Gonchi Mateos has posted details of the continuing work on IPV6 support for Asterisk:
Hi all,
We´ve been working with the patch originally submitted by Mikael Magnusson.
You can get it for asterisk realease 1.0.8 from asterisk-ipv6_23072005.patch
We've donde quite a lot of SIP testing with Linphone and it seems to be working quite well. Linphone is not handling correctly the reinvites (it answers to them saying the RTP is at the loopback address), there's a bug there, so the audio should go through asterisk in all communications.
As Mikael said, you should configure bindadd=[::0] for dual stack support.
BR
Gonchi
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