- Bronze, 100 points or more
I'm working on a system for a client that is similar to most of the configurations that we do including a load balancing SA520 in front of a UC500. I have it configured to pass SIP and VPN traffic through the SA500 to the UC500, but I ran across a really strange thing the other day. I generally test inbound and outbound calls by calling my cell phone (Verizon). I happened to call a normal terrestrial number and didn't get any audio on subsequent calls. The person on the other end called me back and audio worked fine. At this point, it appeared as though I had a problem in my configuration, so I started trying all sorts of different phone numbers to check the UC500/SA500 configuration. What I came up with, is this:
Equipment configuration: SA520 - UC540 - ESW520FE - ESW540FEwithPoE
Handsets: SPA502G and 7925 wifi phones
SIP Trunk provider: Nexvortex
Codec preference 1: G729r8
Codec prefernece 2: G711ulaw
1. All calls inbound and outbound work to Verizon and ATT cell phones (didn't try Sprint or other providers)
2. Calls will USUALLY work from this customer's site to our corporate office (which has a UC520 and is making use of PSTN lines from local telco Centurylink)
3. Occaisionally audio will drop when calling our corporate office
4. Calls will ALWAYS drop when I call my wife's office
5. Calls will ALWAYS drop when I call the customer's office in South Carolina's State IT department
The State IT department Office has an Avaya IP PBX
Our corporate office is employing a Cisco UC520 IP PBX
Not sure what my wife's office is using
To troubleshoot the problem, I removed the SA520, changed the WAN IP address of the UC540, DNS info, added NAT and Firewall, and ALL calls work perfectly. It appears as though the problem somehow exists within the SA520 configuration, but I'm having trouble figuring out exactly where the problem lies.
I do a little CLI configuration to change codec preference, configure the transcoder for G729 for AA and VM, to reuse 5060 for SIP invites, and to change the rate at which SIP registration takes place. Otherwise, everything is built with CCA 2.2.1.
I have the SBCS working on this, but so far, no luck. Can anybody tell what I'm missing here? It appears to be SA520 related, but I am unsure about where else to go from here. I would greatly appreciate any suggestions that anybody might have.
P.S. Here are some logs from a connected call with failed audio.