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<p class=MsoNormal><font size=2 color=blue face="Gill Sans MT"><span
style='font-size:10.0pt;font-family:"Gill Sans MT";color:blue'>Hi List –<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=blue face="Gill Sans MT"><span
style='font-size:10.0pt;font-family:"Gill Sans MT";color:blue'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=blue face="Gill Sans MT"><span
style='font-size:10.0pt;font-family:"Gill Sans MT";color:blue'>Well, not
getting anywhere, I stumped up for Digium support, and the answer is,
unfortunately, that there is currently no way of resetting the timer when the Dial
application gets a 302 message back from the SIP handset. In other words, the
behaviour exhibited below is standard (even though in my opinion it is
undesirable).<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=blue face="Gill Sans MT"><span
style='font-size:10.0pt;font-family:"Gill Sans MT";color:blue'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=blue face="Gill Sans MT"><span
style='font-size:10.0pt;font-family:"Gill Sans MT";color:blue'>I’ve
decided to have a crack at a patch for this myself. Will keep you posted, since
I know there are at least a couple of other people out there who have been
having this problem.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=blue face="Gill Sans MT"><span
style='font-size:10.0pt;font-family:"Gill Sans MT";color:blue'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=blue face="Gill Sans MT"><span
style='font-size:10.0pt;font-family:"Gill Sans MT";color:blue'>Mike.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=blue face="Gill Sans MT"><span
style='font-size:10.0pt;font-family:"Gill Sans MT";color:blue'><o:p> </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'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Mike Pollitt<br>
<b><span style='font-weight:bold'>Sent:</span></b> Tuesday, 21 February 2006
4:03 PM<br>
<b><span style='font-weight:bold'>To:</span></b> '<st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName>'<br>
<b><span style='font-weight:bold'>Subject:</span></b> [Asterisk-Users] Dial
timeouts and SIP 302 redirects</span></font><o:p></o:p></p>
</div>
<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="Gill Sans MT"><span lang=EN-AU
style='font-size:10.0pt;font-family:"Gill Sans MT"'>I have SIP handsets which
allow the user to forward a call to another number after a specified interval
of ringing time. On the SwissVoice this is refered to as CFNR (Call Forward on
No Response). What actually happens is that after a specified period of time
(default 15 seconds), the handset sends back a “302 Moved
Temporarily” response to Asterisk.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Gill Sans MT"><span lang=EN-AU
style='font-size:10.0pt;font-family:"Gill Sans MT"'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Gill Sans MT"><span lang=EN-AU
style='font-size:10.0pt;font-family:"Gill Sans MT"'>The problem is that when
Asterisk receives the 302 message, it doesn’t reset the ringing timer in
the Dial command. Let’s say I’ve issued a Dial command such as:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Gill Sans MT"><span lang=EN-AU
style='font-size:10.0pt;font-family:"Gill Sans MT"'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Gill Sans MT"><span lang=EN-AU
style='font-size:10.0pt;font-family:"Gill Sans MT"'>exten =>
_34XX,1,Dial(SIP/fred|20)<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Gill Sans MT"><span lang=EN-AU
style='font-size:10.0pt;font-family:"Gill Sans MT"'>exten =>
_34XX,n,Voicemail(fred)<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Gill Sans MT"><span lang=EN-AU
style='font-size:10.0pt;font-family:"Gill Sans MT"'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Gill Sans MT"><span lang=EN-AU
style='font-size:10.0pt;font-family:"Gill Sans MT"'>What happens is that the
SIP handset rings for the default time of 15 seconds, then sends back the 302
message with the new number to forward to. Asterisk faithfully drops into the
Local context with this number, but after a further 5 seconds of ringing the
new number, the original Dial command exits and proceeds to the next priority,
namely the Voicemail command. <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Gill Sans MT"><span lang=EN-AU
style='font-size:10.0pt;font-family:"Gill Sans MT"'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Gill Sans MT"><span lang=EN-AU
style='font-size:10.0pt;font-family:"Gill Sans MT"'>The problem with this is
that the forwarded number only rings for 5 seconds (or not at all if it takes a
few seconds to actually make the new outgoing call, as can happen often with
cellphones), which is not enough time for them to answer it.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Gill Sans MT"><span lang=EN-AU
style='font-size:10.0pt;font-family:"Gill Sans MT"'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Gill Sans MT"><span lang=EN-AU
style='font-size:10.0pt;font-family:"Gill Sans MT"'>Has anyone else had this
problem, and is there a solution?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Gill Sans MT"><span lang=EN-AU
style='font-size:10.0pt;font-family:"Gill Sans MT"'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Gill Sans MT"><span lang=EN-AU
style='font-size:10.0pt;font-family:"Gill Sans MT"'><o:p> </o:p></span></font></p>
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