I have FreePBX Version 12.0.67, Asterisk Version 11.17.1 & a Cisco SPA8000 ATA with Firmware 6.1.10. Does anyone know of a guide that will walk me through how to set the ATA to work with the PBX server? I have 7 analog lines that will be connected to the ATA that are in a ring group from the phone company. I have tried to set it up as a SIP Trunk but not quite sure what I am doing wrong. In the Asterisk Registries it shows the trunk but says it was rejected. I have 2 other SIP trunks from voip.ms and they are registering fine. Thank you in advance for any help anyone can give.
Capture the logs of calls by CLI and post here
jturner,
I setup 2 SPA8800s before, the configuration was very strange.
Basically each phone line on the SPA8800 has a different port and is a separate SIP Trunk in FreePBX.
I could explain further and walk you through it, but if you are trying to use them as gateways to PSTN lines, forget it. Sell them on ebay like i did.
After all the pain of configuring them, they had a persistent echo of your own voice when on calls. I have fixed echo problems before, but this echo was unfixable. Even Cisco TAC was clueless, even though they support the integration of SPA8800s with third party PBXs, it seemed like no one had even heard of a SPA8800 at TAC.
So sold on ebay, got a SIP Trunk for the client, and made a mental note to never attempt SPA8800 configuration again
I sent this one back and got a Grandstream GWX4108. I needed an FXO gateway, not an FXS gateway.
Cool, did the Grandstream work? No echo?
No echo that I have noticed so far, but its not in the production stage yet so we will see.