These are test calls from Sipp clients toward Sipp servers, the RTP does not goes through FreeSWITCH with bypass_media=true. The SIP flow is quite simple: INVITE,180,200,ACK,BYE,200. There is no DTMF, re-INVITE, OPTIONS. The calls duration is set to have 1k to 10k simultaneous calls and 5 to 15 calls per second. About the ESL program or the dialplan: Some headers are set (caller number/name, Privacy). A maximum duration is set with sched_hangup. A maximum ringing duration is set with call_timeout. Some special variables are configured (hangup_after_bridge=false, continue_on_fail=true, sip_ignore_remote_cause=true, bypass_media=true). Then a bridge is used to forward the calls to several gateways (round-robin in the ESL program or with mod_distributor in the dialplan). Both inbound and outbound use Sofia. On 29/09/2021 22:42, Brian West wrote:
What is the call doing?