Steve, > The mission is to actually get a reinvite to work on the lan. There isn't anything special to get this working... normally. I trust you verified the traffic flow with a network monitor tool (tcpdump?), correct? Does SIP debug give you any info (i.e., does it match the right peer) -- you don't show if you allow reinvites globally? What about the nat= setting?
Couple pointers I can give you to get you excited: 1) Reinvites work quite reliably, I use them between the PTSN gateway and the end user's ATA, all the way across the Internet -- nicely reduces latency. 2) If you use RFC2833 for DTMF you can issue an reinvite and still use t/T for transfer. NOTE that you have to modify the source to make asterisk reinvite even when it needs to listen to DTMFs. I give no guarantees how well it will work for you but it does work. See "AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1" in rtp.c. 3) Reinvites *can* work even if both ends are behind NAT. It really depends on the NATing router and the ATA. Sipura's and good NAT routers work, but I would not call it "reliable" -- it's really pushing it a bit... So if you really want to see why your Reinvites do not work, then you probably will have to make your hands dirty and analyze where ast_rtp_bridge() in rtp.c bails out. But since you are on a LAN it makes the situation a lot easier. --Luki _______________________________________________ --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
