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<p>Hello,</p>
<p>I suggest to run kamailio with debug=3 in kamailio.cfg to see if
the ringing state is published by dialog/pua_dialoginfo during the
call setup.</p>
<p>On the other hand, the dialoginfo is a different event than
user/device presence state, so the endpoints have to subscribe for
both dialog info and presence states to get notifications on both
cases. Eventually you can enhance the config (or code) to generate
presence states on dialog-info events -- iirc, with pua (or
pua_json), you can push any PUBLISH (or presence states).</p>
<p>Cheers,<br>
Daniel<br>
</p>
<div class="moz-cite-prefix">On 11.06.20 18:43, Pover A.V. wrote:<br>
</div>
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<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline"><span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">Summary:
Using Kamailio to load balance between Asterisk servers.
Trouble getting call queuing working because can’t get
device state hints to work reliably.<o:p></o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline;font-variant-ligatures:
normal;font-variant-caps: normal;font-variant-numeric:
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<span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">—<o:p></o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline;font-variant-ligatures:
normal;font-variant-caps: normal;font-variant-numeric:
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initial;word-spacing:0px">
<span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">Hello<o:p></o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline"><span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B"><o:p> </o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline;font-variant-ligatures:
normal;font-variant-caps: normal;font-variant-numeric:
inherit;font-variant-east-asian: inherit;font-stretch:
inherit;line-height:inherit;orphans:
2;text-align:start;widows: 2;-webkit-text-stroke-width:
0px;text-decoration-style: initial;text-decoration-color:
initial;word-spacing:0px">
<span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">I’d
be grateful for any advice that can be offered for this
problem to do with presence and call queueing.<o:p></o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline"><span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B"><o:p> </o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline;font-variant-ligatures:
normal;font-variant-caps: normal;font-variant-numeric:
inherit;font-variant-east-asian: inherit;font-stretch:
inherit;line-height:inherit;orphans:
2;text-align:start;widows: 2;-webkit-text-stroke-width:
0px;text-decoration-style: initial;text-decoration-color:
initial;word-spacing:0px">
<span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">I
have 3 Kamailio (v5.3.4) servers that load-balance incoming
traffic from various sources: a legacy PBX, soft phone
clients, and a SIP trunking service.<o:p></o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline"><span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B"><o:p> </o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline;font-variant-ligatures:
normal;font-variant-caps: normal;font-variant-numeric:
inherit;font-variant-east-asian: inherit;font-stretch:
inherit;line-height:inherit;orphans:
2;text-align:start;widows: 2;-webkit-text-stroke-width:
0px;text-decoration-style: initial;text-decoration-color:
initial;word-spacing:0px">
<span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">Sitting
behind the 3 Kamailio servers are 2 Asterisk (v16 LTS)
servers.<o:p></o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline"><span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B"><o:p> </o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline;font-variant-ligatures:
normal;font-variant-caps: normal;font-variant-numeric:
inherit;font-variant-east-asian: inherit;font-stretch:
inherit;line-height:inherit;orphans:
2;text-align:start;widows: 2;-webkit-text-stroke-width:
0px;text-decoration-style: initial;text-decoration-color:
initial;word-spacing:0px">
<span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">Kamailio
acts as a registrar and user location service. Asterisk is
routing calls and (should be) providing device state
information for each extension.<o:p></o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline"><span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B"><o:p> </o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline;font-variant-ligatures:
normal;font-variant-caps: normal;font-variant-numeric:
inherit;font-variant-east-asian: inherit;font-stretch:
inherit;line-height:inherit;orphans:
2;text-align:start;widows: 2;-webkit-text-stroke-width:
0px;text-decoration-style: initial;text-decoration-color:
initial;word-spacing:0px">
<span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">An
external script builds a list of PJSIP endpoints based on
the content of the Kamailio subscriber table, and deploys it
to the Asterisk servers whenever a change is detected. The
AOR for each record is configured to point to Kamailio. No
authentication details are attached to these endpoints, as
Kamailio is doing authentication before dispatching a
request to Asterisk.<o:p></o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline"><span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B"><o:p> </o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline;font-variant-ligatures:
normal;font-variant-caps: normal;font-variant-numeric:
inherit;font-variant-east-asian: inherit;font-stretch:
inherit;line-height:inherit;orphans:
2;text-align:start;widows: 2;-webkit-text-stroke-width:
0px;text-decoration-style: initial;text-decoration-color:
initial;word-spacing:0px">
<span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">Whenever
Kamailio receives an INVITE, it passes this to an Asterisk
server using the dispatcher module (round-robin algorithm
‘4’); asterisk then processes the request.<o:p></o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline"><span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B"><o:p> </o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline;font-variant-ligatures:
normal;font-variant-caps: normal;font-variant-numeric:
inherit;font-variant-east-asian: inherit;font-stretch:
inherit;line-height:inherit;orphans:
2;text-align:start;widows: 2;-webkit-text-stroke-width:
0px;text-decoration-style: initial;text-decoration-color:
initial;word-spacing:0px">
<span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">At
the moment, everything is working except for presence/device
states and call queueing.<o:p></o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline"><span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B"><o:p> </o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline;font-variant-ligatures:
normal;font-variant-caps: normal;font-variant-numeric:
inherit;font-variant-east-asian: inherit;font-stretch:
inherit;line-height:inherit;orphans:
2;text-align:start;widows: 2;-webkit-text-stroke-width:
0px;text-decoration-style: initial;text-decoration-color:
initial;word-spacing:0px">
<span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">I’ve
tried to set up ‘presence’, ‘pua’, ‘dialog’ and
‘(presence/pua)_dialoginfo’ modules on Kamailio with limited
success. The presence state when a user toggles ‘do not
disturb’ on their soft phone is passed to other users who
are subscribed (watching), however when a user makes a call,
no presence updates are seen by subscribed users (the
extensions show as ‘Idle’ or ‘Available’ instead of ‘In Use’
or ‘Busy’).<o:p></o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline"><span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B"><o:p> </o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline;font-variant-ligatures:
normal;font-variant-caps: normal;font-variant-numeric:
inherit;font-variant-east-asian: inherit;font-stretch:
inherit;line-height:inherit;orphans:
2;text-align:start;widows: 2;-webkit-text-stroke-width:
0px;text-decoration-style: initial;text-decoration-color:
initial;word-spacing:0px">
<span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">Asterisk
is able to report that an extension is ‘Ringing’ or ‘InUse’
when a call is made to that extension, because an INVITE to
an internal extension matches a pattern in the dialplan and
Asterisk routes it to a PJSIP endpoint with the same name.
However, when somebody makes a call, Asterisk is unable to
track the state of the extension making the call because the
call is going via Kamailio so Asterisk is not seeing the
caller as an internal extension. We need a way to make
Asterisk think the call is coming from an internal device if
the caller ID matches a PJSIP endpoint.<o:p></o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline"><span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B"><o:p> </o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline;font-variant-ligatures:
normal;font-variant-caps: normal;font-variant-numeric:
inherit;font-variant-east-asian: inherit;font-stretch:
inherit;line-height:inherit;orphans:
2;text-align:start;widows: 2;-webkit-text-stroke-width:
0px;text-decoration-style: initial;text-decoration-color:
initial;word-spacing:0px">
<span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">Both
Asterisk servers are configured to share device state
updates with each other, and this is working.<o:p></o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline"><span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B"><o:p> </o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline;font-variant-ligatures:
normal;font-variant-caps: normal;font-variant-numeric:
inherit;font-variant-east-asian: inherit;font-stretch:
inherit;line-height:inherit;orphans:
2;text-align:start;widows: 2;-webkit-text-stroke-width:
0px;text-decoration-style: initial;text-decoration-color:
initial;word-spacing:0px">
<span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">Without
being able to see device state for an internal calling
party, we will be unable to reliably set up call queues in
Asterisk.<o:p></o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline"><span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B"><o:p> </o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline;font-variant-ligatures:
normal;font-variant-caps: normal;font-variant-numeric:
inherit;font-variant-east-asian: inherit;font-stretch:
inherit;line-height:inherit;orphans:
2;text-align:start;widows: 2;-webkit-text-stroke-width:
0px;text-decoration-style: initial;text-decoration-color:
initial;word-spacing:0px">
<span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">Using
the ‘mohqueue’ module on Kamailio is not something we want
to do because of its reliance on a database, which is
creating a single point of failure – if the database server
is down for any reason, Kamailio refuses to start.<o:p></o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline"><span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B"><o:p> </o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline;font-variant-ligatures:
normal;font-variant-caps: normal;font-variant-numeric:
inherit;font-variant-east-asian: inherit;font-stretch:
inherit;line-height:inherit;orphans:
2;text-align:start;widows: 2;-webkit-text-stroke-width:
0px;text-decoration-style: initial;text-decoration-color:
initial;word-spacing:0px">
<span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">Ideally,
we would achieve some form of distributed call queueing,
however the simplest option seems to be setting up a queue
on one Asterisk server and having an active-passive high
availability configuration, should a server go down.<o:p></o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline"><span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B"><o:p> </o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline;font-variant-ligatures:
normal;font-variant-caps: normal;font-variant-numeric:
inherit;font-variant-east-asian: inherit;font-stretch:
inherit;line-height:inherit;orphans:
2;text-align:start;widows: 2;-webkit-text-stroke-width:
0px;text-decoration-style: initial;text-decoration-color:
initial;word-spacing:0px">
<span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">I’m
looking for advice on how best to keep track of what is
happening in a call, and for Asterisk to know the states of
each extension and make these available in a dialplan hint.
Or any other suggestions that might help to achieve call
queueing in this setup.<o:p></o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline;font-variant-ligatures:
normal;font-variant-caps: normal;font-variant-numeric:
inherit;font-variant-east-asian: inherit;font-stretch:
inherit;line-height:inherit;orphans:
2;text-align:start;widows: 2;-webkit-text-stroke-width:
0px;text-decoration-style: initial;text-decoration-color:
initial;word-spacing:0px">
<span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B"><o:p> </o:p></span></p>
<p class="1qeiagb0cpwnlhdf9xsijm"
style="margin:0cm;margin-bottom:.0001pt;background:white;vertical-align:baseline"><span
style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">Thank
you<o:p></o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<br>
<fieldset class="mimeAttachmentHeader"></fieldset>
<pre class="moz-quote-pre" wrap="">_______________________________________________
Kamailio (SER) - Users Mailing List
<a class="moz-txt-link-abbreviated" href="mailto:sr-users@lists.kamailio.org">sr-users@lists.kamailio.org</a>
<a class="moz-txt-link-freetext" href="https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users">https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
</blockquote>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla -- <a class="moz-txt-link-abbreviated" href="http://www.asipto.com">www.asipto.com</a>
<a class="moz-txt-link-abbreviated" href="http://www.twitter.com/miconda">www.twitter.com/miconda</a> -- <a class="moz-txt-link-abbreviated" href="http://www.linkedin.com/in/miconda">www.linkedin.com/in/miconda</a>
Funding: <a class="moz-txt-link-freetext" href="https://www.paypal.me/dcmierla">https://www.paypal.me/dcmierla</a></pre>
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