On Thu, 2009-11-12 at 14:50 +1100, Michael Wyres wrote: > 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. > > I think this is more a problem that you are not setting your external IP address correctly so the provider can send RTP back to you. Make sure you have either "externip", "externhost" or "stunaddr"(1.6) set correctly. The do a "sip show settings" in the CLI to see if the correct address is set. If you are behind nat "canreinvite" should be set to no.
-- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001
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