[Freeswitch-users] Greetings and a couple of questions

[Doc]

First, to all, greetings.

I am just beginning a quest in trying to setup a simple FS box to route 
my incoming skype account to a SIP ATA (SPA-1001).

I have this installed on Ubuntu 8.04 (since I read that it came with a 
stock tickless kernel with 100HZ tick).  I have (as far as I can tell) 
compiled the alsa 1.0.20 drivers and included the mod_skypiax dummy 
file.  I followed install instructions in the FS wiki for Hardy & FS & 
skypiax.

1) I am able to see a call come in, and it gets routed to the sample IVR 
to start.  The first thing off is that when I dial from PSTN -> 
Skype-In, it does not let me push any buttons.  If I launch a second 
skype client, and dial the skype user on FS directly, it works fine.  
Any ideas?

2) No matter how Skype works, when I hang up, it throws 3 or 4 errors 
(2010-02-16 20:41:45.384633 [ERR] mod_skypiax.c:826 rev 
16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? 
sent=-1), and when I try subsequent dials, I get pages and pages of this 
error, and no audio.  Eventually it crashes FS.  Any ideas?

3) When dialing extension to extension, or even testing out the IVR, it 
all works fine - no errors.  The only thing it does make me do, and I 
haven't tracked it down yet is it waits like 30 seconds before 
responding (unless I press the # after I type the extension number).  
Any ideas?

4) When I run startskype.sh, I see these errors (and are they worth 
concerning?) :

expected keysym, got XF86KbdLightOnOff: line 70 of pc
expected keysym, got XF86KbdBrightnessDown: line 71 of pc
expected keysym, got XF86KbdBrightnessUp: line 72 of pc
Could not init font path element /usr/share/fonts/X11/cyrillic, removing 
from list!
Could not init font path element 
/var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!
ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null
ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null

Thanks for the patience in letting me email to the group.  I used 
Asterisk in the past, and even though I am finding a bit of a learning 
curve using FS, I am enjoying it.  Hopefully someone will have some 
insight at least into where or how I can settle the above items down, 
and I can start building a proper dialplan.

Thanks,

Doc