> Callmanager does nothing than construct and tear down calls and the actual > RTP stream does not flow through the Callmanager but is direct from IP > device to IP device. How does this work with Asterisk? I read something > that lead me to believe that Asterisk has to process the entire call, is > this the case? The simple answer is : Depends on how you configure it. If you set your sip account as "canreinvite=yes", it will behave as CallManager.
But it also depends on other things, like if it's SIP calling a SIP you can have direct IP to IP flow. But a SIP calling an IAX/MGCP/H323/ZAP will be different : Asterisk will stay in the middle to handle conversion. Even a SIP calling a SIP that use a different codec will have asterisk stay in the path. > Confidentiality Notice: This e-mail message, including any attachments, is > for the sole use of the intended recipient(s) and may contain confidential > and privileged information. Any unauthorized review, use, disclosure or > distribution is prohibited. If you are not the intended recipient, please > contact the sender by reply e-mail and destroy all copies of the original > message. Wait, maybe I shouldn't have read this email to begin with ! hth _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users