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
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http://www.g7ltt.com
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Randolph, NJ
http://www.g7ltt.com
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