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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Background: </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>We are currently implementing an Asterisk based solution for
a customer to enable teleworker phone access. We have connected an Asterisk box
running SUSE 9.3 with an AVM Fritz PCI ISDN card installed running CAPI to a Tenovis
box. Softphones using SIP (referred to as SIP user) have been configured and
can register no problem with Asterisk. The SIP users can call each other with
no problem.</span></font></p>
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font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Problem: </span></font></p>
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font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Incoming calls to the SIP users work fine, but outgoing
calls do not. Outgoing calls ring the called number no problem (dialing using chan_capi
works fine), but when the called number answers, Asterisk does not receive any
notification that the call has been answered, and hence the softphone keeps
ringing. If the hash (#) is pressed on the called phone, the call is then shown
as answered, Asterisk sees it as answered, but there is only oneway voice. The
called party can hear the SIP user, but the SIP user cannot hear the called
party. Asterisk also does not get notification that the call was terminated if
the called party disconnects the call.</span></font></p>
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font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Attempted Solutions:</span></font></p>
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font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>We believe this to be a DTMF problem, but are not sure. We
have tried changing the DTMF in the sip.conf and capi.conf files, but nothing
seems to solve the problem.</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>If anyone has solved this problem, or had any experience
with a similar setup, we would greatly appreciate any assistance.</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Regards to all,</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Joe</span></font></p>
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