<html><head><meta http-equiv="Content-Type" content="text/html; charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class="">Hi Daniel, <div class=""><br class=""></div><div class="">I would be very grateful for your assessment and a little directional advice.</div><div class=""><br class=""></div><div class="">Many thanks!</div><div class="">Philipp<br class=""><div class="">
<meta charset="UTF-8" class=""><div><br class=""></div></div><div><blockquote type="cite" class=""><div class="">Am 11.08.2021 um 17:59 schrieb Philipp Trenz <<a href="mailto:mail@philipptrenz.de" class="">mail@philipptrenz.de</a>>:</div><br class="Apple-interchange-newline"><div class=""><meta http-equiv="content-type" content="text/html; charset=utf-8" class=""><div dir="auto" class=""><div dir="ltr" class="">Hi Daniel, </div><div dir="ltr" class=""><br class=""></div><div dir="ltr" class="">thanks for your quick reply and your questions!</div><div dir="ltr" class=""><br class=""></div><div dir="ltr" class="">What I‘m aiming for is a SaaS solution where users can build audio on demand flows (like Podcasts, Audio Books, Audio guides etc.) via phone. Currently, they bring their VoIP providers, which require registering with username and password. I also wasn’t aware that usually it works via trusted IPs, thanks for that hint. Already learning a lot here!</div><div dir="ltr" class=""><br class=""></div><div dir="ltr" class="">As our users usually want to have a local phone number to promote, I think also in the future we will not go with blocks of numbers. But as of now I cannot say for sure.</div><div dir="ltr" class=""><br class=""></div><div dir="ltr" class="">I think outgoing will be a feature we want to provide later on, e.g. to be connected to someone.</div><div dir="ltr" class=""><br class=""></div><div dir="ltr" class="">Thanks again!</div><div dir="ltr" class=""><br class=""></div><div dir="ltr" class="">Philipp</div><div dir="ltr" class=""><br class=""></div><div dir="ltr" class=""><blockquote type="cite" class="">On 11. Aug 2021, at 12:28, Daniel-Constantin Mierla <<a href="mailto:miconda@gmail.com" class="">miconda@gmail.com</a>> wrote:<br class=""><br class=""></blockquote></div><blockquote type="cite" class=""><div dir="ltr" class="">
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8" class=""><p class="">Hello,</p><p class="">couple of questions to see where to guide you:</p><p class=""> * are the VoIP providers requesting you to register to their
SIP servers? The usual interconnect for a proxy like Kamailio is
based on trusted IP addresses, saving roundtrips for traffic
authentication using user/password. It is also easier to protect
with firewalls and safer when employees leave the company (can't
take IP with them, as opposite of knowing the user/pass).</p><p class=""> * do you have a block of numbers and asterisk does a single
registration to each provider, or right now the asterisk register
all the numbers to providers?</p><p class=""> * now you mention having incoming calls only, do you plan to
support outgoing calls as well?</p><p class="">Cheers,<br class="">
Daniel<br class="">
</p>
<div class="moz-cite-prefix">On 11.08.21 11:50, Philipp Trenz wrote:<br class="">
</div>
<blockquote type="cite" cite="mid:6189E7BE-E6EF-498C-9FF7-DEE4FF0FD19A@philipptrenz.de" class="">
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8" class="">
Hey there,
<div class=""><br class="">
</div>
<div class="">I’m Philipp, an IT Systems Engineer from Potsdam,
Germany and want to deepen my knowledge in SIP communication on
scale.</div>
<div class=""><br class="">
</div>
<div class="">Currently, my setup is quite simple: A single
stateless Asterisk instance, fully managed via ARI, which
registers to multiple VoIP providers, only processes incoming
calls to play audio files (no outgoing or conference calls).</div>
<div class=""><br class="">
</div>
<div class="">Now, with growing load on the single Asterisk
instance, I would like to have one or two Kamailio instances,
which will balance the load of incoming calls to multiple
Asterisk instances. On which Asterisk instance they end up is
irrelevant for me, as the calls can be processed on every
instance.</div>
<div class=""><br class="">
</div>
<div class="">I looked into some Kamailio and Astricon conference
speeches and checked out some tutorials. But still I’m not quite
sure where to start. So here are some basic questions which
would really help me out to get going:</div>
<div class=""><br class="">
</div>
<div class="">
<ul class="MailOutline">
<li class="">As I don’t want to have registrations of all my
Asterisk instances at the VoIP providers, I should use the
UAC module to let Kamailio do the registering to the VoIP
providers and register my Asterisk instances to Kamailio,
right?</li>
<li class="">Will Kamailio automatically apply new outbound
registrations when added to the database or do I have to
trigger that manually?</li>
<li class="">Should I use one or two instances of Kamailio,
how hard is it to configure them fail-safe?</li>
<li class="">For writing my first config file, should I start
blank or from the standard config, what’s best practice?</li>
</ul>
<div class=""><br class="">
</div>
<div class="">Thanks in advance, looking forward to your
replies!</div>
<div class=""><br class="">
</div>
<div class="">Philipp</div>
<br class="">
</div>
<br class="">
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</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></pre>
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