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<DIV dir=ltr align=left><SPAN class=687592618-16012009><FONT face=Arial
color=#0000ff size=2>Yes, this is the first method I tried. The transfer
only works if it is done before a media path is set up to the first box (not
answered by the IVR). If it is answered then transferred, I get a 500
internal server error back from the ITSP and the call dies. I never see
anything hit the second box.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=687592618-16012009><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=687592618-16012009></SPAN> </DIV><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<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>Lenz
Emilitri<BR><B>Sent:</B> Friday, January 16, 2009 10:09 AM<BR><B>To:</B>
Asterisk Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re:
[asterisk-users] How to transfer a call from one AsteriskServerto
another<BR></FONT><BR></DIV>
<DIV></DIV>
<P>I guess you already tried this? </P>
<P><A
href="http://www.voip-info.org/wiki-Asterisk+cmd+Transfer">http://www.voip-info.org/wiki-Asterisk+cmd+Transfer</A></P>
<P>Thanks</P>
<P>l.</P>
<P><BR></P><BR>
<DIV class=gmail_quote>2009/1/16 Paul <SPAN dir=ltr><<A
href="mailto:bulkmail@monafamily.com">bulkmail@monafamily.com</A>></SPAN><BR>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<DIV>
<DIV dir=ltr align=left><SPAN><FONT face=Arial color=#0000ff size=2>I do have
it functioning with Dial(). I was looking for a way to completely
move the call from the first box though. When using Dial() media moves,
but the call is still tied to the first box. In looking at captures when
the call is ended, the first box invites out to the ITSP again, then after
receiving a 200ok sends a bye.</FONT></SPAN></DIV>
<DIV dir=ltr align=left> </DIV>
<DIV dir=ltr align=left><SPAN><FONT face=Arial color=#0000ff size=2>Also while
testing, once the call was up on the second box, I stopped Asterisk on the
first box which kills the call.</FONT></SPAN></DIV>
<DIV dir=ltr align=left> </DIV>
<DIV dir=ltr align=left> </DIV><BR>
<DIV lang=en-us dir=ltr align=left>
<HR>
<FONT face=Tahoma size=2><B>From:</B> <A
href="mailto:asterisk-users-bounces@lists.digium.com"
target=_blank>asterisk-users-bounces@lists.digium.com</A> [mailto:<A
href="mailto:asterisk-users-bounces@lists.digium.com"
target=_blank>asterisk-users-bounces@lists.digium.com</A>] <B>On Behalf Of
</B>Lenz Emilitri<BR><B>Sent:</B> Friday, January 16, 2009 12:17
AM<BR><B>To:</B> Asterisk Users Mailing List - Non-Commercial
Discussion<BR><B>Subject:</B> Re: [asterisk-users] How to transfer a call from
one AsteriskServer to another<BR></FONT><BR></DIV>
<DIV>
<DIV class=Wj3C7c>
<P>Why don't you simply Dial() the call to a separate box keeping Asterisk out
of the audio path?</P>
<P>l.<BR></P><BR>
<DIV class=gmail_quote>2009/1/16 Paul <SPAN dir=ltr><<A
href="mailto:bulkmail@monafamily.com"
target=_blank>bulkmail@monafamily.com</A>></SPAN><BR>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<DIV>
<DIV><FONT face=Arial size=2>
<DIV><SPAN><FONT face=Arial size=2>Can anyone tell me how I can completely
move an established call off of one Asterisk server to
another?</FONT></SPAN></DIV>
<DIV> </DIV>
<DIV><SPAN><FONT face=Arial size=2>In our case we have a server with our
IVR. Depending upon digits entered, the call can be transferred to any
of our other servers depending where the extension or queue
reside.</FONT></SPAN></DIV>
<DIV><SPAN><FONT face=Arial size=2>We would like to completely move the call
off of the first box so we don't tie up resources on it.</FONT></SPAN></DIV>
<DIV> </DIV>
<DIV><SPAN><FONT face=Arial size=2>In our lab we are testing with
1.4.22.1</FONT></SPAN></DIV>
<DIV> </DIV>
<DIV><SPAN><FONT face=Arial size=2>Our provider which delivers inbound calls
to us uses a Sonus gateway. So far, testing has shown that if we
transfer the inbound call prior to any media playback, it works. But,
if the IVR plays media, then it is failing, with a 500 internal server error
being returned.</FONT></SPAN></DIV>
<DIV> </DIV>
<DIV><SPAN><FONT face=Arial size=2>Thanks for any
help</FONT></SPAN></DIV></FONT></DIV>
<DIV> </DIV>
<DIV align=left> </DIV>
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