On Sat, 7 Jul 2007, 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 ?
The ability to pass audio between the endpoints if they are behind NAT firewalls... You might be able to get it work, but I wouldn't bet on it. See: http://www.voip-info.org/wiki-Asterisk+sip+canreinvite http://www.voip-info.org/wiki/index.php?page=Asterisk+Letting+SIP+clients+connect+directly http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions > Will I keep the ability to : > - record CDR, > - listen to DTMF tones > - ... > > What do you think ? I think it's challenging when NAT is involved! Gordon _______________________________________________ --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
