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:[EMAIL PROTECTED] > Sent: Tuesday, June 14, 2005 9:13 PM > To: [email protected] > Cc: [EMAIL PROTECTED] > 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 <[EMAIL PROTECTED]> > > Subject: [Asterisk-Users] Call being answered, but no audio on > either > > end (Intermittent) > > To: "Asterisk Users (E-mail)" <[email protected]> > > Message-ID: > > <[EMAIL PROTECTED]> > > 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! > > > > > > ------------------------------ > > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
