asterisk users - Nov 2009 - 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.

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


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Landy Landy
Sent: Thursday, 12 November 2009 13:30
To: asterisk-users at lists.digium.com
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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