I did the configurations, but I get a busy/congested error
[Mar 7 16:47:46] -- Executing [011216210040528@default:2] Dial("SIP/8001-00000092", "SIP/INTLDialer/11216210040528") in new stack
[Mar 7 16:47:46] == Using SIP RTP CoS mark 5
[Mar 7 16:47:46] WARNING[23244]: chan_sip.c:6033 sip_call: No audio format found to offer. Cancelling call to 11216210040528
[Mar 7 16:47:46] -- Couldn't call SIP/INTLDialer/11216210040528
[Mar 7 16:47:46] == Everyone is busy/congested at this time (0:0/0/0)
[Mar 7 16:47:46] -- Executing [011216210040528@default:3] Hangup("SIP/8001-00000092", "") in new stack
[Mar 7 16:47:46] == Spawn extension (default, 011216210040528, 3) exited non-zero on 'SIP/8001-00000092'
[Mar 7 16:47:46] -- Executing [h@default:1] AGI("SIP/8001-00000092", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CHANUNAVAIL----------") in new stack
[Mar 7 16:47:46] -- <SIP/8001-00000092>AGI Script
agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0