[Asterisk-Users] SIP reinvite still does not occour

[Roger Schreiter]

Hi,

I have in my sip.conf

disallow=all
allow=alaw

in order to avoid any codec problems disturbing reinvite.

And of course I have:
canreinvite=yes

In extensions.conf there is only one Dial command. It
has no qualifiers like t or T.
Just Dial(SIP/01234567 at 1.2.3.4)

Anyway, asterisk does not try to reinvite.

asterisk tells
  -- Attempting native bridge of SIP/01234567 ...

but in the debug output there no reinvite.

Using tcpdump I can see, that the audio data are
going via the asterisk box in the middle, not direct
between the endpoints.


Is there anything else, which can prevent a reinvite?

dtmp-settings? nat-settings?


Thanks for any hints!
Roger.