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<DIV><SPAN class=765334410-09052004><FONT color=#0000ff
size=2>Dan,</FONT></SPAN></DIV>
<DIV><SPAN class=765334410-09052004><FONT color=#0000ff size=2> This will
probably not work. Once Asterisk tells the Panasonic PBX to transfer the
call, the call will no longer go through the extension the Asterisk PBX is
attached to. It seems the only solution to this (doing it the way you are)
is to have two zaptel cards in the Asterisk PBX, attache them to two extensions
of the Panasonic PBX and the calls come in one extension and then are sent out
the other KEEPING BOTH BUSY as long as the call is in progress. As Sam
said, you will really need to have an intelligent connection to the Panasonic
PBX to do this on much more than a single call at a time
basis.</FONT></SPAN></DIV>
<DIV><SPAN class=765334410-09052004><FONT color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=765334410-09052004><FONT color=#0000ff size=2>Disclaimer: I am
an Asterisk Newbie, though I have been following it for several years, my hands
on experience is about a week old.</FONT></SPAN></DIV>
<DIV><SPAN class=765334410-09052004><FONT color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=765334410-09052004><FONT color=#0000ff size=2>Andy
Farnsworth</FONT></SPAN></DIV>
<DIV><SPAN class=765334410-09052004><FONT color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV></DIV>
<DIV><FONT face=Tahoma size=2>-----Original Message-----<BR><B>From:</B>
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] <B>On Behalf Of </B>Sam
Bingner<BR><B>Sent:</B> Saturday, May 08, 2004 10:54 PM<BR><B>To:</B>
asterisk-users@lists.digium.com<BR><B>Subject:</B> RE: [Asterisk-Users] x100p /
Answer-> Flash -> Dial<BR><BR></DIV></FONT>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
<DIV><SPAN class=515135221-08052004><FONT face=Arial color=#0000ff size=2>Even
if you could get that to work properly, which I dont know... the callprogress
detection is horrible; if you want to do that reliably you need a T1,ISDN or
IP interface to the switch (something that actually provides proper call
progress)</FONT></SPAN></DIV>
<DIV><SPAN class=515135221-08052004><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=515135221-08052004><FONT face=Arial color=#0000ff
size=2>Sam</FONT></SPAN></DIV>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
<DIV></DIV>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left><FONT
face=Tahoma size=2>-----Original Message-----<BR><B>From:</B>
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] <B>On Behalf Of </B>Dan
Fernandez<BR><B>Sent:</B> Saturday, May 08, 2004 11:44 AM<BR><B>To:</B>
asterisk-users@lists.digium.com<BR><B>Subject:</B> [Asterisk-Users] x100p /
Answer-> Flash -> Dial<BR><BR></FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>I have an X100P connected to an extension of
a Panasonic PBX. When a call from the PSTN comes in, it is
routed directly to the extension where the x100p is . I
want * to answer the call, play a message and then transfer the
call to another extension via the Zap channel where the call was received (I
need to flash the zap channel) . If this extension doesn't answer I want
then to dial an IAX channel.</FONT></DIV>
<DIV><FONT face=Arial size=2>The problem is that when I do a Flash on
the zap channel, and then try to dial a new extension via that zap
channel I get the following error "can't create zap
channel". </FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>If I do a
SendDTMF(XXXX) the call does get transfer to the new
extension but then * gets out of the call loop and don't know it is
answered or not by the new extension.</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>Am I missing something? Why am I getting
the "can't creat za channel"</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>Thanks in advance.</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial
size=2>Dan</FONT> </DIV></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>