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

[Manny A. Wise]

Well, as the user stated on the original message, the asterisk server is
behind a NAT and the client is also behind a NAT..

if you make it work just by opening ports, let me know..I have never been
able to get it to work, that's why I don't use sip, just plain iax2 for
everything. :-)

 

Manny

 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Bharath
Sent: Wednesday, November 23, 2005 10:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public
domain

Thanks Michael,
I think thats is the problem, I have opened only ports 5060-5082, I need to
open 10000-20000 as well. I will try that and post the result when i get
back home.
Thanks

On 11/23/05, Michael West <mwest at westmarkinc.com> wrote:

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

(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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