Here’s a head scratcher… I have hosted FreePBX through freepbxhosting.com. Long story short, I lost connectivity to the interface, and the phone will not register. I found out out I can connect successfully from another ISP on my end. Trace Route shows a router (a Time Warner router in Chicago) with %50 packet loss, continually, for the last few days. Connecting through my other ISP (4g phone), no issues, using a completely different route. I found that getting a free SIP proxy account (sipoutbound.com) will let me register successfully when I set it up in my phone. The issue I have now is (I’m guessing) the lack of RTP packets. Both inbound and outbound caller have dead air only. No audio. My phone is behind a NAT firewall. Any ideas on how to redirect RTP? I would hope that the buggy router would be addressed soon but starting to wonder…
Many Cable companies don’t like SIP and it’s ensuing RTP traffic so messed with it. They where told to stop doing that, some still don’t follow that edict. Good Luck
We provide hosting that is ideal for your circumstances.
1 - We include a free VPN router to connect back to our hosting facility
2 - We are “on net” with Time Warner and have a direct fiber optic connection to their “t-bone” backbone network. In fact one of our data centers is colocated with TWC’s largest NAP/POP in the midwest
3 - We host both FreePBX and PBextended, the commercial Asterisk PBX from Schmooze.
One thing to keep in mind is dropping 50% of ping packets is somewhat meaningless. The first thing a router does when it gets bust is ignore ICMP messages (ping is part of ICMP). To really test you need a two part test tool that actually measures loss and jitter of UDP RTP packets, just like the ones you will use.
If I can be of help please send me a PM with your contact info.