No subject

Sun Jul 19 19:54:31 CDT 2009

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.

------=_NextPart_000_010C_01CB6EAA.3AC2C610
Content-Type: text/html;
	charset="us-ascii"
Content-Transfer-Encoding: quoted-printable

<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
xmlns:w=3D"urn:schemas-microsoft-com:office:word" =
xmlns:st1=3D"urn:schemas-microsoft-com:office:smarttags" =
xmlns=3D"http://www.w3.org/TR/REC-html40">

<head>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<meta name=3DGenerator content=3D"Microsoft Word 11 (filtered medium)">
<!--[if !mso]>
<style>
v\:* {behavior:url(#default#VML);}
o\:* {behavior:url(#default#VML);}
w\:* {behavior:url(#default#VML);}
.shape {behavior:url(#default#VML);}
</style>
<![endif]--><o:SmartTagType
 namespaceuri=3D"urn:schemas-microsoft-com:office:smarttags" =
name=3D"PersonName"/>
<!--[if !mso]>
<style>
st1\:*{behavior:url(#default#ieooui) }
</style>
<![endif]-->
<style>
<!--
 /* Font Definitions */
 @font-face
	{font-family:Tahoma;
	panose-1:2 11 6 4 3 5 4 4 2 4;}
 /* Style Definitions */
 p.MsoNormal, li.MsoNormal, div.MsoNormal
	{margin:0in;
	margin-bottom:.0001pt;
	font-size:12.0pt;
	font-family:"Times New Roman";}
a:link, span.MsoHyperlink
	{color:blue;
	text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
	{color:blue;
	text-decoration:underline;}
span.EmailStyle17
	{mso-style-type:personal-reply;
	font-family:Arial;
	color:navy;}
@page Section1
	{size:8.5in 11.0in;
	margin:1.0in 1.25in 1.0in 1.25in;}
div.Section1
	{page:Section1;}
-->
</style>

</head>

<body lang=3DEN-US link=3Dblue vlink=3Dblue>

<div class=3DSection1>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<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>Rizwan Hisham<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, October 18, =
2010
9:43 AM<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]
clustering</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>&nbsp;</o:p></span></font></p>

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. 
 
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>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>On Mon, Oct 18, 2010 at 4:39 PM, Gareth Blades &lt;<a
href=3D"mailto:list-asterisk at skycomuk.com">list-asterisk at skycomuk.com</a>=
&gt;
wrote:<o:p></o:p></span></font></p>

Use camailio or opensips as the registrar server so it accepts =
the sip 
registrations. You can have copies running on a couple of boxes =
using 
either a shared databases or a database on each server configured in 
master-master replication mode. Opensips can be configured to use =
the 
same database table that asterisk uses for authentication. Then you =
can 
use the load balancer module to send the call to whichever asterisk =
box 
has the most free lines. 
Normally you try and use opensips for most things such as call =
routing 
and registrations and leave asterisk to do the application type =
stuff 
such as conference calls and voicemail.<o:p></o:p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><br>
&lt;snip&gt;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>From what I have read, Asterisk and
Kamalio can co-exist;&nbsp; some posters have used Kamalio to =
&#8220;supplement&#8221;
the features that Asterisk either doesn&#8217;t provide or doesn&#8217;t
provide in as nice a form as the OP desired &#8211; can&#8217;t really =
speak
beyond this as I am not one of them.<o:p></o:p></span></font></p>

</div>

</div>

</div>

</body>

</html>

------=_NextPart_000_010C_01CB6EAA.3AC2C610--