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<p class=MsoNormal><span lang=EN-US style='font-size:12.0pt;font-family:"Times New Roman","serif"'>Hi
all, <br>
<br>
I have a problem with my asterisk box and an X100P FXO card. I am able to place
outgoing calls from my SIP phone (Cisco 7940) to any external number using my
PSTN line, but when I call my PSTN line from my cell phone, the Cisco doesn't
ring (and no message appears in the Asterisk CLI). <br>
<br>
</span><span style='font-size:12.0pt;font-family:"Times New Roman","serif"'>Here
are my config files: <br>
<b>zaptel.conf</b> <o:p></o:p></span></p>
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<p class=MsoNormal><span lang=EN-US style='font-size:12.0pt;font-family:"Times New Roman","serif"'><br>
fxsks=1 <br>
loadzone = be <br>
defaultzone = be <o:p></o:p></span></p>
</td>
</tr>
</table>
</div>
<p class=MsoNormal><span lang=EN-US style='font-size:12.0pt;font-family:"Times New Roman","serif"'><br>
<br>
</span><b><span style='font-size:12.0pt;font-family:"Times New Roman","serif"'>zapata.conf</span></b><span
style='font-size:12.0pt;font-family:"Times New Roman","serif"'> <o:p></o:p></span></p>
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<tr>
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<p class=MsoNormal><span lang=EN-US style='font-size:12.0pt;font-family:"Times New Roman","serif"'><br>
[channels] <br>
context=incoming_calls <br>
usecallerid=yes <br>
hidecallerid=no <br>
immediate=no <br>
<br>
signalling=fxs_ks <br>
callerid = test <123> <br>
echocancel=yes <br>
group=1 <br>
channel=>1 <o:p></o:p></span></p>
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<p class=MsoNormal><span lang=EN-US style='font-size:12.0pt;font-family:"Times New Roman","serif"'><br>
<br>
</span><b><span style='font-size:12.0pt;font-family:"Times New Roman","serif"'>extensions.conf</span></b><span
style='font-size:12.0pt;font-family:"Times New Roman","serif"'> <o:p></o:p></span></p>
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<tr>
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<p class=MsoNormal><span lang=EN-US style='font-size:12.0pt;font-family:"Times New Roman","serif"'><br>
[globals] <br>
<br>
[general] <br>
autofallthrough=yes <br>
<br>
[default] <br>
exten => s,1,Verbose(1|Unrouted call handler) <br>
exten => s,n,Answer() <br>
exten => s,n,Wait(1) <br>
exten => s,n,Playback(tt-weasels) <br>
exten => s,n,HangUp() <br>
<br>
[incoming_calls] <br>
;exten => _X.,1,NoOp() <br>
;exten => _X.,n,Dial(SIP/1000) <br>
;exten => _X.,1,Dial(SIP/1000) <br>
exten => s,1,Dial(SIP/1000,20,tr) <br>
<br>
[outgoing_calls] <br>
exten => _X.,1,NoOp() <br>
exten => _X.,n,Dial(Zap/1/${EXTEN}) <br>
<br>
[internal] <br>
exten => 1000,1,Verbose(1|Extension 1000) <br>
exten => 1000,n,Dial(SIP/1000,30) <br>
exten => 1000,n,HangUp() <br>
<br>
exten => 500,1,Verbose(1|Echo test application) <br>
exten => 500,n,Echo() <br>
exten => 500,n,HangUp() <br>
<br>
[phones] <br>
;include => internal <br>
;include => incoming_calls <br>
include => outgoing_calls <o:p></o:p></span></p>
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<p class=MsoNormal><span lang=EN-US style='font-size:12.0pt;font-family:"Times New Roman","serif"'><br>
<br>
</span><b><span style='font-size:12.0pt;font-family:"Times New Roman","serif"'>sip.conf</span></b><span
style='font-size:12.0pt;font-family:"Times New Roman","serif"'> <o:p></o:p></span></p>
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<tr>
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<p class=MsoNormal><span lang=EN-US style='font-size:12.0pt;font-family:"Times New Roman","serif"'><br>
[general] <br>
context=phones <br>
bindport=5060 <br>
bindaddr=0.0.0.0 <br>
srvlookup=yes <br>
<br>
[1000] <br>
type=friend <br>
context=phones <br>
host=dynamic <br>
username=1000 <br>
secret=1000 <br>
allow=all <o:p></o:p></span></p>
</td>
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</div>
<p class=MsoNormal><span lang=EN-US style='font-size:12.0pt;font-family:"Times New Roman","serif"'><br>
<br>
zttool shows that the card is working fine and is configured. When I use ztcfg I
get no errors. I've been looking for this for days now and it's driving me
nuts. I read somewhere that you may have to disable AUDIO_RINGCHECK in the
wctdm driver, but I'm using the wcfxo (listed in lsmod together with zaptel). <br>
<br>
</span><span style='font-size:12.0pt;font-family:"Times New Roman","serif"'>All
help is appreciated! <br>
<br>
Thanks, <br>
<br>
Tom</span><o:p></o:p></p>
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