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

[Bharath]

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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