[Asterisk-Users] SIP Extension behind NAT,
Asterisk on a public domain
Michael West
mwest at westmarkinc.com
Wed Nov 23 07:52:42 MST 2005
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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