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<DIV dir=ltr><FONT face=Arial color=#000000 size=2>Kris</FONT></DIV>
<DIV dir=ltr><FONT face=Arial size=2></FONT> </DIV>
<DIV dir=ltr><FONT face=Arial size=2>Thanks for the advice </FONT></DIV>
<DIV dir=ltr><FONT face=Arial size=2></FONT> </DIV>
<DIV dir=ltr><FONT face=Arial size=2>going to give it a try </FONT></DIV>
<DIV dir=ltr><FONT face=Arial size=2></FONT> </DIV>
<DIV dir=ltr><FONT face=Arial size=2>I think for wan outages I need to
investigate what can be done under SRST. Maybe I can use some sort of
translation or something to at least make sure calls to the main remote site
phone number ends up on an IP phone during the outage.</FONT></DIV>
<DIV dir=ltr><FONT face=Arial size=2></FONT> </DIV></DIV>
<DIV dir=ltr><BR> </DIV>
<DIV dir=ltr>erik</DIV>
<DIV dir=ltr> </DIV>
<DIV dir=ltr>
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</DIV>
<DIV dir=ltr><FONT face=Tahoma size=2><B>From:</B> Kris Seraphine
[mailto:baryonyx5@gmail.com]<BR><B>Sent:</B> Wed 2006-10-18 03:08<BR><B>To:</B>
Erik Erasmus (E)<BR><B>Cc:</B> cisco-voip@puck.nether.net<BR><B>Subject:</B> Re:
[cisco-voip] attendant console in remote SRST branch<BR></FONT><BR></DIV>
<DIV>Hi<BR><BR>I support a centralized cluster with 40+ remote sites; all of
which use the attendant console. Most of the sites are under 50 phones but
a few are over 100. The AC works pretty well over the WAN. The only
problem that we occasionaly have that I attribute to the WAN is that the parked
calls sometimes don't disappear when a call is picked up. The only reason
I attribute this to a latency issue is that I don't see this issue with any of
my other customers who use the AC on the same LAN as the servers. This
customer currently doesn't prioritize the AC flows but that's in the
works. <BR><BR>As for WAN outages, you're pretty much out of
luck. There's no way to preserve the AC functionality when the
connection to the server is down. You can route calls to dns using alias
commands but line state, drag and drop will be gone. <BR><BR>The MPLS WAN has
been very reliable with maybe 6-7 outages during work hours over the last 12
months between all the sites. Only one outage was extended and that was
the local telco's fault. <BR><BR>
<DIV><SPAN class=gmail_quote>On 10/17/06, <B class=gmail_sendername>Erik Erasmus
(E)</B> <<A href="mailto:ErasmuE4@telkom.co.za">ErasmuE4@telkom.co.za</A>>
wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
<DIV>
<DIV><FONT face=Arial color=#000000 size=2>Hi</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I have a scenario now where a customer wants to
run an HQ and 5 remote branches with a centralised call manager cluster and
the branches on srst across an MPLS IP wan. The problem is I have never proven
the viability of running the attendant console that comes with ccm 4-1/4-2
across a wan link. Looking for practical advise on workable
solutions.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>1. Has anyone deployed this with success - how
heavy is attendant console on bandwidth (assume we can configure
propper QoS) - say 100 users in the branch</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>2. My biggest concern is what happens when the
branch goes into srst mode. How does one then cover for the pilot number for
the specific site. Normally when doing it at an HQ I use the approach where I
create the ac1, ac2 users etc for attendants and the pilot dn is the main
number like xxx 1000 etc. if one now configure ac users for the remote site/s
each with their own pilot number things should work Ok with the WAN up but not
sure how to provide cover for the main number when srst kicks in and the
entire attendant solution goes down. The branch can not afford having the main
pilot number un attended during a wan failure.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Not using attendant console also poses a problem
for a big branch with 100 plus users with high inbound call volumes to the
site. Simply putting a 7960 down with an expansion module or two will not do
it. The attendants won't be able to effectively handle high call volumes, no
drag and drop, now line state info etc.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Any advise will be appreciated.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>erik erasmus</FONT></DIV>
<DIV><FONT face=Arial size=2>Telkom SA Ltd</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV></DIV>
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clear=all><BR>-- <BR>kris seraphine </DIV></BODY></HTML>
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