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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

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10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

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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>&nbsp;</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'>&agrave;</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'>&agrave;</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'>&agrave;</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>&nbsp;</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 &nbsp;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 &nbsp;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>&nbsp;</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>&nbsp;</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>&nbsp;</o:p></span></font></p>

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