[asterisk-users] canreinvite

[C F]

The others answered correctly personal I like using rtp debug.
As for making sure in the DialPlan that the RTP goes end to end
without asterisk.
1. Make sure they both use the same codec and protocol.
2. Don't put any options in app_dial, like tTwW or anything else that
will force asterisk to stay in the stream to listen for DTMF.

On 9/11/07, bilal ghayyad <bilmar_gh at yahoo.com> wrote:
> Dear C F;
> So in that case, if I placed canrenvite=yes for both
> endpoint, it is not condition that traffic will be
> directly via the endpoint while signaling via Asterisk
> as still Asterisk should detect whethor it is
> necessary to stay in the path or not? Please advise.
>
> How can I know that the traffic went directly between
> the endpoints and did not go via the asterisk?
>
> Regards
> Bilal Ghayad
> Mobile: 009659849460
>
>
> -----------------
> By default assuming you have no global setting
> otherwise, if asterisk
> doesnt see a need to stay in the path then it wont.
> hence if it has to
> transcode between different codecs, capture DTMF or
> different
> protocols it will stay in the path.
>
> On 9/9/07, bilal ghayyad <bilmar_gh at yahoo.com> wrote:
> > Hi List;
> >
> > If I need traffic to be directly between the
> > endpoints, then I have to set the canreinvite = yes?
> >
> > If I did not configure the canrenvite at all, then
> by
> > default it will pass the traffic via Asterisk and
> not
> > directly between the endpoints?
> >
> > What if one endpoint was SIP and configured with
> > canreinvite=yes while other endpoint was IAX2 and
> > configured with canreinvite=yes, then they can send
> > traffic to each other directly or it will be via
> > Asterisk?
> >
> > Regards
> > Bilal
>
>
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