[Asterisk-Users] No reinvite - reason?

[Roger Schreiter]

Hi,

I put reinvite=yes in my sip.conf.
For testing, I restricted the codecs to alaw.
I have no modifiers in my dial command.

Thus, there should be no reason not to reinvite.

Call (sip, authenticated) comes in and is forward
via SIP (not authenticated) to another asterisk box.
Unfortunately, media path still passes through the asterisk
box in the middle.

Using sip debug I even can't find any attempt of a reinvite.

Now I would like to know, why the asterisk box in the middle
does not try to reinvite.


Any suggestions?
Roger.