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

[Bharath Khambadkone]

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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