[Asterisk-Users] canreinvite always =no * no matter what we try :-(

[Luki]

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