Analysis and Fix of Issues with Asterisk infrastructure

All customers are complying about calls dropped or not being completed. The user perception is that the call does not reach the endpoint. Looking through the Asterisk (FreePBX distro) servers, investigating logs, we could not find anything that could explain what is happening. For a better troubleshooting, I will try explain some specific scenarios (please reference the drawing below for a clear understanding): a) Customer F dials from its Siemens PBX and there is a SIP trunk registered on our Asterisk Server. We provide only GSM calls, so the server will check the number dialed by user (on Customer F Siemens PBX) on a portability database (PHP AGI script that runs) and decide which Outbound Route will use to complete the GSM call. This will point to which mobile operator (GSM SIM card) will be used. The server will send the call to GSM gateways. Each GSM gateway is configured as a SIP trunk with the Continue if Busy option on FreePBX checked. Looking on CDR Report of gateways, there are messages that the call was cancelled. We discussed with GSM gateway support team about this issue and they told us that when gateway has the CANCELED status, this means that SIP side has canceled the call. We tried to identify any relationship on that without success. b) The Customer F has a call center that operates with a predictive dialer. There are 45 positions on the call center and the predictive dialer sends about 60 calls simultaneously at campaign time. The predictive dialer complying about Fast Busy. We could not find anything related with this status. Doing some research we find out that Fast Busy is the same as network congestion. Again, we checked all logs and could not find anything that could point us to any issue with network.