Hi Daniel,
thanks for your quick reply and your questions!
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!
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.
I think outgoing will be a feature we want to provide later on, e.g. to be connected to
someone.
Thanks again!
Philipp
On 11. Aug 2021, at 12:28, Daniel-Constantin Mierla
<miconda(a)gmail.com> wrote:
Hello,
couple of questions to see where to guide you:
* 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).
* 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?
* now you mention having incoming calls only, do you plan to support outgoing calls as
well?
Cheers,
Daniel
On 11.08.21 11:50, Philipp Trenz wrote:
Hey there,
I’m Philipp, an IT Systems Engineer from Potsdam, Germany and want to deepen my knowledge
in SIP communication on scale.
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).
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.
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:
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?
Will Kamailio automatically apply new outbound registrations when added to the database
or do I have to trigger that manually?
Should I use one or two instances of Kamailio, how hard is it to configure them
fail-safe?
For writing my first config file, should I start blank or from the standard config,
what’s best practice?
Thanks in advance, looking forward to your replies!
Philipp
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