23 feb 2007 kl. 09.52 skrev Michiel van Baak:
Hey, We have asterisk 1.2.4 (old I know) with a couple of snom phones, a couple of grandstream phones and around 65 philips dect stations. Now the problem: All calls do peer to peer RTP except the calls from dect station to dect station. snom to dect or dect to snom do peer to peer. So the sip config looks fine because all the 'static deskphones' honor the REINVITE and start talking to eachother. Our supplier told us they dont send SDP with the INVITE. Can this be the problem causing dect to dect calls to always use asterisk in the RTP path ?
If they do not send SDP with the INVITE there will be no media at all in the call. Very simple. Can be that they do not support re-invites. If so, you should see an error message in the SIP communication. /O _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
