major undertaking. But since you are using an AGI to control the Queue
command instead of using it from the dialplan, you have more control over
this problem than you realize. For simplicity of illustration, let's say
your AGI simply wants to take a call and send it to the next agent in the
queue. Your Agents are Agent007, AgentQ and AgentM. Because you did the
Polycom transfer from Agent007 to pussygalore, Agent007 is marked as busy in
the queue although the call is no longer active for 007. One possible
workaround would be to have a duplicate "bail queue" set up the same
way.
If my AGI does a "core show channels" and sees that 007 is not on the
phone,
I can do queue(bail) instead of queue(normal).
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style=3D'text-align:center'><font size=3D3
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<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>
</span></font></div>
<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font
size=3D2
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'> asterisk-users-bounces
at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] <b><span
style=3D'font-weight:
bold'>On Behalf Of </span></b>vip killa<br>
<b><span
style=3D'font-weight:bold'>Sent:</span></b> Wednesday,
February 23, 2011
12:44 PM<br>
<b><span
style=3D'font-weight:bold'>To:</span></b>
<st1:PersonName w:st=3D"on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><br>
<b><span
style=3D'font-weight:bold'>Subject:</span></b> Re:
[asterisk-users]
REFER and dialplan broken (as documented inchan_sip.c on line
11951)</span></font><o:p></o:p></p>
</div>
<p class=3DMsoNormal><font size=3D3 face=3D"Times New
Roman"><span style=3D'font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=3DMsoNormal><font size=3D3 face=3D"Times New
Roman"><span style=3D'font-size:
12.0pt'>I recognize all the options given yet as I explained before they
are
not viable. I do not have the resources to pay someone, I do not have the
expertise to fix this issue because according to an asterisk developer
"any fix in that area would be deeply architectural in
nature"...
what other options are
there?<o:p></o:p></span></font></p>
<div>
<p class=3DMsoNormal><font size=3D3 face=3D"Times New
Roman"><span style=3D'font-size:
12.0pt'><o:p> </o:p></span></font></p>
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<p class=3DMsoNormal><font size=3D3 face=3D"Times New
Roman"><span style=3D'font-size:
12.0pt'><snip><o:p></o:p></span></font></p>
<p class=3DMsoNormal><font size=3D3 face=3D"Times New
Roman"><span style=3D'font-size:
12.0pt'>From what I see, the “source fix” on the
Asterisk level
would indeed be a major undertaking. But since you are using an AGI to
control the Queue command instead of using it from the dialplan, you have more
control over this problem than you realize. For simplicity of
illustration, let’s say your AGI simply wants to take a call and send
it
to the next agent in the queue. Your Agents are Agent007, AgentQ and
AgentM. Because you did the Polycom transfer from Agent007 to
pussygalore, Agent007 is marked as busy in the queue although the call is no
longer active for 007. One possible workaround would be to have a duplicate
“bail
queue” set up the same way. If my AGI does a “core
show
channels” and sees that 007 is not on the phone, I can do queue(bail)
instead of
queue(normal).<o:p></o:p></span></font></p>
</div>
<p class=3DMsoNormal><font size=3D3 face=3D"Times New
Roman"><span style=3D'font-size:
12.0pt'><o:p> </o:p></span></font></p>
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