asterisk users - Jun 2005 - RE: Call being answered, but no audio on either end

Thanks Gene.

Here is my localnet:


localnet=172.16.64.0/255.255.240.0

Which matches our subnets network address and subnet mask. Are you
recommending that I make it more restrictive?


Thanks,
Geoff

> -----Original Message-----
> From: Gene Willingham [mailto:gwillingham@comcast.net]
> Sent: Tuesday, June 14, 2005 9:13 PM
> To: asterisk-users@lists.digium.com
> Cc: gmanning@zoom.com
> Subject: RE: Call being answered, but no audio on either end 
> 
> 
> 
> I think I found the source of this.  Been tracing it for a 
> week.  Look in
> sip.conf.  It appears the definition of localnet has a 
> bearing on how some
> sip devices handle invites and NAT.
> 
> I had changed the localnet to 192.168.3.0, but did not change 
> the netmask.
> 
> localnet=192.168.3.0/255.255.0.0; All RFC 1918 addresses are 
> local networks
> 
> When I changed the netmask to 255.255.255.0 the problem 
> appeared to go away.
> It appears the more restrictive localnet the better results 
> at handling sip
> devices behind NAT devices.
> 
> Gene 
> 
> >   19. Call being answered,	but no audio on either end
> >       (Intermittent) (Geoff Manning)
> > ------------------------------
> > 
> > Message: 19
> > Date: Tue, 14 Jun 2005 17:30:31 -0400
> > From: Geoff Manning <gmanning@zoom.com>
> > Subject: [Asterisk-Users] Call being answered,	but no audio on
> either
> > 	end (Intermittent)
> > To: "Asterisk Users (E-mail)"
<asterisk-users@lists.digium.com>
> > Message-ID:
> > 	<D1696C471C6CD511A0BE00D0B7A932DE0957C97C@southe01.zoomtel.com>
> > Content-Type: text/plain;	charset="iso-8859-1"
> > 
> > The best type of error possible, intermittent.
> > 
> > We have PSTN numbers being switched to SIP then forwarded 
> to our Asterisk
> > server which sits inside our LAN
> > 
> > Every once and a while (maybe 1 out of every 20 calls) goes 
> like this:
> > 
> >     -- Executing Answer("SIP/213.199.36.50-0818e3e8",
"")
> in new stack
> >     -- Executing Ringing("SIP/213.199.36.50-0818e3e8",
"")
> in new stack
> >     -- Executing Dial("SIP/213.199.36.50-0818e3e8", 
> "ZAP/g1/:8213") in new
> > stack
> >     -- Called g1/:8213
> >     -- Zap/1-1 answered SIP/213.199.36.50-0818e3e8
> >     -- Hungup 'Zap/1-1'
> >   == Spawn extension (from-gv-uk, 441252580625, 3) exited 
> non-zero on
> > 'SIP/213.199.36.50-0818e3e8'
> > 
> > Looks normal right? During this whole exchange, neither 
> side can hear the
> > other. Not even a ringing sound.
> > 
> > The above looks no different than the successful calls.
> > 
> > Has anyone seen this type of behavior before?
> > 
> > Thanks!
> > 
> > 
> > ------------------------------
> 
>