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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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<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>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>On Mon, Oct 18, 2010 at 4:39 PM, Gareth Blades <<a
href="mailto:list-asterisk@skycomuk.com">list-asterisk@skycomuk.com</a>>
wrote:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Use camailio or opensips as the registrar server so it accepts the sip<br>
registrations. You can have copies running on a couple of boxes using<br>
either a shared databases or a database on each server configured in<br>
master-master replication mode. Opensips can be configured to use the<br>
same database table that asterisk uses for authentication. Then you can<br>
use the load balancer module to send the call to whichever asterisk box<br>
has the most free lines.<br>
Normally you try and use opensips for most things such as call routing<br>
and registrations and leave asterisk to do the application type stuff<br>
such as conference calls and voicemail.<o:p></o:p></span></font></p>
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<snip><o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>From what I have read, Asterisk and
Kamalio can co-exist; some posters have used Kamalio to “supplement”
the features that Asterisk either doesn’t provide or doesn’t
provide in as nice a form as the OP desired – can’t really speak
beyond this as I am not one of them.<o:p></o:p></span></font></p>
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