What parameters to change to avoid the BUSY signal?

Question

Asterisk

SIP phone, when the peer status starts to exceed >50ms, it will not get through, the answer is a busy signal.
Tell me what parameters on asterisk or on the Grandstream telephone set need to be changed to avoid this?

Everyone is busy/congested at this time (1:1/0/0)
    -- Executing [[email protected]:38] ExecIf("IAX2/500-to-200-9501", "0?Set(DIALSTATUS=)") in new stack
    -- Executing [[email protected]:39] GosubIf("IAX2/500-to-200-9501", "0?s-BUSY,1") in new stack
    -- Executing [[email protected]:40] MacroExit("IAX2/500-to-200-9501", "") in new stack
    -- Executing [[email protected]:10] GotoIf("IAX2/500-to-200-9501", "0?exit") in new stack
    -- Executing [[email protected]:11] Set("IAX2/500-to-200-9501", "SV_DIALSTATUS=BUSY") in new stack
    -- Executing [[email protected]:12] GosubIf("IAX2/500-to-200-9501", "0?docfu,1") in new stack
    -- Executing [[email protected]:13] GosubIf("IAX2/500-to-200-9501", "0?docfb,1") in new stack
    -- Executing [[email protected]:14] Set("IAX2/500-to-200-9501", "DIALSTATUS=BUSY") in new stack
    -- Executing [[email protected]:15] NoOp("IAX2/500-to-200-9501", "Voicemail is 'novm'") in new stack
    -- Executing [[email protected]:16] GotoIf("IAX2/500-to-200-9501", "1?s-BUSY,1") in new stack
    -- Goto (macro-exten-vm,s-BUSY,1)
    -- Executing [[email protected]:1] NoOp("IAX2/500-to-200-9501", "Extension is reporting BUSY and not passing to Voicemail") in new stack
    -- Executing [[email protected]:2] GotoIf("IAX2/500-to-200-9501", "0?exit,1") in new stack
    -- Executing [[email protected]:3] PlayTones("IAX2/500-to-200-9501", "busy") in new stack
    -- Executing [[email protected]:4] Busy("IAX2/500-to-200-9501", "20") in new stack
  == Spawn extension (macro-exten-vm, s-BUSY, 4) exited non-zero on 'IAX2/500-to-200-9501' in macro 'exten-vm'
  == Spawn extension (from-trunk, 506, 1) exited non-zero on 'IAX2/500-to-200-9501'

Answer 1

qualify=2000 ?