When someone call us on our external number, and colleague1 picks up the phone is all going perfect. But when colleague1 doesn’t know the correct answer and puts the customer on hold the customer gets music as planned. Now colleague1 calls internally with colleague2. When colleague1 hangs up on colleague2 and picks up the customer again, he can hear the customer but the customer cannot hear colleague1.
colleague1 is talking with colleague2, now colleague3 calls colleague1 he picks him up and put the phone in conference so colleague1, colleague2 and colleague3 can talk and speak to each other. But now the sound dropped for colleague2 or colleague3 (thats random)
The sound dropout is not always its like 50/60% of all conferences, so its hard to troubleshoot.
We have 1 VoIP server in our datacenter. Everyone has a SNOM360 or SNOM370 VoIP Phone on his desk. We are all working from our homes, so we do not have a office with people in it. We are all using different Internet providers.
We have a voip connection from Matrix and MyDivert.
What did i try so far:
I have changed some settings with NAT to on/off
Switched all codecs to: alaw
Our incoming connection from Matrix and MyDivert are also alaw
used a stunserver on the phones at home.
Before we used FreePBX we had a old Asterisk server without a web gui. When we used that everything was working smoothly, beside when u wanted to change voicemail or menu sound it takes u like 10 times more time then with FreePBX.
So hopefully someone can help me/push me in the correct way, like i said the sound drop out (one way) is not always but like 6 out of 10 phone calls tested on the same person.