On this video we cover the setup for a SIP Trunk between 2 Asterisk Servers. The sip.conf and dialplan configuration. We use Ekiga to test calls between both servers.
Quick question. Is it possible to forward a google number directly to an asterisk server. So when the call is dialed it's routed to our servers which in turn can be redirected to other numbers or extensions. Do you know if that is possible? great videos btw.
not sure if anyone cares but if you're stoned like me during the covid times you can watch pretty much all the latest series on instaflixxer. Been streaming with my girlfriend recently xD
man, i create an asterisk server on ubuntu 18.04,but all the cals from outside of my network doesnt works,basically they works but i dont have any audio from them why this happens ,from my lan all works ok
thanks for this video. It was very difficult for me to understand this.
Thanks for leaving in the configuration errors. It kinda also showed some basic asterisk debugging ;-)
От души, благодарю!!!
Thanks this is a great vdeo, it really help me in my project.
thanks for you video. I could connect two Asterisk servers, was a great help for me
Thank you, That's what I am looking for.
Quick question. Is it possible to forward a google number directly to an asterisk server. So when the call is dialed it's routed to our servers which in turn can be redirected to other numbers or extensions. Do you know if that is possible? great videos btw.
Excellent effort. Thanks for the video.
Thank you so much! I don't know that I need the feature "insecure=invite", thanks you a lot again, from Lima-Peru.
Thanks for your video. I am expecting more video's from you.
also you have to register your trunk on each server
how can I set it up on the computers which are on the different networks?
Thanks for the video, hope to see more from you :)
not sure if anyone cares but if you're stoned like me during the covid times you can watch pretty much all the latest series on instaflixxer. Been streaming with my girlfriend recently xD
@Zion Mayson definitely, have been using Instaflixxer for since november myself =)
Plz upload full video of sip trunk configuration from basic
Thank You! Подобный обзор я сделал на своем канале, только немного простым языком ;-)
how could I create sip trunk?
why dont you just use reload command instead using sip reload and dialplan reload, if u will just use reload command it will reload all asterisk conf.
Hi Rahul. reload command reload everything which takes resources and time. In this scenario we need only sip and dialplan to be reloaded.
The video is nice I rate it 3/10 CZ of errors seems like you didn't do your homework before shooting video secondly in my case it say number not found
video dosent really start before 5:25 -_-
man, i create an asterisk server on ubuntu 18.04,but all the cals from outside of my network doesnt works,basically they works but i dont have any audio from them why this happens ,from my lan all works ok
Waste of time