asterisk users - Mar 2005 - No sound when calling in from pstn

I am just starting out with * so bear with me please.

I have tdm400p with 4 fxo modules on it. When I call into the asterisk 
box from my mobile, I can see the asterisk console picks the call up 
and routes it to my computer with x-lite. There was no sound coming 
from either - just silence. I then decided to route it directly to 
voice mail to see if that would narrow the problem down, but it too 
doesn't make any sound. I can see asterisk display what it is saying 
when going through the voicemail sequence, it is just I cannot hear it 
on my mobile.

here is my zapata.conf

context=incomingcall ; from-1800
signalling=fxs_ks
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
group=1
channel=>1

context=incomingcall
signalling=fxs_ks
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=3
channel=>2-3

context=from-fax
signalling=fxs_ks
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=3
channel=>4

Any help is greatly appreciated.

Regards,
Greg

I have just installed * from the latest CVS and I can make calls via X-Lite
to outside numbers but for only 30 to 35 Seconds at a time then Broadvoice
will hang up on me but X-Lite will not know it. Once I hang up X-Lite then I
will get the following error message:

Spawn extension (default, Number-I-Dialed, 1) exited non-zero on
'SIP/200-6a22'
-- Got SIP response 404 "Not Found" back from 147.135.0.128


I have also just installed a TDM22B and I get the same thing when I use a
regular analog phone with it. Call goes through then dropped after about 30
seconds.

Extension to Extension calls work just fine.

Can anyone see anything wrong with my config files? I have tried using a
direct proxy name but I always get a "404 Not found" right away.

Sip.conf
register =>
425XXXXXXX@sip.broadvoice.com:PPPPPPPPPP:425XXXXXXX@sip.broadvoice.com/200

[sip.broadvoice.com]
;type=friend
type=peer
host=sip.broadvoice.com
username=425XXXXXXX
secret=PPPPPPPPPP
fromdomain=sip.broadvoice.com
fromuser=425XXXXXXX
insecure=very
;context=from-broadvoice
context=from-pstn
dtmfmode=inband
canreinvite=no
qualify=yes
user=phone

[200]
type=friend
secret=010101
auth=md5
nat=yes
host=dynamic
reinvite=no
canreinvite=no
dtmfmode=inband
callerid="Fred F"<200>
dissallow=all


Extensions.conf
[default]

exten => 1000,1,Dial,Zap/1|20
exten => 1000,2,Voicemail,u1000
exten => 1000,3,Hangup
exten => 1000,102,Voicemail,b1000
exten => 1000,103,Hangup

exten => 2000,1,Dial,Zap/2|20
exten => 2000,2,Voicemail,u2000
exten => 2000,3,Hangup
exten => 2000,102,Voicemail,b2000
exten => 2000,103,Hangup

exten => _NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30) ; Dial
Broadvoice for 30 seconds
exten => _NXXNXXXXXX, 2, congestion() ; No answer, nothing
exten => _NXXNXXXXXX, 102, busy() ; Busy

> I have tdm400p with 4 fxo modules on it. When I call into the asterisk 
> box from my mobile, I can see the asterisk console picks the call up 
> and routes it to my computer with x-lite. There was no sound coming 
> from either - just silence. I then decided to route it directly to 
> voice mail to see if that would narrow the problem down, but it too 
> doesn't make any sound. I can see asterisk display what it is saying 
> when going through the voicemail sequence, it is just I cannot hear it 
> on my mobile.
> 
> here is my zapata.conf
> 
> context=incomingcall ; from-1800
> signalling=fxs_ks
> faxdetect=incoming
> usecallerid=yes
> echocancel=yes
> echocancelwhenbridged=no
> group=1
> channel=>1
You might consider looking at the 
 /usr/src/asterisk/configs/zapata.conf.samples
and read about rxgain and txgain. Those are parameters to control
the volume.