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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Please post your BOARD.INI file (configuration of AudioCodes).<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Also, do you expect to do single-stage dialing (MP104 takes SIP
invite information and turns that into DTMF output), or two-stage dialing
(MP104 only answers and connects the audio/RTP path)?<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>John Balogh, Sr. Systems Engineer<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>PSU, ITS, TNS, Network Planning<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>sip:jdb@psu.edu<o:p></o:p></span></p>
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color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Daniel -
Asterisk<br>
<b>Sent:</b> Wednesday, December 02, 2009 12:33 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> [asterisk-users] Help configuring Audiocodes MP-104 FXO<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'>Hi list,<br>
<br>
I'm trying to get ready the MP-104 FXO to use qith my box, but when I send
calls I hear only dial tone and after a few seconds I get busy signal.<br>
<br>
I very appreciate your advices.<br>
<br>
Command line results and SIPconfigurations follows:<br>
<br>
<b>CLI></b><br>
-- Executing [7991696900@total:1]
Playback("SIP/101-09dd8918", "beep") in new stack<br>
-- <SIP/101-09dd8918> Playing 'beep' (language 'es')<br>
-- Executing [7991696900@total:4]
Dial("SIP/101-09dd8918", "SIP/201/991696900") in new stack<br>
-- Called 201/991696900<br>
-- SIP/201-09ddc890 answered SIP/101-09dd8918<br>
<br>
<br>
<b>sip.conf</b><br>
[201]<br>
secret = ****<br>
callerid = Mobile_01 <201><br>
type = friend<br>
host = dynamic<br>
context = total<br>
dtmfmode=rfc2833<br>
qualify = yes<br>
call-limit=5<br>
disallow = all<br>
allow = gsm<br>
allow = ulaw<br>
allow = alaw<br>
allow = g729<o:p></o:p></p>
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