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<DIV><FONT face=Arial size=2>Well after thinking a little more about your
scenario I have had situations where I have hairpinned calls back to the device
they have come in on and in my experience without canreinvite=no the device will
usually complain that a loop was detected. In situations were I am
using SER(SIP EXPRESS ROUTER) I have fixed the problem by
rewriting the headed to trick the device out of the loop detect
scenario. Anyway in theory if the 7206 doesn't detect a loop if you have
the sip.conf setup to the canreinvite=yes then the audio channel should not be
proxied by *. </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>In this calling multiple channels how do you deal
with the cellphone voicemail not answering the call immediately if the phone is
off or out of range which wouldn't allow for the opportunity for the other
extension to be answered. </FONT></DIV>
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style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=bbomar@raccoon.com href="mailto:bbomar@raccoon.com">B. J. Bomar</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, February 05, 2004 1:37
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> RE: [Asterisk-Users] Release
phone call</DIV>
<DIV><BR></DIV>
<DIV><SPAN class=147303518-05022004><FONT face=Arial color=#0000ff size=2>The
way we have it setup is simply calling multiple numbers/channels. It is
either setup manually in the configs, or through a very ugly menu interface I
constructed.</FONT></SPAN></DIV>
<DIV><SPAN class=147303518-05022004><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=147303518-05022004><FONT face=Arial color=#0000ff size=2>B.
J.</FONT></SPAN></DIV>
<DIV><SPAN class=147303518-05022004><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=147303518-05022004><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=147303518-05022004><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=147303518-05022004></SPAN> </DIV>
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<DIV></DIV>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left><FONT
face=Tahoma size=2>-----Original Message-----<BR><B>From:</B> <A
href="mailto:asterisk-users-admin@lists.digium.com">asterisk-users-admin@lists.digium.com</A>
[mailto:asterisk-users-admin@lists.digium.com] <B>On Behalf Of </B>Glenn
Dalgliesh<BR><B>Sent:</B> Thursday, February 05, 2004 12:31<BR><B>To:</B>
asterisk-users@lists.digium.com<BR><B>Subject:</B> Re: [Asterisk-Users]
Release phone call<BR><BR></FONT></DIV>
<DIV><FONT face=Arial size=2>I don't really have a answer for you on you
issue but have a question about what "find-me" is. I see it on the feature
list but am unable to find any real information about it. Is this simply
call forward or is their more to it. </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>thanks</FONT></DIV>
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style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=bbomar@raccoon.com href="mailto:bbomar@raccoon.com">B. J.
Bomar</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, February 05, 2004
1:01 PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] Release
phone call</DIV>
<DIV><BR></DIV>
<DIV><SPAN class=653515017-05022004><FONT face=Arial size=2>Hello all, I
am trying to figure out how to have * release a phone call. We are
noticing some call quality issues on people who have a "find-me" feature,
and answer the call through a cell phone. Here is the call path we
are seeing, and all VoIP connections are using SIP.</FONT></SPAN></DIV>
<DIV><SPAN class=653515017-05022004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=653515017-05022004><FONT face=Arial size=2>PSTN --->
Cisco 7206 ---> * Server</FONT></SPAN></DIV>
<DIV><SPAN class=653515017-05022004><FONT face=Arial
size=2> ^-----------|
^-------------|</FONT></SPAN></DIV>
<DIV><SPAN class=653515017-05022004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=653515017-05022004><FONT face=Arial size=2>Hopefully the
diagram makes sense, but in case it doesn't, let me try to explain.
A call comes in from PSTN into our Cisco7206 with PRI card. It then
goes to our * server, which then forwards the call back through the Cisco
to a cell phone on PSTN. I am wanting to have * release the call to
the Cisco once the call is connected. Any thoughts or
ideas?</FONT></SPAN></DIV>
<DIV><SPAN class=653515017-05022004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=653515017-05022004><FONT face=Arial
size=2>Thanks.</FONT></SPAN></DIV>
<DIV><SPAN class=653515017-05022004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=653515017-05022004><FONT face=Arial size=2>B.
J.</FONT></SPAN></DIV>
<DIV><SPAN class=653515017-05022004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=653515017-05022004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=653515017-05022004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN
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