Hello and thanks for replying!
> 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?), Actully ethereal, It is encouraging to hear that it does not take anything special. I've tried what seems to be a simple arrangement, no nat two phones on the lan same codec, lack of canreinvite line and also tried canreinvite=yes I am not using a global nat=yes statement. also tried nat=no on each phone just in case of a default option. > 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? I've not set nat= or canreinvite= globally just on each phone I can certainly try that but having specific settings on the phones seems to almost guarantee I know where I stand with those two :-) I've not torn apart the sip debug on this yet as I am quite new to SIP but will do so if need be..... Was just trying the simple approach first. > > 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... Yep I will eventually go there but right now still just trying to get it to work for a test on the lan and have not seen it fly yet. asterisk always creates a 'native bridge' and seems to hold on for dear life so far as I have seen :-) > > 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. Yep! Still new at this but enjoy getting hands dirty. Thanks for your time! Steve > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --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
