[Asterisk-Users] Comparing Callmanager to Asterisk

[Time Bandit]

> 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.

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Wait, maybe I shouldn't have read this email to begin with !

hth