Provide configuration for FreeSWITCH

I need a FreeSWITCH (FS) configuration for the following functionality. It should connect: * 2 Linphone endpoints, extensions 100 and 101 * 1 Linphone endpoint + 1 Yealink T41S phone on the same extension 102 (both should ring simultaneously on incoming call) * 1 external SIP trunk provider for incoming and outgoing calls For outgoing calls, all numbers dialed on internal endpoints that match full international number format (e.g. +447712345678 or 00447712345678) should go via the external SIP trunk, all numbers that are longer than 3 digits (i.e. that are not internal extensions) but don't match the international format should be rejected as invalid with a voice message (e.g. "the number you dialed is incorrect"). The FS endpoint at extension 101 should NOT be able to call outside (only internal calls to other extensions). For incoming calls from the external SIP trunk, there are 2 external (PSTN) numbers (e.g. +1 905 878 5000 and +1 905 878 5001) - when receiving a call for the first number, it should be redirected to the 100 extension, for the second number to the 102 extension. * 1 internal PBX (Asterisk) that has 2 more endpoints with extensions 200 and 201. They should be accessible to the FS (FreeSWITCH) endpoints and the FS extensions (100-102) should be accessible to the Asterisk endpoints. Endpoints connected to Asterisk should be able to make outside calls too via the external SIP trunk connected to FS. Additional details: Linphone and Yealink endpoints connected to FS should be able to see the presence status (busy, ready, etc.) of other FS endpoints. There is NO need for NAT traversal, the FS itself, its endpoints and the internal PBX (Asterisk) are on the same private network (say [login to view URL]); the external SIP trunk has a public static IP (say [login to view URL]), the internal network also has a public static IP (say [login to view URL]). The required ports could be opened as needed. The FS configs should be in XML format. The communication should always go via FS (no direct RTP). The endpoints configured with FS should be communicating voice with Opus/G.722/PCMA codecs and video with VP8/H.264. When communicating with the external SIP trunk provider and the internal PBX (Asterisk), the communication should be voice only with G.722 and PCMA. As part of this work, FS should be built from sources using the latest release (1.6.18 at this moment) and all components not required for the functionality mentioned above should be removed during compilation (with ./configure options). The FS configs should also disable everything not absolutely needed (modules, etc.). I need a bare minimum setup. Everything should be provided as instructions and configuration files (e.g. FS ./configure options and configs, instructions for Linphone and Yealink basic settings, ports to open, etc.). It is NOT needed to deploy the configuration on my server, I'll do it myself with your instructions.