I am calling into a mitel conference call. When I dial, it doesn’t take the dial tones. Usually these means the Sip provider is sending the wrong key strokes. For example 11111 when I only pushed 1. How do you trouble shoot this from cli?
Add DTMF to your logfile/console output, BUT, make sure your dtmfmode is copasetic with the trunk
how do you add DTMF to the log output?
Settings->logfiles
I turned on DTMF. When you dial the number 0123456789, then it says enter the conference code, it is not accepted. I don’t see anything about DTMF in the log below.
my output is:
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx.c: Executing [0123456789 @from-internal:1] Macro(“SIP/101-00004177”, “user-callerid,LIMIT”) in new stack
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx.c: Executing [0123456789 @from-internal:2] Set(“SIP/101-00004177”, “ROUTEUSER=101”) in new stack
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx.c: Executing [0123456789 @from-internal:3] Set(“SIP/101-00004177”, “ROUTEUSER=101”) in new stack
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx.c: Executing [0123456789 @from-internal:4] GotoIf(“SIP/101-00004177”, “1?notblind”) in new stack
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx_builtins.c: Goto (from-internal,0123456789 ,7)
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx.c: Executing [0123456789 @from-internal:7] GotoIf(“SIP/101-00004177”, “1?restrictedroute-426ce6f8feb7acb896953e90e72870a3,0123456789 ,2:outbound-allroutes,0123456789 ,2”) in new stack
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx_builtins.c: Goto (restrictedroute-426ce6f8feb7acb896953e90e72870a3,0123456789 ,2)
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx.c: Executing [0123456789 @restrictedroute-426ce6f8feb7acb896953e90e72870a3:2] Gosub(“SIP/101-00004177”, “sub-record-check,s,1(out,0123456789 ,no)”) in new stack
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx.c: Executing [out@sub-record-check:1] NoOp(“SIP/101-00004177”, "Outbound Recording Check from 101 to 0123456789 ") in new stack
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx.c: Executing [out@sub-record-check:7] Gosub(“SIP/101-00004177”, “recordcheck,1(no,out,0123456789 )”) in new stack
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx.c: Executing [0123456789 @restrictedroute-426ce6f8feb7acb896953e90e72870a3:3] Set(“SIP/101-00004177”, “FAXOPT(gateway)=yes”) in new stack
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx.c: Executing [0123456789 @restrictedroute-426ce6f8feb7acb896953e90e72870a3:4] ExecIf(“SIP/101-00004177”, “0 ?Set(CDR(accountcode)=)”) in new stack
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx.c: Executing [0123456789 @restrictedroute-426ce6f8feb7acb896953e90e72870a3:5] Set(“SIP/101-00004177”, “MOHCLASS=onholdmusic”) in new stack
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx.c: Executing [0123456789 @restrictedroute-426ce6f8feb7acb896953e90e72870a3:6] Set(“SIP/101-00004177”, “_NODEST=”) in new stack
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx.c: Executing [0123456789 @restrictedroute-426ce6f8feb7acb896953e90e72870a3:7] Macro(“SIP/101-00004177”, “dialout-trunk,2,10123456789,off”) in new stack
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx.c: Executing [s@macro-dialout-trunk:6] Set(“SIP/101-00004177”, “DIAL_NUMBER=10123456789”) in new stack
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx.c: Executing [s@sub-flp-2:1] ExecIf(“SIP/101-00004177”, “0?Set(TARGET_FLP_2=110123456789)”) in new stack
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx.c: Executing [s@macro-dialout-trunk:15] Set(“SIP/101-00004177”, “OUTNUM=10123456789”) in new stack
[2018-12-05 15:27:36] VERBOSE[14102][C-00004125] pbx.c: Executing [s@macro-dialout-trunk:22] Set(“SIP/101-00004177”, “__CRM_DESTINATION=10123456789”) in new stack
[2018-12-05 15:27:37] VERBOSE[14102][C-00004125] pbx.c: Executing [s@macro-dialout-trunk:28] ExecIf(“SIP/101-00004177”, “1?Set(CONNECTEDLINE(num,i)=10123456789)”) in new stack
[2018-12-05 15:27:37] VERBOSE[14102][C-00004125] pbx.c: Executing [s@macro-dialout-trunk:32] Dial(“SIP/101-00004177”, “SIP/SipLogicTrunk/10123456789,300,M(setmusic^onholdmusic)Tb(func-apply-sipheaders^s^1)”) in new stack
[2018-12-05 15:27:37] VERBOSE[14102][C-00004125] app_stack.c: Spawn extension (from-trunk, 0123456789 , 1) exited non-zero on ‘SIP/SipLogicTrunk-00004178’
[2018-12-05 15:27:37] VERBOSE[14102][C-00004125] app_dial.c: Called SIP/SipLogicTrunk/10123456789
Sorry can’t help with restricted routes it is closed source
So first, there’s nothing really special about “restricted routes” which is basically Extension Routing or Class of Service. When it comes to the Outbound Routes part all it does is create a bunch of custom outbound route contexts that just have those long names. That would mean that there is a context in the dialplan called and in there will be lines like
[restrictedroute-426ce6f8feb7acb896953e90e72870a3]
include => outrt-1
include => outrt-10
include => outrt-3
To be clear this means that it has no bearing on your actual problem, it is just how routes are ordered for that “restricted route”.
Second, your call trace ends at [2018-12-05 15:27:37] VERBOSE[14102][C-00004125] app_dial.c: Called SIP/SipLogicTrunk/10123456789 so we’re not seeing the call being even answered or what happens after it was answered so are you sure you looked further into the log or did you copy and paste everything you thought was for the call?
So if you have it, please provide a full trace of the call with a full sip debug so we can see the INVITEs and the packets being sent and what is really in there. However, keep in mind this is a bridged call so unless there is some In-Call feature code that Asterisk is expecting to accept from the parties in a bridged call, the DTMF is most likely going to be passed through and thus not seen in the logs. Generally the DTMF logging is for when a use calls into the PBX and the PBX is what answers the call (IVR, etc) and then it logs the DTMF it is accepting.
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.