@smirciat wrote:
So go ahead and add me to the long list of people struggling with inbound SIP calls dropping after precisely 32 seconds each time. I have read so many different posts on this issue, and I do realize it is a NAT or firewall issue. My FreePBX 13 setup is behind a router with recommended ports forwarded to the PBX. I set up Settings > Asterisk SIP Settings as recommended with NAT - yes and IP Configuration - Dynamic IP, although I've tried every combination of settings I can think of. Turning on SIP trace shows the ACK packet being retransmitted until it times out. I have read the wiki link for "SIP+Retransmissions" which is definitely my issue, but I jat can't figure out what other settings to try.
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