Dear list, I'm trying to get Asterisk to work dual-stack on Linux and I'm left with a question.
Imagine that a user (on the road) connects to Asterisk from various places. Many of them probably don't have IPv6 support yet. However, his house and office do have IPv6 connectivity. I would like to make sure that whenever IPv6 is available, the connection will be made over IPv6, but offer IPv4 as a "fallback" option. The pitfall, in my opinion, is to create one sip.conf entry for that user which supports the voicecalls over IPv4 and IPv6. However, settings like nat=, directmedia= and/or canreinvite= seem to be addressfamily unrelated. I want to configure it in a way that when I connect using IPv6, no NAT options should be set and the mediapath (almost) always should be directly between the peers and not over the Asterisk server (so, "nat=no" and "canreinvite=yes"). But, when a user comes via IPv4, changes are that he's on NAT. When that happens obviously the connections should traverse the NAT using options like "nat=yes" and "canreinvite=no". There's little to no documentation available as far as my google-skills go. There's some in sip.conf, and I couldn't find anything on the website. Does anyone have some pointers for me, either for the configuration of the sip.conf entry or for more documentation on this? Best regards, Wouter Schoot -- _____________________________________________________________________ -- 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
