In my Freepbx I changed bindport to 6070. Extensions are working perfectly.
I’m trying to configure a SIP trunk. My VOIP provider works on the default port 5060.
Then the SIP trunk configuration I set:
port = 5060
But in trying to make a call asterisk insists on using the 6070 port to connect to the SIP provider.
In the Asterisk console I get the following message saying forbidden, and you can see that he insists on using the same bindport door.
Look:
Received response: “Forbidden” from ‘“ext2” sip:[email protected]:6070 ;tag=as3b872729’
For testing only, I changed back to the default value bindport 5060, and could make calls over the SIP trunk.
What more can I set up? I would like to put the asterisk to listen on port 6070, but the SIP trunk needs to connect on the default port of 5060 on my provider.
Thank U
Details of Trunk
PEERS Details:
username=username
fromuser=username
type=peer
secret=password
qualify=no
nat=no
port=5060
insecure=very
host=vono.net.br
fromdomain=vono.net.br
dtmfmode=rfc2833
domain=vono.net.br
disallow=all
allow=ilbc&gsm&alaw&ulaw
context=from-vono
canreinvite=no
Register String
username:[email protected]:5060/username