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<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>Are the users registered to both active servers?<o:p></o:p></span></p>
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color:black'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>‘sip show peers’ in the console should make this obvious. If users
are to call each other, they both need to be registered to the same server, or
their client needs to be configured to register to both.<o:p></o:p></span></p>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Nhadie
Ramos<br>
<b>Sent:</b> Tuesday, July 22, 2008 21:52<br>
<b>To:</b> asterisk-users@lists.digium.com<br>
<b>Subject:</b> [asterisk-users] sometimes extensions can't be called<o:p></o:p></span></p>
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<p class=MsoNormal>Hi All,<br>
<br>
I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime
on both asterisk. users register via domain, i have that domain on
round-robin. users can register and sometimes can call each other, but
sometimes even if an extension is register and i tried calling it, i got this
on the the cli:<br>
<br>
[Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)<br>
[Jul 23 12:44:52] == Everyone is busy/congested at this time
(1:0/0/1)<br>
<br>
but xlite or ip phone shows the extension is registered. but asterisk says
it's busy. phones are behind NAT and using stun server. sip keep-alive is
enabled onxlite or ip phone. but it's just very inconsistent. i don't know
where to look at to fix this. any idea?<br>
<br>
nhadie<o:p></o:p></p>
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