3 Edgerouters (main site, remote site, remote site) with working IPSEC VPNs for 2 years+
PIAF inside main site with local IP, static IP and VPN subnets added to SIP settings in FreePBX
All of a sudden 3 weeks ago I have working audio in main office (where the PIAF server is) but no audio whatsoever in remote offices. Traffic passes fine between sites (can pull up Yealink phone web pages in each outer office, or PBX admin page from outer offices).
But I see this in my Asterisk log when calling a remote extension:
Obviously Flowroute is our SIP provider.
TCPdump shows no effort to send RTP from the PBX so I can't even see where it's trying to send to. I see INVITE -> OPTIONS -> ACK -> BYE on the SIP signaling between server and remote phone (which is the internal IP of the remote phone on a separate VPN'd subnet)
Two users sitting 10 feet away in the same remote office can ring each other's phones, but no voice transmits.
I've changed PBX servers, I've changed just about every setting except the direct media stuff, changed all sorts of firmware versions on firewalls, all sorts of firewall settings, natted remote site, unnatted remote site, set firewalls to default configs and rebuilt from scratch, built PIAF server from scratch and placed it in office but remote phones still don't transmit audio.
I've never had to dig this far down on a no-traffic issue, if anybody has had the dreaded "lack of RTP activity in 31 seconds" message and resolved it, please post it here.
Thanks in advance to anyone who dares respond.
PIAF inside main site with local IP, static IP and VPN subnets added to SIP settings in FreePBX
All of a sudden 3 weeks ago I have working audio in main office (where the PIAF server is) but no audio whatsoever in remote offices. Traffic passes fine between sites (can pull up Yealink phone web pages in each outer office, or PBX admin page from outer offices).
But I see this in my Asterisk log when calling a remote extension:
Obviously Flowroute is our SIP provider.
TCPdump shows no effort to send RTP from the PBX so I can't even see where it's trying to send to. I see INVITE -> OPTIONS -> ACK -> BYE on the SIP signaling between server and remote phone (which is the internal IP of the remote phone on a separate VPN'd subnet)
Two users sitting 10 feet away in the same remote office can ring each other's phones, but no voice transmits.
I've changed PBX servers, I've changed just about every setting except the direct media stuff, changed all sorts of firmware versions on firewalls, all sorts of firewall settings, natted remote site, unnatted remote site, set firewalls to default configs and rebuilt from scratch, built PIAF server from scratch and placed it in office but remote phones still don't transmit audio.
I've never had to dig this far down on a no-traffic issue, if anybody has had the dreaded "lack of RTP activity in 31 seconds" message and resolved it, please post it here.
Thanks in advance to anyone who dares respond.