[Asterisk-Users] SIP (peer to peer?)

[Daniel Chabrol]

Hello!

I think that's true. In older asterisk versions I saw such a "hand-over" 
between 2 sip phones and asterisk. But with the current versions I can't 
get it to work. I think you have to set "canreinvite=yes" at both 
clients that this can work. Additionally both ends need to have a common 
codec. If not asterisk has to stay between and convert codecs.
But like I said I've problems with handover now. Has someone else 
encountered this loss of handovers in current asterisk versions? In my 
case I tried BudgetTone100 <-> Asterisk <->BudgetTone100.

Wim Venneman wrote:

> Hi all,
>
>Has anyone have an idea why, if you capture the files on a Asterisk network (ex with Ethereal) you always see the communication between the two sip phones( hard or soft) passing through the asterisk server (on UDP layer)
>Isn't SIP a protocol that (after that it has established the call) , he connects the two users with each other?
>
> 
>
>Maybe a stupid question, but I'm not a SIP expert.
>
> 
>
>Thank you for your help.
>
> 
>
>Wim
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>  
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