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  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=default
Anyone got an idea?