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<p class="MsoNormal"><span style="mso-fareast-language:EN-US">Hello Micah,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">using Kamailio as front-end/balancer for one or more asterisk instance(s) is a classic use case for Kamailio.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">Have a look to the Asterisk log why you get some authentication request, probably you need to “tell” Asterisk to trust the Kamailio (IPs).<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">Cheers,<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">Henning<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">-- <o:p>
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<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">Henning Westerholt –
</span><span style="mso-fareast-language:EN-US"><a href="https://skalatan.de/blog/"><span lang="EN-GB" style="color:#0563C1">https://skalatan.de/blog/</span></a></span><span lang="EN-GB" style="mso-fareast-language:EN-US"><o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">Kamailio services –
</span><span style="mso-fareast-language:EN-US"><a href="https://gilawa.com/"><span lang="EN-GB" style="color:#0563C1">https://gilawa.com</span></a></span><span style="mso-fareast-language:EN-US">
<span lang="EN-GB"><o:p></o:p></span></span></p>
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<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><b>From:</b> sr-users <sr-users-bounces@lists.kamailio.org>
<b>On Behalf Of </b>Micah Quinn<br>
<b>Sent:</b> Friday, September 10, 2021 1:05 AM<br>
<b>To:</b> sr-users@lists.kamailio.org<br>
<b>Subject:</b> [SR-Users] Kamailio/RTPengine as a proxy for FreePBX/Asterisk...<o:p></o:p></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><span style="font-size:12.0pt;color:black">Hello all,<o:p></o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><span style="font-size:12.0pt;color:black"><o:p> </o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><span style="font-size:12.0pt;color:black">I'm new to Kamailio, so bear with me as I stumble through this. First, I'll describe what I'm trying to achieve at a high level and then perhaps somebody can advise me
on whether Kamailio is a good fit for this solution or not. I'd like to be able to deploy a small appliance type server to our customer's sites that just runs Kamailio and a VPN connection back to our datacenter. At our datacenter, we run virtualized instances
of Asterisk for each of our customers. The idea is that Kamailio would act as a transparent proxy through to the Asterisk instance under nominal conditions and as a basic SIP router in the case that the Asterisk instance is unavailable. This degraded functionality
would then at least allow extension to extension calling even if the Internet or Asterisk instance is down.<o:p></o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><span style="font-size:12.0pt;color:black"><o:p> </o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><span style="font-size:12.0pt;color:black">I'm currently using dispatcher with a single entry in preparation for a time when we might want to failover to another Asterisk instance. I'm forwarding all REGISTER
and INVITE messages to the server chosen from ds_select_dst. Initially this all seems to work as I can register with a softphone and pjsip show endpoints shows my softphone connected. However, when I attempt to call any extension (my own or another) Asterisk
responds to the INVITE message with a "401 Unauthorized" message and the typical "The person at extension XXXX is unavailable...". <o:p></o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><span style="font-size:12.0pt;color:black"><o:p> </o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><span style="font-size:12.0pt;color:black">I know that more details might be necessary to troubleshoot this, but I didn't want to include everything in one post and risk cluttering it up with unnecessary information.
If anyone can confirm that this is a reasonable way to approach the problem, I can then provide whatever relevant data is necessary to get deeper into it. (I've used sngrep, logging, asterisk cli, etc.)<o:p></o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><span style="font-size:12.0pt;color:black"><o:p> </o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><span style="font-size:12.0pt;color:black">Thanks in advance for any help.<o:p></o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><span style="font-size:12.0pt;color:black"> <o:p></o:p></span></p>
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