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:


Thanks in advance, looking forward to your replies!

Philipp


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