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<DIV><FONT face=Arial size=2>What you can do is have the called person press a
digit to accept the call. If the user dosent then you can set the h extension to
update the call logs</FONT></DIV>
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<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=eric.jacksch@tenebris.ca href="mailto:eric.jacksch@tenebris.ca">Eric
Jacksch</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 Mailing List -
Non-Commercial Discussion</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Wednesday, December 20, 2006 11:47
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [asterisk-users] Follow-me
challenge</DIV>
<DIV><BR></DIV><FONT face="Verdana, Helvetica, Arial"><SPAN
style="FONT-SIZE: 12px">Sorry, didn’t realize you were sending the call out on
a Zap channel.<BR><BR>Yes, as soon as the call goes out a Zap channel it is
“answered” as far as Asterisk is concerned. I send out all my findme
traffic via SIP.<BR><BR><BR>On 2006-12-19 21:19, "Chris Johnson" <<A
href="mailto:masterof1@gmail.com">masterof1@gmail.com</A>>
wrote:<BR><BR></SPAN></FONT>
<BLOCKQUOTE><FONT face="Verdana, Helvetica, Arial"><SPAN
style="FONT-SIZE: 12px">On 12/18/06, <B>Eric Jacksch</B> <<A
href="mailto:eric.jacksch@tenebris.ca">eric.jacksch@tenebris.ca</A>>
wrote:<BR></SPAN></FONT>
<BLOCKQUOTE><FONT face="Verdana, Helvetica, Arial"><SPAN
style="FONT-SIZE: 12px">Is the problem just when you don't answer the cell
phone? Many cell phones go to a voice announcement when they're
turned off or not answered, and Asterisk thinks the call has been
answered. The other issue could be that your gateway (asterisk1) is
answering the call before the outbound leg is answered. One
workaround would be to use a macro that requires you to press a key to
accept the call on your cell. (See the M option to the dial command
and <A
href="http://www.voip-info.org/wiki/view/Asterisk+tips+findme)">http://www.voip-info.org/wiki/view/Asterisk+tips+findme)</A>
<A
href="http://www.voip-info.org/wiki/view/Asterisk+tips+findme%29"><http://www.voip-info.org/wiki/view/Asterisk+tips+findme%29></A>
<BR><BR>Also, I see that you're using the r option — you might want to
drop that.<BR><BR>I'm also not convinced that it will ever find 102,107 in
your dialplan. You might want to look at using ${DIALSTATUS} and
making it a bit more
explicit.<BR><BR>Cheers,<BR>Eric<BR></SPAN></FONT></BLOCKQUOTE><FONT
face="Verdana, Helvetica, Arial"><SPAN style="FONT-SIZE: 12px"><BR>Dropped
the r option and line 107. <BR>The M option had the same result. <BR>Based
on another comment :<BR><BR>Is your other server patching through to a Zap
channel (analog)? <BR>If so, as soon as the dial goes out, an analog Zap
channel is considered<BR>"answered", which could be your
issue.<BR><BR>Doesn't sound like follow-me will work properly with an analog
trunk <BR><BR>
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