On Thu, Jul 03, 2003 at 03:57:09PM +0000, WipeOut . wrote: > IIRC asterisk by default will not participate in the call between two SIP phones.. > It will help establish the session to the correct UA and then have nothing more to > do with it unless the call is transferred to another UA in which case Asrerisk will > again be involved in setting up the call..
Asterisk will be handling the signalling for the call, not the voice stream. If you look at 'show channels' or 'sip show channels' while the call is up you will see that Asterisk is aware of it. Running tcpdump will show you that the phones are still talking to Asterisk. > So no when 2 SIP UA'a are connected there should be no CPU load on the Asterisk > server.. Well there is a minimum amount just for keeping track of the call, but per call is very low. -Andrew _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
