[asterisk-users] Which features are lost when canreinvite is turned on ?

[Anthony Francis]

Olivier wrote:
> Hi,
>
> My setup is :
> PSTN --------- ISTP Network ----------- Router ------------- Asterisk 
> ---------- SIP Phones
>
> Phones are located in the same location.
> I'm thinking about installing new phones in other locations (small 
> agency, home workers), registering those phones to the same Asterisk 
> server.
>
> As every location has DSL access, I think I should have those phones 
> directly exchanging RTP data with ITSP media gateway, without passing 
> through Asterisk server, with canreinvite = yes option.
>
> Before, trying this, I'm wondering which features I would loose in the 
> process ?
> Will I keep the ability to :
> - record CDR,
> - listen to DTMF tones
> - ...
>
> What do you think ?
>
> Regards
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>
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You would never lose CDR's because of this feature, and your DTMF should 
be out of band (in sip messages) anyway. A re invite really just makes 
the audio connect directly between the sip endpoints in a connection, 
the sip proxies still receive messages.

To understand this better you should read this document: 
http://www.ietf.org/rfc/rfc2543.txt

Hope this helps,
Anthony