[asterisk-users] canreinvite

[Wai Wu]

Don't know about IAX. As for SIP, You will know what ip address and port
the audios should be transmitted to by looking at the sdp session. Just
goto the * console and enable sip debug.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of bilal
ghayyad
Sent: Tuesday, September 11, 2007 10:14 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] canreinvite

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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