asterisk users - Jul 2004 - Asterisk sees inbound call, but won't answer

Good evening,

I am just getting started with Asterisk. I have it installed, and I believe
I am on the right track, overall, to get it working, but I can't get the
linejack to answer any calls.

At this point, all I'm trying to do is have Asterisk answer an inbound call
on my linejack, /dev/phone0, and play a greeting or tone. I figure, once I
am able to get asterisk to actually answer the phone, then I can continue
playing with the system.

Starting asterisk with -vvvvcd gets me to the CLI prompt. This is what
happens on incoming calls:
	Asterisk Ready.
	*CLI> We have caller ID
	Urgent handler

But, the call is never answered. I usually let the phone ring for about 30
seconds, though I did go over a minute once.

Since it is seeing caller ID between the first and second rings (like it
should be), that leads me to think I'm really close to having this work.

Digging around, I discovered that "show channels" is saying there are
0
active channels. The command returns:
	show channels
	        Channel  (Context    Extension    Pri )   State Appl.
Data 0 active channel(s)
	*CLI>

That leads me to think I am missing something that tells asterisk that
/dev/phone0 is a channel.

I've tried various configuration settings, and reading and re-reading
numerous docs and examples from voip-info and other sources.

Any ideas on what I'm missing? I'm including my config files below. 

One more question. Will asterisk treat the linejack as a 2way trunk? Or will
it only work for either inbound or outbound applications? In other words,
while it is idle, will asterisk listen for (and answer) inbound calls or use
it for any outbound calls that need to be made? Or does asterisk have to be
configured to use the linejack as a one way device, either answering inbound
or placing outbound calls.

Many thanks,
Greg

----------------------
asterisk.conf:
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run
astlogdir => /var/log/asterisk
----------------------
modules.conf [this one is tweaked based on what I wanted to play with and
using info from voip-info]:
[modules]
autoload=yes
noload => chan_agent.so
noload => chan_iax2.so
noload => chan_local.so
noload => chan_mgcp.so
noload => chan_modem.so
noload => chan_modem_aopen.so
noload => chan_modem_bestdata.so
noload => chan_modem_i4l.so
noload => chan_oss.so
load => chan_phone.so
noload => chan_sip.so
noload => chan_skinny.so
load => codec_a_mu.so
load => codec_adpcm.so
load => codec_alaw.so
load => codec_g726.so
load => codec_gsm.so
load => codec_ilbc.so
load => codec_lpc10.so
load => codec_ulaw.so
load => format_g726.so
load => format_g729.so
load => format_gsm.so
load => format_h263.so
load => format_ilbc.so
load => format_jpeg.so
load => format_pcm.so
load => format_pcm_alaw.so
load => format_vox.so
load => format_wav.so
load => format_wav_gsm.so
load => cdr_csv.so
load => pbx_config.so
load => pbx_spool.so
load => pbx_wilcalu.so
load => res_adsi.so
load => res_crypto.so
load => res_indications.so
load => res_monitor.so
load => res_musiconhold.so
load => res_parking.so
load => app_adsiprog.so
load => app_agi.so
noload => app_alarmreceiver.so
load => app_authenticate.so
load => app_cdr.so
load => app_chanisavail.so
load => app_controlplayback.so
load => app_cut.so
load => app_db.so
load => app_dial.so
load => app_directory.so
load => app_disa.so
load => app_echo.so
load => app_enumlookup.so
load => app_eval.so
load => app_exec.so
noload => app_festival.so
noload => app_getcpeid.so
load => app_groupcount.so
load => app_hasnewvoicemail.so
noload => app_ices.so
noload => app_image.so
noload => app_intercom.so
load => app_lookupblacklist.so
load => app_lookupcidname.so
load => app_macro.so
load => app_milliwatt.so
load => app_mp3.so
noload => app_nbscat.so
load => app_parkandannounce.so
load => app_playback.so
load => app_privacy.so
load => app_qcall.so
load => app_queue.so
load => app_random.so
load => app_read.so
load => app_record.so
load => app_sayunixtime.so
load => app_senddtmf.so
load => app_sendtext.so
load => app_setcallerid.so
load => app_setcdruserfield.so
load => app_setcidname.so
load => app_setcidnum.so
load => app_sms.so
load => app_softhangup.so
load => app_striplsd.so
load => app_substring.so
load => app_system.so
load => app_talkdetect.so
load => app_transfer.so
load => app_txtcidname.so
load => app_url.so
load => app_userevent.so
load => app_voicemail.so
load => app_waitforring.so
load => app_zapateller.so
;
[global]
chan_phone.so=yes
----------------------
phone.conf:
[interfaces]
mode=fxo
format=slinear
echocancel=medium
;silencesupression=yes
context=linejack
;txgain=100%
;rxgain=1.0
device => /dev/phone0
----------------------
extensions.conf [what I had when I gave up, globals not used yet]:
[general]
static=yes
writeprotect=yes

[globals]
INCOMING => Phone/phone0
OUTGOING => Phone/phone0

[linejack]
exten => s,1,Wait(3)
exten => s,2,Answer()
exten => s,3,Playback(hello)
exten => s,4,Playback(office-iguanas)
exten => h,1,hangup()
----------------------
voicemail.conf:
[general]
format=wav49|gsm|wav
serveremail=greg@netio.org
attach=yes
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
tz=central
saycid=yes
review=yes

[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H
'digits/hundred' M
'hours'

[default]
1234 => 4242,Example Mailbox,root@localhost

[other]
1234 => 5678,Company2 User,root@localhost
----------------------