<html><body><div style="color:#000; background-color:#fff; font-family:times new roman, new york, times, serif;font-size:12pt"><div style="RIGHT: auto"><SPAN style="RIGHT: auto">It is needed to purchase . Any other option that is open source and free ???</SPAN></div>
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<DIV style="BORDER-BOTTOM: #ccc 1px solid; BORDER-LEFT: #ccc 1px solid; PADDING-BOTTOM: 0px; LINE-HEIGHT: 0; MARGIN: 5px 0px; PADDING-LEFT: 0px; PADDING-RIGHT: 0px; HEIGHT: 0px; FONT-SIZE: 0px; BORDER-TOP: #ccc 1px solid; BORDER-RIGHT: #ccc 1px solid; PADDING-TOP: 0px" class=hr readonly="true" contenteditable="false"></DIV><B><SPAN style="FONT-WEIGHT: bold">寄件人︰</SPAN></B> Michelle Dupuis <mdupuis@ocg.ca><BR><B><SPAN style="FONT-WEIGHT: bold">收件人︰</SPAN></B> kingman chui <chuikingman@yahoo.com.hk>; Asterisk Users List <asterisk-users@lists.digium.com> <BR><B><SPAN style="FONT-WEIGHT: bold">傳送日期︰</SPAN></B> 2012年11月16日 (週五) 8:08 AM<BR><B><SPAN style="FONT-WEIGHT: bold">主題︰</SPAN></B> RE: [asterisk-users] 回覆︰ "Simple" failover configuration<BR></FONT></DIV><BR>
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<DIV dir=ltr><FONT color=#000000 size=2 face=Tahoma>Or...you could use HAAST (<A href="http://www.generationd.com/" rel=nofollow target=_blank>http://www.generationd.com/</A>) - it detects failure, switches IP to an Asterisk peer, updates routes, updates ARP tables, synchronizes settings between peers, etc.</FONT></DIV>
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<DIV dir=ltr><FONT size=2 face=tahoma>-M-</FONT></DIV>
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<DIV dir=ltr><FONT size=2 face=tahoma>P.S. I work for generationd, so I think the product is amazing :) </FONT></DIV>
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<DIV style="BORDER-BOTTOM: #ccc 1px solid; BORDER-LEFT: #ccc 1px solid; PADDING-BOTTOM: 0px; LINE-HEIGHT: 0; MARGIN: 5px 0px; PADDING-LEFT: 0px; PADDING-RIGHT: 0px; HEIGHT: 0px; FONT-SIZE: 0px; BORDER-TOP: #ccc 1px solid; BORDER-RIGHT: #ccc 1px solid; PADDING-TOP: 0px" class=hr readonly="true" contenteditable="false"></DIV><FONT size=2 face=Tahoma><B>From:</B> asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of kingman chui [chuikingman@yahoo.com.hk]<BR><B>Sent:</B> Thursday, November 15, 2012 6:34 PM<BR><B>To:</B> Asterisk Users List<BR><B>Subject:</B> [asterisk-users] 回覆︰ "Simple" failover configuration<BR></FONT><BR></DIV>
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<DIV><SPAN>I think you can use virtual IP with a group of sip server that run in HA .</SPAN></DIV>
<DIV style="BACKGROUND-COLOR: transparent; FONT-STYLE: normal; FONT-FAMILY: times new roman, new york, times, serif; COLOR: rgb(0,0,0); FONT-SIZE: 16px"><SPAN>so, only one of sipserver is handle the call and other is standby ...</SPAN></DIV>
<DIV style="BACKGROUND-COLOR: transparent; FONT-STYLE: normal; FONT-FAMILY: times new roman, new york, times, serif; COLOR: rgb(0,0,0); FONT-SIZE: 16px"><SPAN>Is this what you want ...?????</SPAN></DIV>
<DIV style="BACKGROUND-COLOR: transparent; FONT-STYLE: normal; FONT-FAMILY: times new roman, new york, times, serif; COLOR: rgb(0,0,0); FONT-SIZE: 16px"><SPAN>Regard/chui king man</SPAN></DIV>
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<DIV style="BORDER-BOTTOM: #ccc 1px solid; BORDER-LEFT: #ccc 1px solid; PADDING-BOTTOM: 0px; LINE-HEIGHT: 0; MARGIN: 5px 0px; PADDING-LEFT: 0px; PADDING-RIGHT: 0px; HEIGHT: 0px; FONT-SIZE: 0px; BORDER-TOP: #ccc 1px solid; BORDER-RIGHT: #ccc 1px solid; PADDING-TOP: 0px" class=yiv381909470hr></DIV><B><SPAN style="FONT-WEIGHT: bold">寄件人︰</SPAN></B> Danny Nicholas <danny@debsinc.com><BR><B><SPAN style="FONT-WEIGHT: bold">收件人︰</SPAN></B> 'Asterisk Users Mailing List - Non-Commercial Discussion' <asterisk-users@lists.digium.com> <BR><B><SPAN style="FONT-WEIGHT: bold">傳送日期︰</SPAN></B> 2012年11月15日 (週四) 11:31 PM<BR><B><SPAN style="FONT-WEIGHT: bold">主題︰</SPAN></B> Re: [asterisk-users] "Simple" failover configuration<BR></FONT></DIV><BR>You can actually configure at least some Polycom phones to 3 or more SIP<BR>servers. Your problem is going to be that when one of your servers is down<BR>for whatever
reason, the "line key" attached to that server will be "off".<BR>In a "Dual Server" environment, I would lean toward putting something like<BR>Kamailio (sp) in line so it can determine which server is the active one.<BR><BR>-----Original Message-----<BR>From: <A href="mailto:asterisk-users-bounces@lists.digium.com" rel=nofollow target=_blank ymailto="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</A><BR>[mailto:<A href="mailto:asterisk-users-bounces@lists.digium.com" rel=nofollow target=_blank ymailto="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</A>] On Behalf Of Eric Wieling<BR>Sent: Thursday, November 15, 2012 9:27 AM<BR>To: <A href="mailto:chris@acsdi.com" rel=nofollow target=_blank ymailto="mailto:chris@acsdi.com">chris@acsdi.com</A>; Asterisk Users Mailing List - Non-Commercial Discussion<BR>Subject: Re: [asterisk-users] "Simple" failover
configuration<BR><BR>Polycom phones after firmware 2.x register to BOTH the primary and backup<BR>servers. <BR><BR>On Thu, Nov 15, 2012 at 8:59 AM, Chris Nighswonger<BR><<A href="mailto:cnighswonger@foundations.edu" rel=nofollow target=_blank ymailto="mailto:cnighswonger@foundations.edu">cnighswonger@foundations.edu</A>> wrote:<BR><BR> <BR> Would the simplest approach to failover be to just configure my<BR> primary asterisk server as the first SIP server and my backup as the<BR> second?<BR> <BR>--<BR>_____________________________________________________________________<BR>-- Bandwidth and Colocation Provided by <A href="http://www.api-digital.com/" rel=nofollow target=_blank>http://www.api-digital.com/</A> -- New to<BR>Asterisk? Join us for a live introductory webinar every Thurs:<BR> <A
href="http://www.asterisk.org/hello" rel=nofollow target=_blank>http://www.asterisk.org/hello</A><BR><BR>asterisk-users mailing list<BR>To UNSUBSCRIBE or update options visit:<BR> <A href="http://lists.digium.com/mailman/listinfo/asterisk-users" rel=nofollow target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR><BR><BR>--<BR>_____________________________________________________________________<BR>-- Bandwidth and Colocation Provided by <A href="http://www.api-digital.com/" rel=nofollow target=_blank>http://www.api-digital.com/</A> --<BR>New to Asterisk? Join us for a live introductory webinar every Thurs:<BR> <A href="http://www.asterisk.org/hello" rel=nofollow target=_blank>http://www.asterisk.org/hello</A><BR><BR>asterisk-users mailing list<BR>To UNSUBSCRIBE or update options visit:<BR> <A href="http://lists.digium.com/mailman/listinfo/asterisk-users" rel=nofollow
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