I also wanted to add to this:
If you have users behind NATs then the canreinvite=yes will essentially make the phone ring but when it's picked up the call will break up and the two parties can't talk. Just went through this in my setup, the only way to get the two sides to talk was to set canreinvite=no.
This may be because I had both the calling and called sip devices behind a NAT on each end of the call. It may work if only one end is behind a NAT???
-- Start Your Own ISP! http://www.YourOwnISP.com
----- Original Message ----- From: "Kevin Walsh" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]>
Sent: Monday, October 25, 2004 5:30 AM
Subject: RE: [Asterisk-Users] Bandwdith usage
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.
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