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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Rizwan Hisham<br>
<b><span style='font-weight:bold'>Sent:</span></b> Monday, October 18, 2010
9:43 AM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [asterisk-users]
clustering</span></font><o:p></o:p></p>
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12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>Unfortunately we are too
late to switch to Kamailio. I mean we have developed our pbx with call features
and routing on asterisk only. If we switch to some other software that means we
will have to redo a lot of development again. I was thinking of using DUNDi and
distributing the registrations on different servers.<br>
<br>
I just dont get one point. lets say if i have 2 users registered on different
asterisk servers and one of the server fails (dundi doea not get anything in
return from lookup). But I can still get the contact information for the user
who was registered on the failed server from db (realtime peer) for incoming
calls. But what happens when that user tries to make a outgoing call? How do I
redirect the call to the server which is still working?<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'><snip>
<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'>Sorry for
second post, but I have a Polycom 501 registered to 3 servers. I hit the line
button and if the server I pick is down, I don’t get a dial tone. Hope
this is useful.<o:p></o:p></span></font></p>
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