On 19:56, Tue 11 Mar 08, NOC ph wrote: > Yes, here's my sip.conf > > [general] > register = 1000:[EMAIL PROTECTED]/1000 > callevents = yes > port = 5060 > nat = yes > canreinvite = no > #bindaddr = 172.16.1.1 - if I en able this call cannot go out... > localnet = 172.16.1.0/24 > externip = 203.172.25.11 > > Thanks... >
Ok, try to enable all logging in pf and 'set loginterface' etc. After that, run: tcpdump -n -e -x -i pflog0 There you will see the blocked traffic. Maybe that will give you an idea. > > Michiel van Baak wrote: > > On 17:31, Tue 11 Mar 08, NOC ph wrote: > >> Hi Mich, > >> > >> I added the following line for the RTP its still the same, I can hear > >> ring but no voice when answer from the other side. Any more ideas? > > > > Firewall rules look ok now. > > > > Like I said, did you set externip and localnet settings in > > asterisk sip.conf ? > > > > _______________________________________________ > -- 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 -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD "Why is it drug addicts and computer aficionados are both called 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
