still got problem, but at least I got error message this time:
$this->bbcode_second_pass_code('', ' == Using SIP RTP CoS mark 5
-- Executing [xxxxxxxxxxx@outbound:1] Set("SIP/101-00000000", "CALLERID(dnid)=xxxxxxxxxxx") in new stack
-- Executing [xxxxxxxxxxx@outbound:2] Goto("SIP/101-00000000", "xxxxxxxxxxx,1") in new stack
-- Goto (outbound,xxxxxxxxxxx,1)
-- Executing [xxxxxxxxxxx@outbound:1] Dial("SIP/101-00000000", "Gtalk/myemail/xxxxxxxxxxx@voice.google.com") in new stack
[Apr 24 21:14:13] WARNING[24749]: channel.c:5414 ast_request: No channel type registered for 'Gtalk'
[Apr 24 21:14:13] WARNING[24749]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'Gtalk' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/101-00000000' status is 'CHANUNAVAIL'
')
xxxxxxxxxxx=the number I dialed. I got busy tone.
By the way. I followed this tutorial to install asterisk on Debian with no problem.
http://forums.plugpbx.org/index.php/topic,247.0.html