Hello.
I'm trying to test an Asterisk server by using a VOIP provider for
international calls but, I'm having problems trying to get my server
communicate with theirs. I don't know if I'm having all these issues becuase
I'm behind NAT or what. I have the following in my server's sip.conf:
[provider]
type=peer
host=<theprovider's server>
username=<username>
secret=<password>
port=5060
canreinvite=YES
dtmfmode=rfc2833
I've tried opening all ports to test this but, still doesn't work. Now, I need
to know which especific ports to open in order to allow sip flow correctly.
Also enabled/opened ports 5060 - 5070 and the rtp: rtpstart=10000
rtpend=20000
Don't know what else to try. Please help.
Thanks in advanced for your help.
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