I completely reinstalled asterisk and everything seemed to be fine. But after a while when i dial one of the phone numbers that is registered within asterisk, the console only shows: “Using SIP RTP CoS mark 5”, after which the line is imidiately disconnected.
The strange thing is that it only happens to one of the phone numbers… So in the shown code below 31NUMBER1 keeps working all the time, while 31NUMBER2 stops working after some time…
I am using 2 budgetphone SIP accounts from the netherlands. With the following settings:
[general]
externip=IP
srvlookup=yes
subscribecontext=ext-local-custom
limitonpeer=yes
host=dynamic
context=default
language=en
maxexpirey=3600
defaultexpirey=3600
disallow=all
allow=ulaw ; budgetphone gebruikt deze om te kunnen faxen
;allow=gsm
;allow=alaw
register =>31NUMBER1:PASS1@budgetphone.nl/31NUMBER1
register =>31NUMBER2:PASS2@budgetphone.nl/31NUMBER2
[31164712136]
type=friend
call-limit=50
qualify=300
username= 31NUMBER1
secret= PASS1
fromuser= 31NUMBER1
fromdomain=budgetphone.nl
insecure=very
nat=yes
host=budgetphone.nl
port=5060
canreinvite=no ; Typically set to NO if behind NAT
disallow=all
allow=g729,alaw,ulaw,gsm
context=defaultAnyone got an idea?