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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>Bruce B<br>
<b><span style='font-weight:bold'>Sent:</span></b> Wednesday, December 08, 2010
11:33 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] How
to quickly move on to Dahdi channels when SIP provider fails?</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Thanks for the input guys. I really appreciate all the input and I am
sure they work but I thought there would be a much better way to do this.
Sounds like patching things to me. Why doesn't Asterisk take advantage of the
qualify values to make sure if the SIP connection is up or not? Shouldn't this
become a native feature of the PBX rather than trying to do work-around?<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>#1 You’re probably right, but this
should be a bounty or feature request. Is the problem an Asterisk one or a
sip-provider one? Do other PBX products do this kind of failover internally?<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'>#2 You get what you pay for.<o:p></o:p></span></font></p>
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