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

[Steve Gladden]

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




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