[Asterisk-Users] gnophone and sip phone

[Zen Kato]

Hi,

I installed 'gnophone' on my notePC(RHL9 linux-2.4.20-30.9)
with asterisk CVS-02/05/04. I have three unsolved problems:
(1)call from gnophone to sip phone is OK, but gnophone's 
   speaker volume is very low even though setting highest 
   volume with gmix, the speaker volume is very high. 
   The sip hardphone side: my voice returns back to 
   earphone of handset(echo?).

(2)can not make a call from sip hardphone to gnophone
   *CLI says as follows;
Mar  6 11:27:55 NOTICE[81926]: chan_iax.c:4098 socket_read: Rejected connect attempt from 192.168.0.11, request 's at default' does not exist
Urgent handler
Mar  6 11:27:55 WARNING[81926]: chan_iax.c:3951 socket_read: Call rejected by 192.168.0.11: No such context/extension
Mar  6 11:27:55 NOTICE[81926]: chan_iax.c:1050 iax_destroy: Avoiding IAX destroy deadlock
    -- Called 916 at 192.168.0.11
Urgent handler
    -- Nobody picked up in 5000 ms
    -- Hungup 'IAX[192.168.0.11:5036]/7'
Mar  6 11:28:00 WARNING[294929]: channel.c:517 ast_channel_free: Channel 'IAX[192.168.0.11:5036]/7' may not have been hung up properly
Urgent handler
Mar  6 11:28:10 WARNING[294929]: pbx.c:1834 ast_pbx_run: Timeout, but no rule 't' in context 'sip'
------------- end of *CLI ---------------------

my iax.conf is;
[916]
type=friend
host=dynamic
defaultip=192.168.0.11
port=5036
secret=916
context=default

my extensions.conf is;
[default]
......
exten => 916,1,Dial(IAX/916 at 192.168.0.11,5,r)
.....

What is the meaning of 's at default' above *CLI?
Do I miss something in 'iax.conf'?

(3)When I start 'gnophone', I have to do the following
sequence;
1.start mpg123 some.mp3
2.start 'asterisk'
3.stop mpg123
4.start 'gnophone'

Because, asterisk graps sound device and the others can not
use sound device after asterisk started. How can I release
'sound device' after asterisk started?

----
Zen