[Asterisk-Users] NAT=YES

[Rich Adamson]

FYI, there is no such thing as "reinvite". Someone started using
that in postings a long time ago and a few people keep posting
it, but it doesn't exist. 
(Check /usr/src/astersik/configs/sip.conf.sample)

------------------------
> Add canreinvite=no and reinvite=no to the relevant stanza in sip.conf
> 
> Mark
> 
> Klint, Peter wrote:
> > Good morning
> > 
> > Does anyone have experience with NAT=YES?  I have the following
> > configuration and am a bit confused as to why the Asterisk server
> > initially sends out RTP to the remote host private IP and then switches
> > to the public IP.
> > 
> > Configuration Info:
> > I have all users in SIP.CONF configured with NAT=YES
> > Asterisk has a public IP
> > Remote host is behind a firewall with NAT
> > 
> > When I sniff on the Asterisk public network, I see the following.
> > 
> > 1. INVITE from remote host public IP to Asterisk public IP
> > 2. 183 response from Asterisk public IP to remote host public IP
> > 3. RTP from Asterisk public IP to the remote host private IP
> > 4. RTP from remote host public IP to Asterisk public IP
> > 5. RTP from Asterisk public IP to the remote host public IP
> > 
> > Is there a way to prevent step 3 from happening?  Or, is there a way to
> > delay the invalid RTP from being sent from the Asterisk in step 3?
> > Does anyone know why the Asterisk sends RTP to remote host private IP?
> > I would expect NAT=YES to correct this issue.
> > 
> > Thanks,
> > 
> > Peter
> > 
> > 
> > 
> > 
> > ------------------------------------------------------------------------
> > 
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> -- 
> 
> Mark, G7LTT/KC2ENI
> Randolph, NJ
> http://www.g7ltt.com
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