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<DIV><FONT face=Arial size=2>Have you looked at features.conf?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Lyle</FONT></DIV>
<BLOCKQUOTE
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=paul.fielding@shaw.ca href="mailto:paul.fielding@shaw.ca">Paul
Fielding</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> Monday, January 17, 2005 8:53
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Asterisk-Users] transfers
with zap channel</DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial size=2>The outside line isn't actually being dropped -
the outside line hanging up is me hanging up the outside line after finding
that my transfer failed.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I must be not understanding how the flash-hook
works then. My understanding was that when I flash-hook and get a second
dialtone I should be able to dial the extention I want to reach (7007 is
another extension, via SIP). Normally, if I pick up the analog phone and
dial 7007 it rings the extention fine. Apparently, though, when
you get that second dialtone, it has different rules? I haven't
been able to find documentation on this, can it be found anywhere? For
example, why does dialing 700 park the call? I haven't found anything on
this... *shrug*...</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Paul</FONT></DIV>
<DIV> </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=lyle@lcrcomputer.net href="mailto:lyle@lcrcomputer.net">Lyle
Giese</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> Monday, January 17, 2005 7:22
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Asterisk-Users] transfers
with zap channel</DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial size=2>How long between getting parked is the orginal
call dropping? </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Depending on your dialplan, yes dialing 700x
will almost immediately send the call to call parking. (IMHO, poor extension
planning can also cause this.)</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I don't use the t or T
options<PERIOD>. IMHO, you just lose the ability to use the #
key and confused the heck out of my users. Took it out and use the
flash method only in my dial plan. Dial 700, park the call. Dial
the other extension, tell them to pick up 701. Or use meetme for
conference calling?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I know I need to play with three way calling
here also.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Lyle</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<BLOCKQUOTE
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=paul.fielding@shaw.ca href="mailto:paul.fielding@shaw.ca">Paul
Fielding</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> Monday, January 17, 2005 6:12
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] transfers
with zap channel</DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial size=2>Ok, I've seen discussion before on doing
transfers (attended and unattended), there seems to be much confusion over
it.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>As things sit, I've been trying
(unsuccessfully) to do transfers with a zap channel (analog phone attached
to TDM400). I have no idea what I'm missing. My current
understanding is that I need to have transfer=yes in zapata.conf, do a
flash hook, dial the 2nd number, flash hook again and we're linked
(attended). Then if I hang up the call will be
transfered.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>However, when I try to do this things don't
work. Here's what I do:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>- connection is made between Zap/3 (analog
phone) and Zap/1 (outside line).</FONT></DIV>
<DIV><FONT face=Arial size=2>- flash hook to get dialtone (I do get
dialtone)</FONT></DIV>
<DIV><FONT face=Arial size=2>- attempt to transfer to extension 7007 - I
dial 7007</FONT></DIV>
<DIV><FONT face=Arial size=2>- after dialing the 2nd zero, and before
dialing the 2nd seven, I hear Asterisk announce (seven - zero - one) and
then Zap/3 is hung up (I get a busy signal). Zap/1 gets
parked.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Here's what the log shows:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> -- Zap/1-1 answered
Zap/3-1<BR> -- Attempting native bridge of Zap/3-1 and
Zap/1-1<BR> -- Started three way call on channel
1<BR> -- Started music on hold, class 'default', on
Zap/3-1<BR> -- Attempting native bridge of Zap/3-1 and
Zap/1-1<BR> -- Starting simple switch on
'Zap/1-2'<BR> -- Started music on hold, class 'default',
on Zap/3-1<BR> == Parked Zap/3-1 on 701. Will timeout back to
dostuff,7001,1 in 45 seconds<BR> -- Added extension
'701' priority 1 to parkedcalls<BR> -- Playing
'digits/7' (language 'en')<BR> -- Hungup
'Zap/1-1'<BR> == Spawn extension (dostuff, 7001, 1) exited non-zero
on 'Parked/Zap/3-1<ZOMBIE>'<BR> -- Stopped music
on hold on Parked/Zap/3-1<ZOMBIE><BR> -- Playing
'digits/0' (language 'en')<BR> -- Playing 'digits/1'
(language 'en')<BR> -- Parking call to
'Zap/1-2'<BR> -- Hungup 'Zap/1-2'<BR>
-- Stopped music on hold on Zap/3-1<BR> == Zap/3-1 got tired of
being parked<BR> -- Hungup 'Zap/3-1'</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I'm not sure what I'm missing.
Apparently something to do with parked calls, so I must be
misunderstanding how do to the call transfer.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I've also tried enabling Asterisk transfer on
the channel (exten => 7010,1,Dial(${CORDLESS},20,tT)).</FONT></DIV>
<DIV><FONT face=Arial size=2>My understanding of this method is that this
allows one to hit the pound (#) to start a transfer. Yet pound does
nothing. Is it fair to assume that the tT only works on SIP
channels, or am I missing something else.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Any help is much appreciated....</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Paul</DIV>
<DIV><BR></DIV></FONT>
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