I have iptables FORWARD to ACCEPT by default: iptables -P FORWARD ACCEPT
and still have the same problems. Now, the dsl modem is also opened. not blocking any ports as well. --- On Sat, 11/14/09, Michelle Dupuis <[email protected]> wrote: > From: Michelle Dupuis <[email protected]> > Subject: Re: [asterisk-users] Can't connect to voip provider over NAT > To: "'Asterisk Users List'" <[email protected]> > Date: Saturday, November 14, 2009, 1:03 PM > I'll start with a guess - your > asterisk box or firewall is blocking SIP > ports. Diagnose that first (stop iptables/check > iptables if unsafe) and try > again... > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] > On Behalf Of Landy Landy > Sent: Saturday, November 14, 2009 10:15 AM > To: Asterisk Users List > Subject: Re: [asterisk-users] Can't connect to voip > provider over NAT > > According to my provider they´re not receiving any request > from us but, now > everytime I try to place a call through them I´m getting: > > *CLI> sip show peers > Name/username > Host Dyn Nat > ACL Port Status > 100 > (Unspecified) > D 5060 > Unmonitored > 101 > (Unspecified) > D 5060 > Unmonitored > 102/102 > 172.16.0.15 D > 5060 > Unmonitored > 103/103 > (Unspecified) D > 5060 > Unmonitored > 104 > (Unspecified) > D 5060 > Unmonitored > 105 > (Unspecified) > D 5060 > Unmonitored > 106 > (Unspecified) > D 5060 > Unmonitored > 107 > (Unspecified) > D 5060 > Unmonitored > voipprovider/1800890999 MYEXTERNALIP > N > 5060 Unmonitored > 9 sip peers [Monitored: 0 online, 0 offline Unmonitored: 9 > online, 0 > offline] > > == Using SIP RTP CoS mark 5 > -- Executing [18008909...@default:1] > Dial("SIP/102-b6a05db0", > "SIP/18292574...@voipprovider") in new stack > == Using SIP RTP CoS mark 5 > -- Called 18008909...@voipprovider > > It just hangs here and nothing happens.......... > > > Here´s my sip.conf file: > > [general] > externhost=myexternalip > localnet=172.16.0.0/16 > > register => username:[email protected] > > allow=all > > [voipprovider] > type=peer > host=sip-gw.advancedvoip.com.do > username=username > fromuser=username > secret=password > port=5060 > canreinvite=YES > dtmfmode=rfc2833 > nat=yes > > > > What I´m I doing wrong? > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
