When you meet the similar issue. You could try to get the Asterisk Log by following the guide
U-Series & Gateways: How to Capture Asterisk CLI logs for Yeastar Devices
S-Series: How to Capture Asterisk CLI Logs for Yeasatr S-Series VoIP PBX
Then check if you have the similar log like the following:
-- Accepting call from '21345875' to 's' on channel 0/31, span 2
[2014-10-16 19:15:50] WARNING: chan_dahdi.c:2992 dahdi_enable_ec: Enabled echo cancellation on channel 31
-- Executing [s@DID_trunk_E1Trunk1:1] Macro("DAHDI/31-1", "Routein_E1_to_SIP,1,s") in new stack
-- Executing [s@macro-Routein_E1_to_SIP:1] Set("DAHDI/31-1", "CDR(userfield)=Inbound") in new stack
-- Executing [s@macro-Routein_E1_to_SIP:2] GotoIf("DAHDI/31-1", "0?Blacklist-Handle,s,1") in new stack
-- Executing [s@macro-Routein_E1_to_SIP:3] Set("DAHDI/31-1", "TRUNKDID=") in new stack
-- Executing [s@macro-Routein_E1_to_SIP:4] Goto("DAHDI/31-1", "Routeout_E1_to_SIP,s,1") in new stack
-- Goto (Routeout_E1_to_SIP,s,1)
== Channel 'DAHDI/31-1' jumping out of macro 'Routein_E1_to_SIP'
[2014-10-16 19:15:50] WARNING: pbx.c:4364 __ast_pbx_run: Channel 'DAHDI/31-1' sent into invalid extension 's' in context 'Routeout_E1_to_SIP', but no invalid handler
-- Executing [h@Routeout_E1_to_SIP:1] Hangup("DAHDI/31-1", "") in new stack
== Spawn extension (Routeout_E1_to_SIP, h, 1) exited non-zero on 'DAHDI/31-1'
Enable the Overlap option or switch it to Incoming on BRI/PRI trunk settings.
If the issue log disappears, but you receive SIP errors you need to check if your caller ID from BRI /PRI line is blank or not. Failed to Make Calls from TE/TB to SIP trunk When Caller ID is Blank