I am trying to use asterisk as a gateway between SER and the PSTN. Should the nat=yes config work with these sip.conf settings ?asterisk is trying to send it's response back to the private IP. [general] context=OUTGOING autocreatepeer=yes [Provider] type=friend username=XXXXX secret=XXXXX host=xxxxx.FakeProvider.com nat=yes --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.698 / Virus Database: 455 - Release Date: 6/2/2004
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050712/db17133c/attachment.htm
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 > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com