Thought i’d quickly write this for those having no audio issues with Gtalk.
First, follow the guide here to get it setup properly. Remember to have the DTMF(1) in your dialplan before executing into the actual internal dialplan per the document referred to earlier.
The issue is the headers that are sent out to google contain your internal IP (since you’re NATting), so you need a helper per-se otherwise the RTP is discarded. The solution is simple, use a stun server.
For FreePBX users, edit the /etc/asterisk/rtp_custom.conf, rest of you, simply edit the /etc/asterisk/rtp.conf in general section
Add the following line in bold, here i am using Google’s Stun server.
PS> Ice support must already be there, anyway…
And you should get two way audio without an issue.
Have a great weekend.