[asterisk-users] Problems with realtime sip

[Shaymardanov Rushan]

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:test at my.sip-provider.org>;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.