Audio trouble for months now, only outgoing calls, please help

Hi there :slight_smile:

first about my setup, running centos 6.2, asterisk, freepbx
I have a modem/router with NAT enabled. Asterisk and my extension from which I make the call are on the same local network, behind the modem/router.
I have forwarded ports 10000-20000 to asterisk and configured asterisk accordingly.
My external IP is Asterisk and freepbx are on, my extension I make the calls from is ‘903’ and is on
I have configured my extension and connected it. Also I have setup sip trunks and configured outbound rules etc.
This all works fine. When I receive calls, all works great, I have two way audio without any trouble during the whole call.
When I make an outbound call, the incoming audio works without flaws, all the time, however my outgoing audio drops after a minute or so, sometimes after 5 minutes. So first the other person can hear me then not. On some calls the outgoing audio starts working again after a bit, but then drops again.
In freepbx in the ‘asterisk sip settings’ I’m not sure how to set the NAT settings properly. Currently:
NAT: no (but have also tried ‘yes’, ‘never’, and ‘route’ whilst keeping same settings below with same audio problems results as described above)
IP Configuration: Static IP
External IP:
Local Networks:
Would this be correct?

I have run ‘sip set debug on’ and ‘rtp set debug on’ to see what happens during the call, the output is here:
I had some help with interpreting the logs on asterisk forum ( basically as I was told, there are no re-invites, so I changed in freepbx, ‘asterisk sip settings’ the ‘re-invite behaviour’ setting to ‘yes’, no effect though. The calls are made to mobile phones and/or pstn landlines that are not connected to asterisks. They are not the problem (suggested in post). I got no clue any more what I can do.
Would appreciate any help. Thank you in advance!

Does your router have any sort of SIP Helper or SIP ALG support? If so switch it off as this can be a source of one way audio problems.

nat=yes should be the correct setting.

Not sure if Router has SIP Helper or SIP ALG support. I do have SIP functionality in the router/modem. However there are no options for me to change any of the sip settings in my router/modem or to turn them of. It’s a Arcadyan VGV7519 not much to find on the net about it. Should it be the modem/router that does stuff to my call, should that not be seen in the logs (from my link above)? Still experiencing same problem.

If the router has SIP functionality but you can’t change it then that could be interfering with the clean flow of SIP traffic. Many routers with SIP ALG functions tend to cause more problems than they help with. I’ve never come across that specific router so don’t really know what to suggest other than to try another router.