[Asterisk-Users] NAT=YES

[Mark Phillips]

OK, so I have a nonexistant line in my settings. Why then when I remove 
it does my phone call fail?

Rich Adamson wrote:
> 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
>>_______________________________________________
>>Asterisk-Users mailing list
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>>
> 
> 
> ---------------End of Original Message-----------------
> 
> 
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-- 

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com