I recently installed asterisk for a demo and want to showcase inbound/outbound PSTN and inbound/outbound VOIP. I’m using the SPA400 as my FXO port and varphonex as my VOIP ITSP. I’m able to make inbound PSTN calls with no problems. When I try to setup outbound PSTN, it seems to be send something invalid to the SPA400. I believe it is not stripping the preceeding 9 or something. In either case, I get a circuit-busy, congestion message. I haven’t even tried VOIP other than succesfully registering a peer. If anyone can help in that space, it would be great. My sip.conf and extensions.conf are as follows:
========================sip.conf==========================
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
register =>spa400@192.168.1.151/spa400
register =>spa921:mytest@192.168.1.142
register =>6000@192.168.1.140/6000
register =>5435377:fuente1@sip.varphonex.com/5435377
[spa400]
type=friend
user=spa400
host=192.168.1.151
dtmfmode=rfc2833
canreinvite=no
context=from-pstn
insecure=very
[spa921]
username=spa921
secret=mytest
type=friend
host=dynamic
context=to-pstn
[6000]
type=friend
context=default
host=dynamic
[varphonex]
type=peer
host=sip.varphonex.com
fromuser=5435377
fromdomain=sip.varphonex.com
secret=fuente1
context=from-varphonex
canreinvite=no
[sip.varphonex.com]
username=5435377
user=5435377
type=friend
insecure=very
host=sip.varphonex.com
fromdomain=sip.varphonex.com
context=from-pstn
================end sip.conf=============================
================extensions.conf===========================
[general]
static = yes
writeprotect = yes
;clearglobalvars = yes
[global]
DIALOUT = 9
[default]
[from-pstn]
exten => spa400,1,Dial(SIP/spa921)
exten => spa400,n,Hangup()
[to-pstn]
exten => _9xxxxxxxx,1,Dial(SIP/spa400,20)
;exten => _9xxxxxxxx.,n,Hangup()
[from-varphonex]
exten => varphonex,1,Dial(SIP/spa921)
[to-varphonex]
exten => spa921,1,Dial(SIP/varphonex)
==========================end============================
Please help.