Hi,
I have an Asterisk behind nat on the internet. I registered it in my ITSP. But couldn’t route calls to ITSP or get call from this. my confs:
sip.conf:
register => tcp://accoun:secret@itsp.com:5560/itsp
[itsp]
host=dynamic
secret=secret
context=itsp_incoming
type=friend
transport=tcp
directmedia=no
;avpf=yes
force_avp=yes
encryption=yes
icesupport=yes
disallow=all
allow=all
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
extensions.conf:
[internal]
exten => _X.,1,Dial(SIP/${EXTEN})
;exten => _X.,n,Answer()
exten => _X.,n,Hangup()
[outgoing]
exten => _X.,1,Dial(SIP/itsp/${EXTEN})
;exten => _X.,n,Answer()
exten => _X.,n,Hangup()
[itsp_incoming]
include => internal
I receive this error:
[Mar 15 01:39:08] WARNING[2500][C-0000002a]: app_dial.c:2437 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
plz help.
B.R