Hey, This is probably a rather stilly question... If I pick up my SIP phone that's registered to my asterisk server and dial a number that asterisk recognises as destined for a SIP trunk (could be a static route, or an ENUM lookup) or another SIP device registered on said asterisk server (internal extension to extension call), what route does the actual audio take? The control connection (port 5060) obviously goes via the asterisk server as it has to work out where to send the control to, but I could quite easily imagine the audio going directly handset to remote server or handset to asterisk to remote, and handset to handset or handset to asterisk to handset. Thanks -- Mike Williams
>>>>> "MW" == Mike Williams <mike.williams@comodo.com> writes:MW> The control connection (port 5060) obviously goes via the asterisk MW> server as it has to work out where to send the control to, but I MW> could quite easily imagine the audio going directly handset to MW> remote server or handset to asterisk to remote, and handset to MW> handset or handset to asterisk to handset. Try looking for "reinvite" or "canreinvite", and you will be enlightened. /Benny
You can make RTP pass through Asterisk, or not. Look in voip-info.org about "Native Bridge" and "sip.conf" "canreinvite" option. And may be this page will be usefull too: voip-info.org/wiki/index.php?page=Asterisk+SIP+not-proxy Regards On 10/31/06, Mike Williams <mike.williams@comodo.com> wrote:> Hey, > > This is probably a rather stilly question... > > If I pick up my SIP phone that's registered to my asterisk server and dial a > number that asterisk recognises as destined for a SIP trunk (could be a > static route, or an ENUM lookup) or another SIP device registered on said > asterisk server (internal extension to extension call), what route does the > actual audio take? > > The control connection (port 5060) obviously goes via the asterisk server as > it has to work out where to send the control to, but I could quite easily > imagine the audio going directly handset to remote server or handset to > asterisk to remote, and handset to handset or handset to asterisk to handset. > > Thanks > > -- > Mike Williams > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users >-- "Su nombre es GNU/Linux, no solamente Linux, mas info en gnu.org"