Joseph Shi [EMAIL PROTECTED] wrote: > (Article auto-converted from unnecessary HTML to nice plain text.) > > Does anybody know if the voice actually gets routed through Asterisk for > calls between SIP devices? I just wonder if calls between SIP devices > would take up any bandwidth or CPU at the Asterisk server. Please > advise. > SIP devices will send re-invitations in an effort to find the most efficient route for the voice data, bypassing the server(s) etc. In a lot of cases, the two endpoints will end up speaking to one another directly.
You can set up Asterisk to keep itself in the loop (canreinvite = no), or it might want to remain in the loop regardless of your settings. For instance, Asterisk will want to remain in the loop if you're recording the call - for obvious reasons. -- _/ _/ _/_/_/_/ _/ _/ _/_/_/ _/ _/ _/_/_/ _/_/ _/ _/ _/ _/_/ _/ K e v i n W a l s h _/ _/ _/ _/ _/ _/ _/ _/_/ [EMAIL PROTECTED] _/ _/ _/_/_/_/ _/ _/_/_/ _/ _/ _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
