asterisk users - Feb 2008 - canreinvite question

Hi All,


if i do this setup:

                                              |---[ext 100]
                          |--[router/nat gw]--|
                          |                   |---[ext 101]
                          |
[asterisk]--[internet]---|
                          |
                          |                   |---[ext 200]
                          |--[router/nat gw]--|
                                              |---[ext 201]


If i set, canreinvite=yes on all ext, assuming all ip phones have the 
same codec, if 100 calls 101, or vice versa will rtp still go thru 
asterisk? and same scenario for 200 to 202 or vice versa.

what if 100 call 200 or 201? or 200 calls 100 or 100? will rtp still go 
thru asterisk?

thank you

regards,
Ron

On Fri, 22 Feb 2008 18:50:16 +0800, Ron <ron at silverbackasp.com>
wrote:
>If i set, canreinvite=yes on all ext, assuming all ip phones have the 
>same codec, if 100 calls 101, or vice versa will rtp still go thru 
>asterisk? and same scenario for 200 to 202 or vice versa.
... and I'd like to add to this question: If the phones have the
option "Enable NAT", I expected them to be able to talk to each other
directly, but they didn't, and I had to set them to
"canreinvite=no"
in sip.conf. Why is that?