Hi,
I have some virtual phone numbers with United World Telecom that redirects thought SIP to our PBX.
Some days ago we lost connectivity from outside sip accounts. Internal calls works fine.
After 2 days trying to figure about the problem, today i installed a fresh copy of last FreePBX distro.
But I experiences the same problem.
My phone number at United World Telecom is only for receiving calls, not outbound calls, so i did not config a trunk, i just configured as an extension and i added an inbound route. This was working fine until last week.
I have set:
NAT = Yes at sip config.
NAT = Yes at inbound extension.
I have the public and private IPs set.
Allow SIP Guests ON
Allow Anonymous Inbound SIP Calls? YES
Cleaned fail2ban
I used TCPDUMP at PBX and i processed with WireShark.
I think the problem would be related to firewall, but i am not sure. I got this messages:
186 59.649982 HIDE REAL IP 10.142.138.3 IPv4 1514 Fragmented IP protocol (proto=UDP 0x11, off=0, ID=129c)
Packet with INVITE information,
but the PBX responses ICMP error:
296 89.149487 10.142.138.3 HIDE REAL IP ICMP 590 Time-to-live exceeded (Fragment reassembly time exceeded)
I tried googling for ‘Time-to-live exceeded’ but i only got ICMP protocol description but no entries related to Asterisk or FreePBX