Sorry, but does that mean if I disable the iptable still I have to do something else? I am talking about Fedora.
Also, do I have to give the extern ip to be my DSL router public IP? What it if is dynamic, so I have to put the ddns name? Regards Bilal > > > Ok, I found the problem. I suggested that I disabled > completely my > shorewall-firewall, because there were no rules loaded. But > I were > mistaken... shorewall loads some kernel-modules, especially > ip_nat_sip > and ip_conntrack_sip, and these modules interfere with > asterisk! > > http://www.mail-archive.com/shorewall-us...@lists.sourceforge.net/msg03968.html > > Regards > Holger > > > Holger Latz schrieb: > > Hi all, > > > > I'd like to connect a softphone at home (nat, > dynamic-ip) to a sip-phone > > in the office via asterisk 1.4.21 (nat, fixed-ip). SIP > works well, the > > phone is ringing, but when I pickup the call, > there's no audio on both > > sides. > > > > I debugged the rtp-traffic at home. As long as the > phone is ringing, > > everything is fine. But after the pickup, asterisk > sends a SIP/SDP > > package with its private address (192.168.100.10). > After the softphone > > received this package, it tries to send RTP data to > this address! > > Obviously those packages never reach asterisk... > > > > Does 'externip' just works for SIP and not for > RTP? > > Where does the the internal IP-address come from and > how can I set the > > right one? > > > > > > My configuration: > > > > [general] > > externip = 85.XXX.XXX.XXX > > nat = yes > > localnet = 192.168.100.0/24 > > > > [42] > > deny=0.0.0.0/0.0.0.0 > > disallow=all > > type=friend > > secret=XXX > > qualify=yes > > port=5060 > > pickupgroup= > > permit=0.0.0.0/0.0.0.0 > > nat=yes > > mailbox...@device > > host=dynamic > > dtmfmode=rfc2833 > > dial=SIP/42 > > context=from-internal > > canreinvite=no > > callgroup= > > callerid=device <42> > > allow=alaw > > accountcode= > > call-limit=50 > > > > > > Regards > > Holger _______________________________________________ -- 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