[Asterisk-Users] SIP reinvite still does not occour

[Roger Schreiter]

Hoa Thai Duy schrieb:
> Roger
> 
> If transcoding happened (from G.729 to GSM, any to any) then Asterisk will
> issue no re-INVITE, for sure.
> 
> Pls. change 
> 
> Disallow=all
> Allow=gsm (only one codec)


Hi,

yes, to avoid transcoding problems I only have one
codec, just alaw. Anything else is disallowed.
That's why I don't understand, why there is no reinvite.

Thanks for answering!

Roger.