[asterisk-gui] No sound

Matt Sales msales at gmail.com
Thu Apr 2 15:07:25 CDT 2009


Bob, you're dealing with a double NATted environment with both your Asterisk
server and phones behind firewalls.  Personally I think this is going to do
nothing but give you headaches.  I would add a 2nd NIC card to your asterisk
server and assign it a public IP address.  Now just have your softphones and
hardphones register to the public IP address.  You will still need to set
nat=yes for those remote extenstions in users.conf

I'm sure others will chime it that this a major security risk but if you use
iptables to block all ports except those required by asterisk (SIP, IAX, &
RTP) and set secure sip passwords you should be ok.

On Thu, Apr 2, 2009 at 2:57 PM, Bob Crandell <bob at assuredcomp.com> wrote:

> I have NAT=yes through out.
> I have both externip and localnet set.
>
>
> On Thu, 2009-04-02 at 00:03 -0700, Marvin Whitfield wrote:
>
> Bob,
> Definately a NAT issue. Had the same issues when I first started
> setting up remote extensions. Look in your sip.conf for externip=...
> and localnet=.... You'll use the to tell asterisk what's internat and
> what's external. Also remember to at NAT=yes to extensions that will
> be remote. The last thing you might need to modify is your firewall
> and phone settings. These vary widely by manufacturer but the general
> idea is that you need asterisk to be able to reach the phone that will
> be behind a NAT firewall so you'll probably need to forward some ports
> or configure the phone to keep the connection alive. Hope that helps.
>
> --
> Marvin
>
> On 4/1/09, Matt Brown (HC) <matt at mbrown.co.uk> wrote:
> > Hi Bob,
> >
> >> Hi guys,
> >>
> >> I've been testing softphones in Windows and SuSE.  From outside the
> >> office, I can make a call to a different extension and to an outside
> >> number but neither the caller nor the callee can hear anything.  I
> >> thought I was being stupid until I bought a Grandstream GXP 2020
> >> which wouldn't speak to me either from outside the office.  It does
> >> work inside the office.  I tested the softphone inside the office
> >> and it works too.  Ok so it's the firewall.  I have port 5060 UDP
> >> and 10001-20000 UDP open and pointed to the Asterisk box.  What am I
> >> missing?  I'm getting so close to being able to go live with this
> >> thing.
> >>
> >
> > Just double check the /etc/asterisk/rtp.conf to make sure port range
> > being used is correct. In addition I agree with Matt that it sounds
> > like a NAT issue. Take a peek at sip.conf or users.conf for the
> > extension in question and check nat=yes is set for that user/extension.
> >
> > Regards
> >
> > Matt Brown
> >
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>
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