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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> Habiyakare
Aimable [mailto:aimable@terracom.rw] <br>
<b><span style='font-weight:bold'>Sent:</span></b> Monday, June 07, 2004 11:49
AM<br>
<b><span style='font-weight:bold'>To:</span></b>
'asterisk-users@list.digium.com'; 'gt'; 'support@digium.com'<br>
<b><span style='font-weight:bold'>Subject:</span></b> Problem with Asterisk PRI
forwarding to SER</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Hi all,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I have a problem. We have a phone system setup like this: <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>SIP phones----------</span></font><font size=2
face=Wingdings><span style='font-size:10.0pt;font-family:Wingdings'>à</span></font><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>SER-------------</span></font><font
size=2 face=Wingdings><span style='font-size:10.0pt;font-family:Wingdings'>à</span></font><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>Asterisk--------------</span></font><font
size=2 face=Wingdings><span style='font-size:10.0pt;font-family:Wingdings'>à</span></font><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>PSTN(PRI
connected to NORTEL DES 100 switch)<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>So when there is a call from the pstn through asterisk and the
phone is busy or the number is invalid ,asterisk tells the switch that
the call is going and the phone is ringing while it is not the case. I cant
find a way of informing Asterisk that the call is progressing or something like
that .<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I am using Asterisk CVS-03/22/04-15:45:54 on Red Hat 9
and Sip Express Router version 12 on Red Hat 9<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I tried to use PRI_causes but still the problem is there.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Any idea on how I can solve this problem?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Thanks<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
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