[Freeswitch-users] Calling an IPv6 host with a SIP URI from portAudio

[Jason White]

I notice that
pa call sip:nnnn at host
fails if the host in question only has an IPv6 address (i.e., an AAAA record
in DNS).

The logs show that FreeSWITCH is trying to use the internal profile, and
failing.

If I write a dialplan extension that accesses the same address using the
internal-ipv6 profile, it succeeds.

In the supplied default.xml dial plan, the SIP URI is processed thus:

<extension name="sip_uri">
  <condition field="destination_number" expression="^sip:(.*)$">
<action application="bridge" data="sofia/${use_profile}/$1"/>
  </condition>
</extension>

I can't find any documentation of use_profile on the wiki, but clearly it
takes the value "internal" in this case.

What would be the best way to fix this so that it will work regardless of
whether the host is reachable over IPv4 or IPv6, or both? I could rewrite the
extension to try multiple SIP profiles, but there could be a better way -
hence the question.