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<DIV dir=ltr align=left><SPAN class=348040406-16092005><FONT face=Arial
color=#0000ff size=2>Hi David,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=348040406-16092005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=348040406-16092005><FONT face=Arial
color=#0000ff size=2>I've got probably the same/a similar problem. Do you
add the phones to the queue (AgentLogin/AddQueueMember)?</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=348040406-16092005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=348040406-16092005><FONT face=Arial
color=#0000ff size=2>If there are entries like: " Spawn extension
(macro-dialout-trunk,s,21) exited non-zero..." in your * log file you might have
the same problem like me.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=348040406-16092005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV><SPAN class=348040406-16092005>
<DIV dir=ltr align=left><SPAN class=348040406-16092005></SPAN><FONT face=Arial
color=#0000ff size=2>I<SPAN class=348040406-16092005> suspect that something
goes wrong with the nested macro calls within the AMP-generated dialplan, so
what I did was to expand macro-dial etc.. for each local SIP extension. This
seems to work, but is not really nice because you will have to configure the
extensions manually (which makes AMP more or less
obsolete)...</SPAN><BR></FONT></DIV></SPAN>
<DIV dir=ltr align=left><SPAN class=348040406-16092005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=348040406-16092005><FONT face=Arial
color=#0000ff size=2>cheers</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=348040406-16092005><FONT face=Arial
color=#0000ff size=2>Jörg</FONT></SPAN></DIV><BR>
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<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>David F.
Bakker<BR><B>Sent:</B> Thursday, September 15, 2005 5:28 PM<BR><B>To:</B>
asterisk-users@lists.digium.com<BR><B>Subject:</B> [Asterisk-Users]
Transfering from a device to a queue crashesAsterisk<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV>Asterisk crashes with no errors when I transfer from a device (my
phone) to a queue Asterisk crashes with no errors. Also if I xfer from a
sip device to another and dont wait for the other user to pickup before
xfering the call gets dropped. Any ideas? Im using the latest cvs of asterisk,
amp 1.10.009 and our phones are polycom 501.</DIV>
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