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
