Running Asterisk 18.104.22.168 with FreePBX 2.10.0 on CentOS 6.2 32-bit. No real issues to report. Trunks work fine, and SIP extensions all work fine with softphones.
I’m experimenting with a Cisco ATA 186. I have it configured as a SIP device for two extensions. The ATA is sitting behind a NAT router. The extensions register just fine and I can make outgoing calls from either extension with no problem. The problem is with receiving incoming calls.
I couldn’t receive incoming calls at all until I enabled anonymous Internet requests in the security section of the NAT router, but after doing that calls started coming in. Unfortunately incoming calls aren’t reliable, in that Asterisk gives a “caller unavailable” message a lot of the time. Sometimes only phone 1 is available, and sometimes it’s only phone 2. Sometimes both are available, and sometimes neither is available. The behavior seems to be related to how the extensions are indicated in the FOP Panel, where they sometimes are half ghosted (not sure what being half ghosted means). However, even when they are half ghosted I can still make outgoing calls through asterisk.
I tried port forwarding 5060-5061 UDP & 10000-20000 UDP to the ATA, with no change. I even tried placing the ATA in the router’s DMZ, still with no better results. I had upgraded the ATA’s firmware to 3.2.0 SIP firmware before starting this project, so I thought maybe the problem was that the firmware was unstable. I downgraded the firmware to 3.1.0 but got no better results. Here is a link to my current settings (proxy IP addresses deliberately munged for image).
I’ve tried configuring it for only one extension, but even a single extension drops in and out on me for incoming calls. Still, it’s 100% solid for outgoing calls.
Clearly, this ATA device isn’t suitable for a production environment of any kind the way it’s performing now. Any advice would be greatly appreciated.