[asterisk-users] Can't connect to voip provider over NAT

[Carlos Chavez]

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: 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.
> 
> 
	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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