[Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain

[Michael West]

I'm pasting something from another user on this list from 14/11/05
 
I would recommend that you do a little research on google, voip-
info.org, and the list archives.

To connect to an Asterisk box that sits behind NAT, you need to forward
ports 5060 and 10000-20000 too the asterisk box, and you need to
configure the externip, localnet, and nat variables in sip.conf. 

audio problems are almost always due to the RTP stream (ports
10000-20000) not being forwarded properly, either due to the port
forwarding setup or the sip.conf settings.

Tom

----------------------------------------------------------

Tom Rymes

Cascade Link Systems

www.cascadelinksystems.com <outbind://12/www.cascadelinksystems.com> 

(603) 375-1414


________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Bharath
Khambadkone
Sent: Wednesday, November 23, 2005 9:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a
public domain


By default AMP had NAT=yes in sip.conf, I read in some posts to change
it to one, i was just trying my luck if that works. I have tried
NAT=yes, The Phone gets registered, I can also make & recieve calls but
as soon as the call is picked I dont hear anything at both ends. Does
this have anything to do with codecs?

Thanks


On 11/22/05, C F <shmaltz at gmail.com> wrote: 

	On 11/22/05, Bharath Khambadkone <bkalthod at gmail.com> wrote:
	> Hello All,
	>  I'm fairly new to asterisk. I have read about the problems
about NAT, But
	> can't seem to find a solution. 
	>  My Asterisk is on a public domain, there is no NAT or
firewall in front of
	
	
	If no nat then why do you have nat=1 in sip.conf?
	
	
	> the asteris box. I have sucessfully connected iax2 softphones
& was able to 
	> recieve & make calls. In the same locations where I have the
iax2 extensions
	> working I have set up a a SIP softphone & a SIP ATA
(Sipura2002). Both teh
	> sip phones are able to register. I can also make & recieve
calls but cannot 
	> hear anything after the call is answered at both ends. I'm not
sure what is
	> causing this problem. By the way I'm using SME server 7(centos
4.2)  with
	> A at H installed.
	>
	>  my Sip.conf :
	>  [2008] ;(Sipura2002)
	>  username=2008
	>  type=friend
	>  secret=2008
	>  record_out=Adhoc
	>  record_in=Adhoc
	>  qualify=no
	>  port=5060
	>  nat=1
	>  mailbox=2008 at device 
	>  host=dynamic
	>  dtmfmode=rfc2833
	>  context=from-internal
	>  canreinvite=no
	>  callerid=device <2008>
	>
	>
	>  [2009] ;X-Lite Soft Phone
	>  username=2009
	>  type=friend 
	>  secret=2009
	>  record_out=Adhoc
	>  record_in=Adhoc
	>  qualify=no
	>  port=5060
	>  nat=1
	>  mailbox=2009 at device
	>  host=dynamic
	>  dtmfmode=rfc2833
	>  context=from-internal 
	>  canreinvite=no
	>  callerid=device <2009>
	>
	>  Thanks in advance..
	>
	>
	>
	>
	>
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