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<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>Try setting ‘qualify=yes’ in the sip.conf for the
users. This will send a SIP options packet every two to the phone to verify the
remote NAT device is allowing traffic from both sources to the phone.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>Afterwards, you’ll usually see this status from the servers,
to verify the phone is reachable:<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>123/123 64.23.49.5 D N 15103 OK (44 ms) <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>If one server is unable to reach the phone, the status will
instead be ‘UNREACHABLE’.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>If it is a NAT device with a stateful firewall, it will likely
only open the port for one source IP, and not both servers. Issues like this
are why I run in an active/standby setup as opposed to active/active.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<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> Wednesday, July 23, 2008 03:40<br>
<b>To:</b> asterisk-users@lists.digium.com<br>
<b>Subject:</b> Re: [asterisk-users] sometimes extensions can't be called<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
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<p class=MsoNormal>Hi,<br>
<br>
I think i notice the problem now, but unfortunately i don't know how to fix
it.<br>
<br>
i'm using 118103 i dial 113102 i got this on asterisk server #1.<br>
<br>
[Jul 23 18:27:48] -- Called 118102<br>
[Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing<br>
<br>
what i did is keep on dialing then hang up dial then hang up, until i
notice that when i dialed it went to asterisk #2 on asterisk 2 i see this:<br>
<br>
[Jul 23 18:30:40] -- Called 118102<br>
<br>
but no ringing, it seems like it's trying to look for it, could it be because
102 is registered only on asterisk #1? but if i execute sip show peers
i can see 118102 on both servers. i also had the problem wherein after i dial
118102, it goes to asterisk #2 and cince there is no ring, i hang up my
phone, then i dialed again this time i see:<br>
<br>
[Jul 23 18:32:47] ERROR[17368]: chan_sip.c:3269 update_call_counter: Call to
peer '118102' rejected due to usage limit of 2<br>
<br>
yup i did set the limit to 2 but there was no asnwer on 118102 and i hangup,
why did i reached the limit?<br>
<br>
Thanks in advanced<br>
<br>
Regards<br>
nhadie<br>
<br>
--- On <b>Wed, 7/23/08, Darryl Dunkin <i><ddunkin@netos.net></i></b>
wrote:<o:p></o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'>From: Darryl Dunkin
<ddunkin@netos.net><br>
Subject: RE: [asterisk-users] sometimes extensions can't be called<br>
To: nhadie.ramos@yahoo.com, asterisk-users@lists.digium.com<br>
Date: Wednesday, July 23, 2008, 5:13 AM<o:p></o:p></p>
<div id=yiv1638160106>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:black'>Are the
users registered to both active servers?</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:black'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:black'>Ħsip
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.</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:black'> </span><o:p></o:p></p>
<div style='border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0in 0in 0in;
border-color:-moz-use-text-color -moz-use-text-color'>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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</span><o:p></o:p></p>
</div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
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<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:
auto'>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>
</td>
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<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Calibri","sans-serif"'> </span><o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'><o:p> </o:p></span></p>
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