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... 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
