3 Oct
2024
3 Oct
'24
3:22 p.m.
Ihor Olkhovskyi via sr-users:
There are 2 main ways to solve this issue.
1. You are responsible for the project and you are taking care of the
configs. In this case you really need to know what Kamailio and it
config is. Here I'd start with
https://www.asipto.com/sw/kamailio-admin-book/ as a basic for a SIP
protocol and later - how it's processed in Kamailio perspective.
havfo's kamailio config is quite advanced i
Yes thanks, at least I want to know in which direction to learn. route
Relay, sip trunk, or path module.
2. Find someone who will do it on a payment basis.
P.S: Have you tried to specify in your SIP client everything, as you
are connecting to your VoIP provider, but adding WEBRTC-to-SIP
<https://github.com/havfo/WEBRTC-to-SIP> server as outbound proxy?
Yes I tried that. Always get 401 unauthorized. But sip softphone with
same credentials work.
Le mer. 2 oct. 2024 à 20:35, Linux Admin via sr-users
<sr-users(a)lists.kamailio.org> a écrit :
Hi. I am using https://github.com/havfo/WEBRTC-to-SIP to have a web
sipcaller. it uses kamailio + rtpengine + web sipcaller (jssip)
right now it is working using internal kamailo account and can call to
another sip.
But I need to use websipcaller to call to voip provider. I tried to
register on my voip provider, and use just outbound websocket
proxy, but
it not working giving 401
the config I am using is
https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg
Please help adopt it to relay anything to external sip provider. I
don't
know what I need, relay, sip trunk or path module.
kamailio configs are very hard for me to understand.
So I need it just as websocket proxy to my voip provider, that doesn't
support websocket.
thanks!
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Best regards,
Ihor (Igor)
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