I hate SIP. The only reason I'm doing this is that its cheaper than deploying the server to a colo facility. My provider has given me a non-standard IP block, so I can't do typical routing. I have Asterisk server <-> MT w\NAT <-> PPPoE <-> MT <-> Provider. I setup a dst-nat on 5060 to the Asterisk box. Audio from Asterisk --> PSTN works great. Audio Asterisk <-- PSTN does not. Ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070329/ddbfcfc3/attachment.htm
What do you mean by 'non-standard' IP block? Is the Asterisk machine behind a NAT, or are only your clients? Did you look at the nat setting sin sip.conf? Do you have a static public address that can be routed to the Asterisk box? ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mike Hammett Sent: Thursday, March 29, 2007 11:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] SIP & NAT I hate SIP. The only reason I'm doing this is that its cheaper than deploying the server to a colo facility. My provider has given me a non-standard IP block, so I can't do typical routing. I have Asterisk server <-> MT w\NAT <-> PPPoE <-> MT <-> Provider. I setup a dst-nat on 5060 to the Asterisk box. Audio from Asterisk --> PSTN works great. Audio Asterisk <-- PSTN does not. Ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070329/7e514d30/attachment.htm
Mike Hammett wrote:> I hate SIP. The only reason I'm doing this is that its cheaper than > deploying the server to a colo facility. My provider has given me a > non-standard IP block, so I can't do typical routing. > > > > I have Asterisk server <-> MT w\NAT <-> PPPoE <-> MT <-> Provider. > > > > I setup a dst-nat on 5060 to the Asterisk box. > > > > Audio from Asterisk --> PSTN works great. Audio Asterisk <-- PSTN does > not.That would be expected since you did not forward the ports used for RTP. See /etc/asterisk/rtp.conf A sample is in the Asterisk source. Did you also set localnet= and externip= options in sip.conf [general]. SIP works just fine with NAT if you have it correctly configured and your server is on a static IP address.