Hello @all,
after two years of no problems with freepbx @100/100 function, we now have some problems. The hardware we are using, configs of our systems was not changed, none of them. Neither on debian virt-Srv, nor on hp switches, on vlans, on freepbx or on pfsense, NO changes @all (cause of: never change a running system).
The major, the biggest change was this one:
Freepbx-Version: FW Console - FreePBX Utility 14.0.13.23
asterisk-Version: Asterisk 13.22.0
=>
Freepbx-Version: FW Console - FreePBX Utility 14.0.16.11
asterisk-Version: Asterisk 13.22.0
Therefore (cause of 14.0.16.11) i used a old backup of this system, at that time there was no problems, seven monts ago ( i do have older backups too); i could use it because there was no config changes in freepbx in the past time between backup and now. But did not help.
I pushed this old vm-freepbx backup to an another srv, connected with another cabel, another hardware/NIC etc, did not help too. 
Right now, these problems torturing us:
-
the incoming call is clear/loud. But the outgoing sound is really, really quite/silent.
-
Sometimes the person called us could not hear us, we will be not heard on the other side, also vice versa.
If we call back seconds later this CLIP, no problems @all, never.
No problems occured on outgoing calls, never ever. -
a lot af calls shows some hums, like in EU 50Hz and also cheep sounds. Some of our clients says the sound is hacked up, is stuttering somehow. We changed the cabel at the phone, all of them (3x Yealinks 48/2s).
But none of this problem will occur if i get connected using the same way/route (internet - cable-provider - cable modem/fixed IP - pfsense - hp J9022A Switches - debian srv/an own NIC for freepbx) through openvpn to freepbx.
If i call this system with android/Zoiper/Linphone the sound is clear, loud, no disturbance, no trouble. We can chat about two hours or more without ANY of these problems. It sounds if the person who i am speaking is right next to me, without any loss on quality of speech all that long time. Works as designed. 
None of these problems was before for about two years; pure 100/100 function, also none of these problems occurs if calling internal-2-internal (s. above, int-openvpn-2-int or int-2-int) or calling int-2-outside, land-line or mobile cell phones. Only incoming @calls by our voip-provider (through IVR) will, are sometimes (not all of them) f… up.
The system uses on two trunks a/m-law, not g772 codec. Because the provider saw a jitter, told us to use only a/m-law because of the higher payload of g722 packets. Although, s. above, about two years not problem with g722-codec.
I think these all describes one problem (which/where?). Therefore i put them together in this posting.
Propably it sounds like some NATting problems, but we do have sound mostly on both sides (in- and out), but sometimes - as described before - we will be not heard, or we could not hear the other person. And, although no change of any config, it was fine about two years. :-[
Something went wrong, sometime.
What for should i look after in logfiles, if the person calling us will not hear us, vice versa?
Somehow i should see it, that something is broken and the system will complains it(?).
What is the difference between a regular call and a faulty call in logfiles like /var/…/asterisk/full?
This situation leaves me lost at where else to look for.
I read in this forum a lot of topics like no audio incoming/outgoing etc. But none of them describes, what we have. Any help is highly appreciated.
Thanks in advance.
ELindemann.