Have you tried "nat=yes" in the definition in sip.conf?

-----Original Message-----
From: [email protected] 
[mailto:[email protected]] On Behalf Of Landy Landy
Sent: Thursday, 12 November 2009 13:30
To: [email protected]
Subject: [asterisk-users] Can't connect to voip provider over NAT

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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