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<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: 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>
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<span style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B">Hello<o:p></o:p></span></p>
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<span style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#1A1A1B"><o:p> </o:p></span></p>
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<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>
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<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>
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