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 > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users 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 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users