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<p class=MsoNormal style='margin-left:35.4pt'><font size=3
face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'> </span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'>Hello</span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'> </span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'>I’m trying to
implement call forwarding unconditional redirect to a PSTN destination. </span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'>Call flow: PSTN</span></font><font
size=2 face=Wingdings><span lang=EN-US style='font-size:10.0pt;font-family:
Wingdings'>à</span></font><font size=2 face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial'>SIP-CallFW</span></font><font
size=2 face=Wingdings><span lang=EN-US style='font-size:10.0pt;font-family:
Wingdings'>à</span></font><font size=2 face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial'>PSTN</span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'>On my PSTN side I have a
cisco AS5300. We succesfuly forwarded to a PSTN destination.</span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'>We have changed calling
party (for billing purposes) and called party to complete call forwarding. In
other words we generate a new call attempt.</span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'>Everything looks ok but
if calling party (PSTN originating side) finishes the call before called party
answers (PSTN destination side), cisco GW receives a CANCEL message and it
answers with message 481 Call leg/Transaction does not exist to the Proxy. </span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'>The call still ringing
and goes to voice mail on PSTN destination side that means cisco GW never
receives a BYE regarding to the original INVITE message.</span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'>Can you give me a hint on
how to handle this issue ?</span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'>Thanks</span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'> </span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'>Captured attached</span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'> </span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'> </span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'> </span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'> </span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'> </span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial'> </span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>Ariadne L. Ramos Solís</span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>Depto. Ingeniería</span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>Aprovisionamiento y Señalización
VoIP</span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>Galaxy Communications Corp.</span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>Tel. 2000117</span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>e-mail <a
href="mailto:aramos@clarocom.com">aramos@clarocom.com</a></span></font></p>
<p class=MsoNormal style='margin-left:35.4pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'> </span></font></p>
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