I have searched around but can not seem to find an answer to my issue, so I am posting it as a new topic.
I just recently downloaded and installed the FreePBX Distro, as we are leaving our hosted provider and bringing everything in-house. The install went smooth, and all our phones are working, thanks to purchasing the commercial Endpoint Manager from Schmoozecom.com, but there is one nagging issue…
We have a second asterisk device, a predictive dialer, and I am having a hell of a time linking the two together. Currently we are using a SIP trunk on the FreePBX with an outbound route and dial patterns to connect those that need access. This works fine, however after getting about 20 people connected, QoS nosedives to the point that we only hear every other word the person on the other line is saying.
Each device has 2 NICs, with eth0 on both going to my local network, and Eth1 going to a private Voice circuit, completely segmented from the internal network.
Using this setup, a regular telephone call looks like this:
Phone -> Cisco WS-C3560X-24T-S -> Core Switch -> PBX Eth0 -> PBX Eth1 -> Cisco Voice Switch -> Provider
And a Dialer Call looks like this:
Phone -> Switches -> Core Switch -> PBX Eth0 -> Core Switch -> Dialer Eth0 -> Cisco Voice Switch -> Provider
As you can see, there is an extra hop back to the Core switch on dialer calls. Multiple that by the 75 agents, and 30 admin people making calls and using data…
Unfortunately, at this time, we are not using VLans, but that is being fixed next week. But the hardware is all higher end HP/Cisco, so it isn’t like I am using netgear or linksys.
If there is no “fix” then I would have to provision the phones to have a direct connection to the dialer, as well as the PBX, but from what I can see in Endpoint Manager, that isn’t doable.
Any insight you guys can give, would be great!