<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=us-ascii">
<META content="MSHTML 6.00.2900.2627" name=GENERATOR></HEAD>
<BODY>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><SPAN class=406181817-18052005><FONT face=Arial size=2>I am still trying to
determine how I can tell within the dial plan which SIP phone an external call
terminates at when the call goes into a queue.</FONT></SPAN></DIV>
<DIV><SPAN class=406181817-18052005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=406181817-18052005><FONT face=Arial size=2>While trying to work
on that piece, I found something else:</FONT></SPAN></DIV>
<DIV><SPAN class=406181817-18052005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=406181817-18052005><FONT face=Arial size=2>When I use
AddQueueMember and RemoveQueueMember, the ASA stats from the CLI command "show
queues" is always at 0%. When I log into a queue from the SIP phone, the ASA
stats will show a percentage.</FONT></SPAN></DIV>
<DIV><SPAN class=406181817-18052005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=406181817-18052005><FONT face=Arial size=2>Is the ASA not
working correctly when I dynamically enter the queue as an
agent?</FONT></SPAN></DIV>
<DIV><SPAN class=406181817-18052005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=406181817-18052005><FONT face=Arial size=2>I am using * 1.0.7
on FC1.</FONT></SPAN></DIV>
<DIV><SPAN class=406181817-18052005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=406181817-18052005><FONT face=Arial size=2>Here is a snippet of
the dial plan:</FONT></SPAN></DIV>
<DIV><SPAN class=406181817-18052005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New" size=2>exten =>
2832,1,Wait,1<BR>exten => 2832,2,Answer<BR>exten =>
2832,3,Playback(vm-goodbye) ;just plays a message to hear
audio quality during test<BR>exten => 2832,4,NoOp("Test") ;
CLI notification - no real reason this is in here<BR>exten =>
2832,5,Queue(inbound-sip)</FONT></SPAN></DIV>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New"
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New" size=2>At this
point, I have no control over the call from the dial plan. What I would like to
do is know, within the dial plan, where the call ended up in the queue. At that
point, I can do this:</FONT></SPAN></DIV>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New"
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New"
size=2>DBget(USERx_IP=SIP/Registry/${EXTEN})<BR>Cut(USERx_IP=USERx_IP,:,1)<BR>System(/bin/echo
-n -e "</FONT><A title="mailto:'@CALL${CALLERIDNAME"
href="BLOCKED::mailto:'@CALL${CALLERIDNAME"><FONT face="Courier New"
size=2>'@CALL${CALLERIDNAME</FONT></A><FONT face="Courier New" size=2>}
${CALLERIDNUM}'" | /usr/bin/netcat --wait=2 -t ${USERx_IP}
${POPPORT})</FONT></SPAN></DIV>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New"
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New" size=2>When I know
the extension (say for internal, SIP to SIP calls) I can do a screen pop to the
far end with a TCP app listening to port XXXX. I just can't seem to figure out
how to tell where the call went - within the dial plan - when the call enters a
queue.</FONT></SPAN></DIV>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New"
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New"
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New" size=2>My
AddQueueMember and RemoveQueueMember are basic.</FONT></SPAN></DIV>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New"
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New"
size=2>AddMemberQueue</FONT></SPAN></DIV>
<DIV><SPAN class=406181817-18052005>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New" size=2>exten =>
78000,1,Answer<BR>exten => 78000,2,Ringing</FONT></SPAN></DIV>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New" size=2>exten =>
78000,3,Wait(1)<BR>exten =>
78000,4,AddQueueMember(inbound-sip|SIP/${CALLERIDNUM})</FONT></SPAN></DIV>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New" size=2><SPAN
class=406181817-18052005><FONT face="Courier New" size=2>exten =>
78000,5,Playback(agent-loginok)</FONT></SPAN></FONT></SPAN></DIV>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New" size=2><SPAN
class=406181817-18052005><SPAN class=406181817-18052005><FONT face="Courier New"
size=2>exten => 78000,6,Wait(1)</FONT></SPAN></SPAN></FONT></SPAN></DIV>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New" size=2><SPAN
class=406181817-18052005><SPAN class=406181817-18052005><SPAN
class=406181817-18052005><FONT face="Courier New" size=2>exten =>
78000,7,Hangup</FONT></SPAN></SPAN></SPAN></FONT></SPAN></DIV>
<DIV><SPAN class=406181817-18052005><FONT face="Courier New" size=2><SPAN
class=406181817-18052005><SPAN class=406181817-18052005><SPAN
class=406181817-18052005></SPAN></SPAN></SPAN></FONT></SPAN> </DIV><SPAN
class=406181817-18052005><FONT face="Courier New" size=2><SPAN
class=406181817-18052005><SPAN class=406181817-18052005><SPAN
class=406181817-18052005></SPAN></SPAN></SPAN></FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=406181817-18052005><SPAN class=406181817-18052005><SPAN
class=406181817-18052005><SPAN class=406181817-18052005><SPAN
class=406181817-18052005></SPAN></SPAN></SPAN>
<DIV><SPAN class=406181817-18052005></SPAN><FONT face="Courier New"
size=2>R<SPAN class=406181817-18052005>emoveMemberQueue does the same thing
except remove the agent.</SPAN><BR></FONT></DIV>
<DIV><FONT face="Courier New" size=2></FONT> </DIV>
<DIV><FONT face="Courier New"><SPAN class=406181817-18052005><FONT face=Arial
size=2>I can see the call movement from the manager interface, but that would
require a whole separate administration effort to make sure that not only the
SIP "extensions" are entered appropriately but also the manager entries are
entered and
correct.</FONT></SPAN></DIV></FONT></SPAN></SPAN></DIV></BODY></HTML>