[Asterisk-Users] No reinvite - reason?

[BJ Weschke]

On 6/12/06, Roger Schreiter <roger at planinternet.de> wrote:
> 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.
>

 One reason might be is if you are passing parameters in app_dial (eg.
tT, etc) that require it to listen for DTMF that would cause it to
hang on to the RTP streams rather than reinvite them away.

-- 
Bird's The Word Technologies, Inc.
http://www.btwtech.com/