I have a problem using asterisk 1.6 with realtime sip. When I add sip channel (my sip provider) to asterisk using realtime sip (http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip), incoming calls don't work for me. In asterisk CLI I get message:
NOTICE[19805]: chan_sip.c:21250 handle_request_invite: Sending fake auth rejection for device "test" <sip:[email protected]>;tag=as0af02b0c. This is what happens in case I use hostname as a value of host parameter in sip table. When I use IP address instead of hostname, everything works fine. >From the other hand, when I setup the same sip channel using sip.conf file, everything works fine as well, even with hostname as host parameter. Rushan Shaymardanov -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
