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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Hello 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'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'>I have being trying to replicate the following call
scenario with my Asterisk box: <a
href="http://www.tech-invite.com/Ti-sip-service-8.html">http://www.tech-invite.com/Ti-sip-service-8.html</a>
<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'>I have a situation that I have to notify the calling
party that the call is being forwarded to another number. So far, in the tests
that I made, calling from a SIP extension to another SIP extension with the
forwarding activated, I get only the <b><span style='font-weight:bold'>SIP
Response 302 (<tt><font face=Arial><span style='font-family:Arial'>MOVED_TEMPORARILY)</span></font></tt></span></b><tt><font
face=Arial><span style='font-family:Arial'> instead of the <b><span
style='font-weight:bold'>SIP Response 181 CALL_IS_BEING_FORWARDED).</span></b><o:p></o:p></span></font></tt></span></font></p>
<p class=MsoNormal><tt><font size=2 face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial'><o:p> </o:p></span></font></tt></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'>The forwarding of the SIP extensions is being set
with AstDB. <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'>My doubt is if, only a SIP Proxy would be able to trigger
SIP Response 181, or if it would be possible with an Asterisk Server. <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US 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 lang=EN-US style='font-size:
10.0pt;font-family:Arial'> <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoAutoSig><b><font size=2 color="#ff9900" face=Arial><span
lang=EN-US style='font-size:10.0pt;font-family:Arial;color:#FF9900;font-weight:
bold'>Marco Cordeiro<o:p></o:p></span></font></b></p>
<p class=MsoAutoSig><b><font size=2 face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial;font-weight:bold'>mhcordeiro@gmail.com<o:p></o:p></span></font></b></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'><o:p> </o:p></span></font></p>
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