On 09/22/2010 08:36 AM, Carlos Chavez wrote: > Do you have a localnet statement in your sip.conf? That and using > nat=no will make sure Asterisk does not replace the IP address in the > Invite. >
I just wanted to give a +1 for this response. I am using openvpn to connect road warriors and remote offices to a central asterisk server. When setting up the configuration for the road warriors I created a new subnet for them, but forgot to include their subnet as a localnet directive in sip.conf. The result was that sip clients on the road warrior network would be able to register, but then when initiating a sip call the 200 response (to the INVITE from the client) from the asterisk server would include a contact address for some external ip that I did not recognize. This hint here allowed me to fix this bug, now calls from the road warrior subnet are coming in fine. Thanks! -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
