Hi, I just test the "reinvite" feature in this version and I realized that the SDP is changed in the early stages of the SIP call and thus * is not sending the reINVITEs.
Is there any way to disable the early rtp bridge but still having the reINVITEs? (may be some parameter in sip.conf ). Let's say the device is behind a NAT router and knows its public IP, if the RTP port choosen by the NAT-router is not the same port the device set in the SDP then the early bridge is not going to work. I'm not asterisk savvy, but may I can comment something out in chan_sip.c Any ideas? Thanks for your help, Humberto __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ --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
