1,DateTime()
exten => 13,2,Wait(1)
exten => 13,3,DateTime()
exten => 13,4,Hangup
exten => t,1,Goto(#,1)
exten => i,1,Playback(invalid)
exten => 600,1,Playback(demo-echotest)
exten => 600,n,Echo
exten => 600,n,Playback(demo-echodone)
exten => 600,n,Goto(s,6)
exten => ${SpeakingClock},1,Wait(1)
exten => ${SpeakingClock},2,setvar(FutureTime=$[${EPOCH} + 10])
exten => ${SpeakingClock},3,Wait,3
exten => ${SpeakingClock},4,SayUnixTime(${FutureTime},,R)
exten => ${SpeakingClock},5,playback(vm-and)
exten => ${SpeakingClock},6,SayUnixTime(${FutureTime},,S)
exten => ${SpeakingClock},7,playback(seconds)
exten => ${SpeakingClock},8,playback(beep)
exten => ${SpeakingClock},9,wait(2)
exten => ${SpeakingClock},10,goto(1)
exten => _394.,1,SetCallderId(89280250)
exten => 
_394.,2,Dial(IAX2/89280250:password@cts-au.freshtel.net/${EXTEN:3},60,r)
[outgoing-firefly-peers]
exten => _62XXXXXXXX,1,Macro(outgoingfirefly,${EXTEN:2},70) ; Firefly
[macro-outgoingfirefly]
exten => s,1,SetCallerID("89280250" <89280250>)
exten => 
s,2,Dial(IAX2/89280250:password@gateway.freshtel.net/${ARG1},${ARG2},r)
exten => s,3,Congestion
[macro-outgoingfreshtel]
exten => s,1,SetCallerID("89280250" <89280250>)
exten => 
s,2,Dial(IAX2/89280250:password@cts-au.freshtel.net/${ARG1},${ARG2},r)
exten => s,3,Congestion
Wilson Pickett
2004-Dec-12  11:39 UTC
[Asterisk-Users] Totally LOST with dialplan and Extensions.
> ; outbound > ; Firefly (Freshtel) > [89280250] ; Firefly > context=89280250Where is this context? If you change it to default, it should work if the rest is right. Otherwise, post what you see as console messages when you try to dial.