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Queue Help

May 6, 2018, 12:40 pm
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@dillydilly90 wrote:

I have a queue with five static agents but I only want one agent dealing with a queue call at any given time. I thought I had a solution but it turns out it’s not working like originally thought. Basically the call comes into the queue and there is a custom extension that acts as an agent. The custom extension has a dial string that calls a ring group with all the ‘static’ agents. However when an agent answers the phone the next call comes in from the queue and starts ringing all the other agents. I don’t want this to happen. I want the current agent to finish with the current call before the next one comes in.

Sadly, we simply don’t have enough man power to have three agents tied up on a phone at the same time.

I was looking at this: [SOLVED] Set Ring Group Busy When One Line Occupied
However I can’t seem to get it to work, all it does is make the second call ring busy.

Posts: 7

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UCP Phone Can't Place or Recive calls

May 6, 2018, 12:44 pm
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@PitzKey wrote:

FreePBX 14. Asterisk 13. UCP 14.0.2.5. WebRTC 14.0.3.7

PBX has a LE cert, and i see 99102 registered.

However, when i call that extension it rings but upon trying to answer nothing happens.
I also cannot place calls.

Here’s a call log:

 Executing [89@ivr-1:1] Goto("SIP/from-SipProvider-00000002", "from-did-direct,102,1") in new stack
    -- Goto (from-did-direct,102,1)
    -- Executing [102@from-did-direct:1] GotoIf("SIP/from-SipProvider-00000002", "1?ext-local,102,1:followme-check,102,1") in new stack
    -- Goto (ext-local,102,1)
    -- Executing [102@ext-local:1] Set("SIP/from-SipProvider-00000002", "__RINGTIMER=15") in new stack
    -- Executing [102@ext-local:2] Macro("SIP/from-SipProvider-00000002", "exten-vm,novm,102,0,0,0") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/from-SipProvider-00000002", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/from-SipProvider-00000002", "TOUCH_MONITOR=1525634691.2") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/from-SipProvider-00000002", "AMPUSER=9995556666") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/from-SipProvider-00000002", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("SIP/from-SipProvider-00000002", "1?Set(REALCALLERIDNUM=9995556666)") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/from-SipProvider-00000002", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/from-SipProvider-00000002", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/from-SipProvider-00000002", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:8] ExecIf("SIP/from-SipProvider-00000002", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
    -- Executing [s@macro-user-callerid:9] GotoIf("SIP/from-SipProvider-00000002", "1?report") in new stack
    -- Goto (macro-user-callerid,s,16)
    -- Executing [s@macro-user-callerid:16] NoOp("SIP/from-SipProvider-00000002", "Macro Depth is 2") in new stack
    -- Executing [s@macro-user-callerid:17] GotoIf("SIP/from-SipProvider-00000002", "1?report2:macroerror") in new stack
    -- Goto (macro-user-callerid,s,18)
    -- Executing [s@macro-user-callerid:18] GotoIf("SIP/from-SipProvider-00000002", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:19] ExecIf("SIP/from-SipProvider-00000002", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
    -- Executing [s@macro-user-callerid:20] Set("SIP/from-SipProvider-00000002", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:21] GotoIf("SIP/from-SipProvider-00000002", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,37)
    -- Executing [s@macro-user-callerid:37] Set("SIP/from-SipProvider-00000002", "CALLERID(number)=9995556666") in new stack
    -- Executing [s@macro-user-callerid:38] Set("SIP/from-SipProvider-00000002", "CALLERID(name)=+19995556666") in new stack
    -- Executing [s@macro-user-callerid:39] GotoIf("SIP/from-SipProvider-00000002", "0?cnum") in new stack
    -- Executing [s@macro-user-callerid:40] Set("SIP/from-SipProvider-00000002", "CDR(cnam)=+19995556666") in new stack
    -- Executing [s@macro-user-callerid:41] Set("SIP/from-SipProvider-00000002", "CDR(cnum)=9995556666") in new stack
    -- Executing [s@macro-user-callerid:42] Set("SIP/from-SipProvider-00000002", "CHANNEL(language)=en") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/from-SipProvider-00000002", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/from-SipProvider-00000002", "__EXTTOCALL=102") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/from-SipProvider-00000002", "__PICKUPMARK=102") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/from-SipProvider-00000002", "RT=") in new stack
[2018-05-06 15:25:05] ERROR[8905][C-00000001]: pbx_functions.c:608 ast_func_read: Function PJSIP_HEADER not registered
    -- Executing [s@macro-exten-vm:6] ExecIf("SIP/from-SipProvider-00000002", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
[2018-05-06 15:25:05] ERROR[8905][C-00000001]: pbx_functions.c:651 ast_func_read2: Function PJSIP_HEADER not registered
[2018-05-06 15:25:05] ERROR[8905][C-00000001]: pbx_functions.c:608 ast_func_read: Function PJSIP_HEADER not registered
    -- Executing [s@macro-exten-vm:7] ExecIf("SIP/from-SipProvider-00000002", "0?MacroExit()") in new stack
[2018-05-06 15:25:05] ERROR[8905][C-00000001]: pbx_functions.c:651 ast_func_read2: Function PJSIP_HEADER not registered
[2018-05-06 15:25:05] ERROR[8905][C-00000001]: pbx_functions.c:608 ast_func_read: Function PJSIP_HEADER not registered
    -- Executing [s@macro-exten-vm:8] ExecIf("SIP/from-SipProvider-00000002", "0?Gosub(ext-intercom,*80102,1())") in new stack
[2018-05-06 15:25:05] ERROR[8905][C-00000001]: pbx_functions.c:651 ast_func_read2: Function PJSIP_HEADER not registered
[2018-05-06 15:25:05] ERROR[8905][C-00000001]: pbx_functions.c:608 ast_func_read: Function PJSIP_HEADER not registered
    -- Executing [s@macro-exten-vm:9] ExecIf("SIP/from-SipProvider-00000002", "0?MacroExit()") in new stack
[2018-05-06 15:25:05] ERROR[8905][C-00000001]: pbx_functions.c:651 ast_func_read2: Function PJSIP_HEADER not registered
[2018-05-06 15:25:05] ERROR[8905][C-00000001]: pbx_functions.c:608 ast_func_read: Function PJSIP_HEADER not registered
    -- Executing [s@macro-exten-vm:10] ExecIf("SIP/from-SipProvider-00000002", "0?ChanSpy(SIP/102,q)") in new stack
[2018-05-06 15:25:05] ERROR[8905][C-00000001]: pbx_functions.c:651 ast_func_read2: Function PJSIP_HEADER not registered
[2018-05-06 15:25:05] ERROR[8905][C-00000001]: pbx_functions.c:608 ast_func_read: Function PJSIP_HEADER not registered
    -- Executing [s@macro-exten-vm:11] ExecIf("SIP/from-SipProvider-00000002", "0?MacroExit()") in new stack
[2018-05-06 15:25:05] ERROR[8905][C-00000001]: pbx_functions.c:651 ast_func_read2: Function PJSIP_HEADER not registered
    -- Executing [s@macro-exten-vm:12] Gosub("SIP/from-SipProvider-00000002", "sub-record-check,s,1(exten,102,dontcare)") in new stack
    -- Executing [s@sub-record-check:1] GotoIf("SIP/from-SipProvider-00000002", "10?initialized") in new stack
    -- Goto (sub-record-check,s,10)
    -- Executing [s@sub-record-check:10] NoOp("SIP/from-SipProvider-00000002", "Recordings initialized") in new stack
    -- Executing [s@sub-record-check:11] ExecIf("SIP/from-SipProvider-00000002", "0?Set(ARG3=dontcare)") in new stack
    -- Executing [s@sub-record-check:12] Set("SIP/from-SipProvider-00000002", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:13] ExecIf("SIP/from-SipProvider-00000002", "0?Set(REC_STATUS=NO)") in new stack
    -- Executing [s@sub-record-check:14] GotoIf("SIP/from-SipProvider-00000002", "5?checkaction") in new stack
    -- Goto (sub-record-check,s,17)
    -- Executing [s@sub-record-check:17] GotoIf("SIP/from-SipProvider-00000002", "1?sub-record-check,exten,1") in new stack
    -- Goto (sub-record-check,exten,1)
    -- Executing [exten@sub-record-check:1] NoOp("SIP/from-SipProvider-00000002", "Exten Recording Check between 9995556666 and 102") in new stack
    -- Executing [exten@sub-record-check:2] Set("SIP/from-SipProvider-00000002", "CALLTYPE=external") in new stack
    -- Executing [exten@sub-record-check:3] ExecIf("SIP/from-SipProvider-00000002", "0?Set(CALLTYPE=)") in new stack
    -- Executing [exten@sub-record-check:4] Set("SIP/from-SipProvider-00000002", "CALLEE=dontcare") in new stack
    -- Executing [exten@sub-record-check:5] ExecIf("SIP/from-SipProvider-00000002", "0?Set(CALLEE=dontcare)") in new stack
    -- Executing [exten@sub-record-check:6] GotoIf("SIP/from-SipProvider-00000002", "1?callee") in new stack
    -- Goto (sub-record-check,exten,11)
    -- Executing [exten@sub-record-check:11] Gosub("SIP/from-SipProvider-00000002", "recordcheck,1(dontcare,external,102)") in new stack
    -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/from-SipProvider-00000002", "Starting recording check against dontcare") in new stack
    -- Executing [recordcheck@sub-record-check:2] Goto("SIP/from-SipProvider-00000002", "dontcare") in new stack
    -- Goto (sub-record-check,recordcheck,3)
    -- Executing [recordcheck@sub-record-check:3] Return("SIP/from-SipProvider-00000002", "") in new stack
    -- Executing [exten@sub-record-check:12] Return("SIP/from-SipProvider-00000002", "") in new stack
    -- Executing [s@macro-exten-vm:13] GotoIf("SIP/from-SipProvider-00000002", "1?macrodial") in new stack
    -- Goto (macro-exten-vm,s,19)
    -- Executing [s@macro-exten-vm:19] GosubIf("SIP/from-SipProvider-00000002", "0?clrheader,1()") in new stack
    -- Executing [s@macro-exten-vm:20] Macro("SIP/from-SipProvider-00000002", "dial-one,,HhTtr,102") in new stack
    -- Executing [s@macro-dial-one:1] Set("SIP/from-SipProvider-00000002", "DEXTEN=102") in new stack
    -- Executing [s@macro-dial-one:2] ExecIf("SIP/from-SipProvider-00000002", "0?Set(__EXTTOCALL=102)") in new stack
    -- Executing [s@macro-dial-one:3] Set("SIP/from-SipProvider-00000002", "DIALSTATUS_CW=") in new stack
    -- Executing [s@macro-dial-one:4] GosubIf("SIP/from-SipProvider-00000002", "0?screen,1()") in new stack
    -- Executing [s@macro-dial-one:5] GosubIf("SIP/from-SipProvider-00000002", "0?cf,1()") in new stack
    -- Executing [s@macro-dial-one:6] GotoIf("SIP/from-SipProvider-00000002", "1?skip1") in new stack
    -- Goto (macro-dial-one,s,9)
    -- Executing [s@macro-dial-one:9] GotoIf("SIP/from-SipProvider-00000002", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:10] GotoIf("SIP/from-SipProvider-00000002", "0?continue") in new stack
    -- Executing [s@macro-dial-one:11] ExecIf("SIP/from-SipProvider-00000002", "0?Set(D_OPTIONS=g)") in new stack
    -- Executing [s@macro-dial-one:12] Set("SIP/from-SipProvider-00000002", "EXTHASCW=ENABLED") in new stack
    -- Executing [s@macro-dial-one:13] GotoIf("SIP/from-SipProvider-00000002", "0?next1:cwinusebusy") in new stack
    -- Goto (macro-dial-one,s,25)
    -- Executing [s@macro-dial-one:25] GotoIf("SIP/from-SipProvider-00000002", "0?next3:continue") in new stack
    -- Goto (macro-dial-one,s,27)
    -- Executing [s@macro-dial-one:27] GotoIf("SIP/from-SipProvider-00000002", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:28] GosubIf("SIP/from-SipProvider-00000002", "1?dstring,1():dlocal,1()") in new stack
    -- Executing [dstring@macro-dial-one:1] Set("SIP/from-SipProvider-00000002", "DSTRING=") in new stack
    -- Executing [dstring@macro-dial-one:2] Set("SIP/from-SipProvider-00000002", "DEVICES=102&99102") in new stack
    -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/from-SipProvider-00000002", "0?Return()") in new stack
    -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/from-SipProvider-00000002", "0?Set(DEVICES=02&99102)") in new stack
    -- Executing [dstring@macro-dial-one:5] Set("SIP/from-SipProvider-00000002", "LOOPCNT=2") in new stack
    -- Executing [dstring@macro-dial-one:6] Set("SIP/from-SipProvider-00000002", "ITER=1") in new stack
    -- Executing [dstring@macro-dial-one:7] Set("SIP/from-SipProvider-00000002", "THISDIAL=SIP/102") in new stack
    -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/from-SipProvider-00000002", "1?zap2dahdi,1()") in new stack
    -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/from-SipProvider-00000002", "0?Return()") in new stack
    -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/from-SipProvider-00000002", "NEWDIAL=") in new stack
    -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/from-SipProvider-00000002", "LOOPCNT2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/from-SipProvider-00000002", "ITER2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/from-SipProvider-00000002", "THISPART2=SIP/102") in new stack
    -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/from-SipProvider-00000002", "0?Set(THISPART2=DAHDI/102)") in new stack
    -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/from-SipProvider-00000002", "NEWDIAL=SIP/102&") in new stack
    -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/from-SipProvider-00000002", "ITER2=2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/from-SipProvider-00000002", "0?begin2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/from-SipProvider-00000002", "THISDIAL=SIP/102") in new stack
    -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/from-SipProvider-00000002", "") in new stack
    -- Executing [dstring@macro-dial-one:9] GotoIf("SIP/from-SipProvider-00000002", "1?docheck") in new stack
    -- Goto (macro-dial-one,dstring,14)
    -- Executing [dstring@macro-dial-one:14] GotoIf("SIP/from-SipProvider-00000002", "0?skipset") in new stack
    -- Executing [dstring@macro-dial-one:15] Set("SIP/from-SipProvider-00000002", "DSTRING=SIP/102&") in new stack
    -- Executing [dstring@macro-dial-one:16] Set("SIP/from-SipProvider-00000002", "ITER=2") in new stack
    -- Executing [dstring@macro-dial-one:17] GotoIf("SIP/from-SipProvider-00000002", "1?begin") in new stack
    -- Goto (macro-dial-one,dstring,7)
    -- Executing [dstring@macro-dial-one:7] Set("SIP/from-SipProvider-00000002", "THISDIAL=SIP/99102") in new stack
    -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/from-SipProvider-00000002", "1?zap2dahdi,1()") in new stack
    -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/from-SipProvider-00000002", "0?Return()") in new stack
    -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/from-SipProvider-00000002", "NEWDIAL=") in new stack
    -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/from-SipProvider-00000002", "LOOPCNT2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/from-SipProvider-00000002", "ITER2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/from-SipProvider-00000002", "THISPART2=SIP/99102") in new stack
    -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/from-SipProvider-00000002", "0?Set(THISPART2=DAHDI/99102)") in new stack
    -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/from-SipProvider-00000002", "NEWDIAL=SIP/99102&") in new stack
    -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/from-SipProvider-00000002", "ITER2=2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/from-SipProvider-00000002", "0?begin2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/from-SipProvider-00000002", "THISDIAL=SIP/99102") in new stack
    -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/from-SipProvider-00000002", "") in new stack
    -- Executing [dstring@macro-dial-one:9] GotoIf("SIP/from-SipProvider-00000002", "1?docheck") in new stack
    -- Goto (macro-dial-one,dstring,14)
    -- Executing [dstring@macro-dial-one:14] GotoIf("SIP/from-SipProvider-00000002", "0?skipset") in new stack
    -- Executing [dstring@macro-dial-one:15] Set("SIP/from-SipProvider-00000002", "DSTRING=SIP/102&SIP/99102&") in new stack
    -- Executing [dstring@macro-dial-one:16] Set("SIP/from-SipProvider-00000002", "ITER=3") in new stack
    -- Executing [dstring@macro-dial-one:17] GotoIf("SIP/from-SipProvider-00000002", "0?begin") in new stack
    -- Executing [dstring@macro-dial-one:18] ExecIf("SIP/from-SipProvider-00000002", "0?Return()") in new stack
    -- Executing [dstring@macro-dial-one:19] Set("SIP/from-SipProvider-00000002", "DSTRING=SIP/102&SIP/99102") in new stack
    -- Executing [dstring@macro-dial-one:20] Return("SIP/from-SipProvider-00000002", "") in new stack
    -- Executing [s@macro-dial-one:29] GotoIf("SIP/from-SipProvider-00000002", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:30] GotoIf("SIP/from-SipProvider-00000002", "0?skiptrace") in new stack
    -- Executing [s@macro-dial-one:31] GosubIf("SIP/from-SipProvider-00000002", "1?ctset,1():ctclear,1()") in new stack
    -- Executing [ctset@macro-dial-one:1] Set("SIP/from-SipProvider-00000002", "DB(CALLTRACE/102)=9995556666") in new stack
    -- Executing [ctset@macro-dial-one:2] Return("SIP/from-SipProvider-00000002", "") in new stack
    -- Executing [s@macro-dial-one:32] Set("SIP/from-SipProvider-00000002", "D_OPTIONS=HhTtr") in new stack
    -- Executing [s@macro-dial-one:33] NoOp("SIP/from-SipProvider-00000002", "Blind Transfer: , Attended Transfer: , User: , Alert Info: ") in new stack
    -- Executing [s@macro-dial-one:34] ExecIf("SIP/from-SipProvider-00000002", "0?Set(ALERT_INFO=)") in new stack
    -- Executing [s@macro-dial-one:35] ExecIf("SIP/from-SipProvider-00000002", "0?Set(ALERT_INFO=)") in new stack
    -- Executing [s@macro-dial-one:36] ExecIf("SIP/from-SipProvider-00000002", "0?Set(ALERT_INFO=)") in new stack
    -- Executing [s@macro-dial-one:37] ExecIf("SIP/from-SipProvider-00000002", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
    -- Executing [s@macro-dial-one:38] ExecIf("SIP/from-SipProvider-00000002", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
    -- Executing [s@macro-dial-one:39] GosubIf("SIP/from-SipProvider-00000002", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
    -- Executing [s@macro-dial-one:40] ExecIf("SIP/from-SipProvider-00000002", "0?Set(CHANNEL(musicclass)=)") in new stack
    -- Executing [s@macro-dial-one:41] GosubIf("SIP/from-SipProvider-00000002", "0?qwait,1()") in new stack
    -- Executing [s@macro-dial-one:42] Set("SIP/from-SipProvider-00000002", "__CWIGNORE=") in new stack
    -- Executing [s@macro-dial-one:43] Set("SIP/from-SipProvider-00000002", "__KEEPCID=TRUE") in new stack
    -- Executing [s@macro-dial-one:44] GotoIf("SIP/from-SipProvider-00000002", "0?usegoto,1") in new stack
    -- Executing [s@macro-dial-one:45] GotoIf("SIP/from-SipProvider-00000002", "1?godial") in new stack
    -- Goto (macro-dial-one,s,50)
    -- Executing [s@macro-dial-one:50] Macro("SIP/from-SipProvider-00000002", "dialout-one-predial-hook,") in new stack
    -- Executing [s@macro-dialout-one-predial-hook:1] MacroExit("SIP/from-SipProvider-00000002", "") in new stack
    -- Executing [s@macro-dial-one:51] ExecIf("SIP/from-SipProvider-00000002", "1?Set(D_OPTIONS=HhtrI)") in new stack
    -- Executing [s@macro-dial-one:52] NoOp("SIP/from-SipProvider-00000002", "") in new stack
    -- Executing [s@macro-dial-one:53] Dial("SIP/from-SipProvider-00000002", "SIP/102&SIP/99102,,HhtrIb(func-apply-sipheaders^s^1)") in new stack
[2018-05-06 15:25:05] WARNING[8905][C-00000001]: app_dial.c:2525 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == DTLS ECDH initialized (automatic), faster PFS enabled
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- SIP/99102-00000003 Internal Gosub(func-apply-sipheaders,s,1) start
    -- Executing [s@func-apply-sipheaders:1] NoOp("SIP/99102-00000003", "Applying SIP Headers to channel") in new stack
    -- Executing [s@func-apply-sipheaders:2] Set("SIP/99102-00000003", "SIPHEADERKEYS=") in new stack
    -- Executing [s@func-apply-sipheaders:3] While("SIP/99102-00000003", "0") in new stack
    -- Jumping to priority 6
    -- Executing [s@func-apply-sipheaders:7] Return("SIP/99102-00000003", "") in new stack
  == Spawn extension (from-internal, 102, 1) exited non-zero on 'SIP/99102-00000003'
    -- SIP/99102-00000003 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
Really destroying SIP dialog '2b1f651b47e1608417d8f05e53026266@[::1]:5060' Method: INVITE
Audio is at 16386
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding codec g722 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 123.45.67.890:64208:
INVITE sip:grvf61iq@k8ld3ajkr543.invalid;transport=wss SIP/2.0
Via: SIP/2.0/WS 11.22.33.44:0;branch=z9hG4bK25f6e49a
Max-Forwards: 70
From: "+19995556666" <sip:9995556666@11.22.33.44:0>;tag=as70ee86f8
To: <sip:grvf61iq@k8ld3ajkr543.invalid;transport=wss>
Contact: <sip:9995556666@11.22.33.44:0;transport=ws>
Call-ID: 595f29653ae63535489564ee2eebe488@11.22.33.44:0
CSeq: 102 INVITE
User-Agent: FPBX-14.0.3.1(13.19.1)
Date: Sun, 06 May 2018 19:25:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
P-Asserted-Identity: "+19995556666" <sip:9995556666@11.22.33.44>
Content-Type: application/sdp
Content-Length: 752

v=0
o=root 1115761169 1115761169 IN IP4 11.22.33.44
s=Asterisk PBX 13.19.1
c=IN IP4 11.22.33.44
t=0 0
m=audio 16386 RTP/SAVPF 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=ice-ufrag:2df544f2623d7b86682dbf1f0a0eb10b
a=ice-pwd:07a7a6504fd7779669833e0c52f298fa
a=candidate:H8c520b6d 1 UDP 2130706431 11.22.33.44 16386 typ host
a=candidate:H8c520b6d 2 UDP 2130706430 11.22.33.44 16387 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 D5:F5:FB:0F:40:34:D4:BF:F5:CD:36:BD:3C:6F:E9:9D:D8:51:EE:AD:9F:3B:17:63:F6:75:C5:03:5D:26:F5:8D
a=rtcp-mux
a=sendrecv

---
    -- Called SIP/99102
    -- Connected line update to SIP/from-SipProvider-00000002 prevented.

<--- SIP read from WS:123.45.67.890:64208 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS 11.22.33.44:0;branch=z9hG4bK25f6e49a
To: <sip:grvf61iq@k8ld3ajkr543.invalid;transport=wss>
From: "+19995556666" <sip:9995556666@11.22.33.44:0>;tag=as70ee86f8
Call-ID: 595f29653ae63535489564ee2eebe488@11.22.33.44:0
CSeq: 102 INVITE
Supported: outbound
User-Agent: SIP.js/0.7.7
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from WS:123.45.67.890:64208 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 11.22.33.44:0;branch=z9hG4bK25f6e49a
To: <sip:grvf61iq@k8ld3ajkr543.invalid;transport=wss>;tag=f4nqbn0gdn
From: "+19995556666" <sip:9995556666@11.22.33.44:0>;tag=as70ee86f8
Call-ID: 595f29653ae63535489564ee2eebe488@11.22.33.44:0
CSeq: 102 INVITE
Contact: <sip:grvf61iq@k8ld3ajkr543.invalid;transport=wss>
Supported: outbound
User-Agent: SIP.js/0.7.7
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:grvf61iq@k8ld3ajkr543.invalid;transport=wss>
    -- SIP/99102-00000003 is ringing
Reliably Transmitting (no NAT) to 123.45.67.890:64208:
OPTIONS sip:grvf61iq@k8ld3ajkr543.invalid;transport=wss SIP/2.0
Via: SIP/2.0/WS 11.22.33.44:0;branch=z9hG4bK66f41a8b
Max-Forwards: 70
From: "Unknown" <sip:Unknown@11.22.33.44:0>;tag=as4275866c
To: <sip:grvf61iq@k8ld3ajkr543.invalid;transport=wss>
Contact: <sip:Unknown@11.22.33.44:0;transport=ws>
Call-ID: 69721aea310a096d56bf680856caa01e@11.22.33.44:0
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.3.1(13.19.1)
Date: Sun, 06 May 2018 19:25:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---

<--- SIP read from WS:123.45.67.890:64208 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 11.22.33.44:0;branch=z9hG4bK66f41a8b
To: <sip:grvf61iq@k8ld3ajkr543.invalid;transport=wss>;tag=c9i2heejlc
From: "Unknown" <sip:Unknown@11.22.33.44:0>;tag=as4275866c
Call-ID: 69721aea310a096d56bf680856caa01e@11.22.33.44:0
CSeq: 102 OPTIONS
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Accept: application/sdp,application/dtmf-relay
Supported: outbound
User-Agent: SIP.js/0.7.7
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from WS:123.45.67.890:64208 --->
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/WS 11.22.33.44:0;branch=z9hG4bK25f6e49a
To: <sip:grvf61iq@k8ld3ajkr543.invalid;transport=wss>;tag=f4nqbn0gdn
From: "+19995556666" <sip:9995556666@11.22.33.44:0>;tag=as70ee86f8
Call-ID: 595f29653ae63535489564ee2eebe488@11.22.33.44:0
CSeq: 102 INVITE
Supported: outbound
User-Agent: SIP.js/0.7.7
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 123.45.67.890:64208:
ACK sip:grvf61iq@k8ld3ajkr543.invalid;transport=wss SIP/2.0
Via: SIP/2.0/WS 11.22.33.44:0;branch=z9hG4bK25f6e49a
Max-Forwards: 70
From: "+19995556666" <sip:9995556666@11.22.33.44:0>;tag=as70ee86f8
To: <sip:grvf61iq@k8ld3ajkr543.invalid;transport=wss>;tag=f4nqbn0gdn
Contact: <sip:9995556666@11.22.33.44:0;transport=ws>
Call-ID: 595f29653ae63535489564ee2eebe488@11.22.33.44:0
CSeq: 102 ACK
User-Agent: FPBX-14.0.3.1(13.19.1)
Content-Length: 0


---
    -- Redirecting update to SIP/from-SipProvider-00000002 prevented.
    -- SIP/99102-00000003 is busy
Scheduling destruction of SIP dialog '595f29653ae63535489564ee2eebe488@11.22.33.44:0' in 13632 ms (Method: INVITE)
  == Everyone is busy/congested at this time (2:1/0/1)
    -- Executing [s@macro-dial-one:54] ExecIf("SIP/from-SipProvider-00000002", "0?MacroExit()") in new stack
    -- Executing [s@macro-dial-one:55] ExecIf("SIP/from-SipProvider-00000002", "0?Set(DIALSTATUS=)") in new stack
    -- Executing [s@macro-dial-one:56] GosubIf("SIP/from-SipProvider-00000002", "0?s-BUSY,1()") in new stack
    -- Executing [s@macro-dial-one:57] MacroExit("SIP/from-SipProvider-00000002", "") in new stack
    -- Executing [s@macro-exten-vm:21] Set("SIP/from-SipProvider-00000002", "SV_DIALSTATUS=BUSY") in new stack
    -- Executing [s@macro-exten-vm:22] GosubIf("SIP/from-SipProvider-00000002", "0?docfu,1()") in new stack
    -- Executing [s@macro-exten-vm:23] GosubIf("SIP/from-SipProvider-00000002", "0?docfb,1()") in new stack
    -- Executing [s@macro-exten-vm:24] Set("SIP/from-SipProvider-00000002", "DIALSTATUS=BUSY") in new stack
    -- Executing [s@macro-exten-vm:25] ExecIf("SIP/from-SipProvider-00000002", "0?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:26] GotoIf("SIP/from-SipProvider-00000002", "1?s-BUSY,1") in new stack
    -- Goto (macro-exten-vm,s-BUSY,1)
    -- Executing [s-BUSY@macro-exten-vm:1] GotoIf("SIP/from-SipProvider-00000002", "0?exit,1") in new stack
    -- Executing [s-BUSY@macro-exten-vm:2] PlayTones("SIP/from-SipProvider-00000002", "busy") in new stack
    -- Executing [s-BUSY@macro-exten-vm:3] Busy("SIP/from-SipProvider-00000002", "20") in new stack
Really destroying SIP dialog '69721aea310a096d56bf680856caa01e@11.22.33.44:0' Method: OPTIONS
Reliably Transmitting (NAT) to 10.192.168.10:5060:
OPTIONS sip:sip.SipProvider.com SIP/2.0
Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bK720f942f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@11.22.33.44>;tag=as2d2717c4
To: <sip:sip.SipProvider.com>
Contact: <sip:Unknown@11.22.33.44:5060>
Call-ID: 04ef14881f30862903dc03b13587ea92@11.22.33.44:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.3.1(13.19.1)
Date: Sun, 06 May 2018 19:25:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.192.168.10:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bK720f942f;rport=5060;received=11.22.33.44
From: "Unknown" <sip:Unknown@11.22.33.44>;tag=as2d2717c4
To: <sip:sip.SipProvider.com>;tag=dfb4940bfc7117e4d7fa62ed6ef36d37.6706
Call-ID: 04ef14881f30862903dc03b13587ea92@11.22.33.44:5060
CSeq: 102 OPTIONS
Server: kamailio (5.0.6 (x86_64/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '04ef14881f30862903dc03b13587ea92@11.22.33.44:5060' Method: OPTIONS

<--- SIP read from UDP:10.192.168.10:5060 --->
BYE sip:18881112222@11.22.33.44:5060 SIP/2.0
Via: SIP/2.0/UDP 10.192.168.10;branch=z9hG4bKa17b.0e47cbc389c49d9be866887adaf7eed0.0
v:SIP/2.0/UDP 10.15.3.4:5082;received=10.15.3.4;rport=5082;branch=z9hG4bK2DB0pBv1vj6ar
Max-Forwards:69
f:"+19995556666"<sip:9995556666@sip.SipProvider.com>;tag=6N5B5QvNgv79B
t:<sip:18881112222@11.22.33.44:5060>;tag=as5e93e1d4
i:fd2cdd4a-cc05-1236-c3ab-02420a0f0304
CSeq:122481282 BYE
Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,NOTIFY
k:timer,path,replaces
Reason:Q.850;cause=16;text="NORMAL_CLEARING"
l:0

<------------->
--- (12 headers 0 lines) ---
Sending to 10.192.168.10:5060 (NAT)
Scheduling destruction of SIP dialog 'fd2cdd4a-cc05-1236-c3ab-02420a0f0304' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 10.192.168.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.192.168.10;branch=z9hG4bKa17b.0e47cbc389c49d9be866887adaf7eed0.0;received=10.192.168.10;rport=5060
Via: SIP/2.0/UDP 10.15.3.4:5082;received=10.15.3.4;rport=5082;branch=z9hG4bK2DB0pBv1vj6ar
From: "+19995556666"<sip:9995556666@sip.SipProvider.com>;tag=6N5B5QvNgv79B
To: <sip:18881112222@11.22.33.44:5060>;tag=as5e93e1d4
Call-ID: fd2cdd4a-cc05-1236-c3ab-02420a0f0304
CSeq: 122481282 BYE
Server: FPBX-14.0.3.1(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (macro-exten-vm, s-BUSY, 3) exited non-zero on 'SIP/from-SipProvider-00000002' in macro 'exten-vm'
  == Spawn extension (ext-local, 102, 2) exited non-zero on 'SIP/from-SipProvider-00000002'
    -- Executing [h@ext-local:1] Macro("SIP/from-SipProvider-00000002", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/from-SipProvider-00000002", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] ExecIf("SIP/from-SipProvider-00000002", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:4] NoOp("SIP/from-SipProvider-00000002", " monior file= ") in new stack
    -- Executing [s@macro-hangupcall:5] AGI("SIP/from-SipProvider-00000002", "attendedtransfer-rec-restart.php,,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
    -- <SIP/from-SipProvider-00000002>AGI Script attendedtransfer-rec-restart.php completed, returning 0
    -- Executing [s@macro-hangupcall:6] Hangup("SIP/from-SipProvider-00000002", "") in new stack
  == Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'SIP/from-SipProvider-00000002' in macro 'hangupcall'
  == Spawn extension (ext-local, h, 1) exited non-zero on 'SIP/from-SipProvider-00000002'

If easier to read on pastebin, here’s a link: https://pastebin.freepbx.org/view/ed65bf06

Appreciate any ideas.

Thanks

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Multiple Fail2Ban notifications.... where should I be looking?

May 7, 2018, 3:38 am
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@jpaquin wrote:

image
image.png1513x334 129 KB

Just in the past two days I’m getting multiple notifications (and seeing the corresponding attempts in the Asterisk log) for Fail2Ban bans.

Shouldn’t the initial ban be lasting longer? I shouldn’t see the bad guy getting to have multiple attempts this close together, right?

Where should I be looking and what should I be looking for?

FreePBX 14.0.3.1
Current Asterisk Version: 13.19.1
No outstanding updates.

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Yum update error

May 7, 2018, 5:26 am
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@vespino wrote:

I’m getting these errors when running yum update:

Downloading packages:
asterisk15-addons-15.4.0-1.sng FAILED 0% [ ] 0.0 B/s | 0 B --:–:-- ETA
http://sng7.com/sng7/sng7/RPMS/asterisk15-addons-15.4.0-1.sng7.x86_64.rpm: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=sng-pkgs clean metadata ] 0.0 B/s | 0 B --:–:-- ETA
Trying other mirror.

Error downloading packages:
asterisk15-addons-core-15.4.0-1.sng7.x86_64: [Errno 256] No more mirrors to try.

I have tried the given suggestion and ran the command again, but without result. Could this be just a temporary issue?

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Custom Dialplan is not working from IVR

May 7, 2018, 8:14 am
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@faisalkhan wrote:

hi guys,

I have written a dialplan in extensions_custom.conf for passing some dtmfs.

for example I have to dial a complete number(12125264560) and then extension. now what I did is that I am passing the dtmf’s of extension.

[from-internal-custom]
exten => 221,1,Set(CALLERID(num)=121252555)
exten => 221,n,NoOp(I am in Quick Dialing)
exten => 221,n,Dial(SIP/SIPROUTES/12125264560,20,M(221))

[macro-221]
exten => s,1,Answer()
exten => s,n,Wait(1)
exten => s,n,SendDTMF(221)
exten => s,n,Wait(1)

now when someone dials into my IVR Number which is different like e.g 2126090000 and they want to dial this 221 which is not on our server but we want this extension to be dialed by the customer. It’s not working.

Is there any solution that we can use this custom dialplan to work with our internal dialplan.

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[SOLVED] Queue blf hints

May 7, 2018, 8:30 am
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@bajramia wrote:

Hi All,
I’m trying to set queue BLF so when the agent is logged in the queue the light to stay on I don’t know if this is possible.

Thank you for you help

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Sangoma S500 Phone, assigning BLF crashes phone

May 7, 2018, 8:55 am
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@bbb wrote:

hi there.

i have a couple of freepbx running and i am absolutely pleased with it. this is just a great piece of software. thank you for that!

now i am about to do a fresh installation of a freepbx system, along with sangoma phones.

freepbx version: 12.7.4-1804-2.sng7
sangoma s500 fw version: 2.0.4.48
as far as i can tell, it is all up-to-date. last updates installed just right now.

provisioning is going over dhcp option and tftp. and it works like a charm.

now to my issues. it seems i cannot assign blf buttons to the phones. when a blf button is activated through endpoint manager, the phone constantly reboots.

i am proceeding like the wiki-article suggests:
https://wiki.freepbx.org/display/PHON/Setting+up+BLF+in+End+Point+Manager

that is basically:

  1. opening the sangoma template
  2. on available phones clicking the s500, the editor pops open
  3. picking for example line key 2, setting type to blf, and inserting label 11 and value 11, since that is the extension i want to monitor

hint, and that looks important to me: i cannot change the alert mode. it is preselected to none. if i click one of the other possible states, it will deactivate to never be activated again. not visual, not audio, and not none. closing the template and reopening it, highlights again none.
EDIT: there is a setting on the options tab of the template “BLF Alert”. this is working and is also correctly changing the state of the option within the selected phone model, if applied.

  1. save model, and then recreate the config file.

when i now propagate the change to the phone - force update phone config - the phone will start rebooting until i remove the blf mapping from the template.

during those reboots, the line key (#1) shows the phone will not register to the freepbx backend.

i hope the description is clear enough. did i leave anything out? i really hope someone knows what is going on. since the tight integration between freepbx and sangoma phones was my key to picking those.

thanks for any idea.

best,
sebastian

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LDAP on non-sangoma Phone

May 7, 2018, 4:22 pm
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@matthewljensen wrote:

I’m trying to setup a good way for our phones to get a phonebook from our FreePBX server. We have grandstream 2170s. Currently, I use advice from here: https://www.reddit.com/r/freepbx/comments/4ezz06/freepbx_grandstream_directory_automation/. That works ok, but we are considering using the gswave app on some cellphones, and the only way to get a phonebook on those is through LDAP. From reading on the wiki, it looks like that might only work with Sangoma phones. Is it not possible to set that up on other phones? Where would those settings be?

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Lagged extensions

May 8, 2018, 4:05 am
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@fetoa wrote:

Hi guys!

I have 2 extensions that are lagged every 40-50 minutes more or less. I get errores like this:
[2018-05-07 04:03:00] NOTICE[15091] chan_sip.c: Peer ‘301’ is now Lagged. (2048ms / 2000ms)
[2018-05-07 04:03:10] NOTICE[15091] chan_sip.c: Peer ‘301’ is now Reachable. (49ms / 2000ms)
[2018-05-07 04:03:44] NOTICE[15091] chan_sip.c: Peer ‘302’ is now UNREACHABLE! Last qualify: 57
[2018-05-07 04:03:55] NOTICE[15091] chan_sip.c: Peer ‘302’ is now Reachable. (50ms / 2000ms)

Both phones are Linksys T23G and are connected via VPN, using each phone openvpn client.

If I execute sip show clients, this is what I get:

301/301 10.8.0.3 D Yes Yes A 5060 OK (90 ms)
302/302 10.8.0.4 D Yes Yes A 5060 OK (88 ms)

Any suggestion to solve this? May be because of the VPN connection of each phone?

Regards

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Few seconds silence on one way on VPN calls

May 8, 2018, 4:24 am
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@fetoa wrote:

Hi again,

I’m suffering an issue whit some calls and it only happens with Yealink phones whit openVPN client and FreePBX 13, with System Admin Pro’s openvpn server.

It normally works in the good way. Sound quality and latency is good, but sometimes, I get few seconds silence on one way. It use to be no more than 5 seconds aproximatelly.

I have only detected with calls between yealink phones that use their own openvpn client integrated and I think is a networking issue.

Does anyone know this case?

Regards.

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Module admin modules missing after upgrade

May 8, 2018, 5:05 am
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@tassie wrote:

Hi there,

I decided to upgrade a freepbx vm from 2.11 to 12 today and all appeared to go smoothly. All modules were up to date, all the latest package updates applied and everything looked fine for the upgrade. But after the upgrade process all the modules are now missing from module admin. I have only two modules listed FreePBX framework 12.0.76.4 (enabled) and System Admin. I’ve attempted to check on line but nothing happens but when looking at FreePBX system status it tells me 34 modules are available for upgrade.
I’m hoping that fixing this problem will allow me to updated all these modules as well as the 4 vulnerable modules I have listed as well.

I’ve searched the forum but can’t seem to find this problem anywhere. Can someone please shed some light on the problem.

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Difference between Trunk context

May 8, 2018, 5:37 am
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@PitzKey wrote:

I’m testing PJSIP Trunks for the first time, in our ChanSIP setups, in the user details on incoming calls we have set context=from-trunk.
I just tested setting up a PJSIP Trunk with the same provider and left the the default context from-pstn it worked fine, then I changed the context to from-trunk and it also works.

What does it affect? What is the difference?

I hope someone can explain this.

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UCP - Error Message

May 8, 2018, 7:08 am
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@MC1 wrote:

Hello
When I try and play a message UCP, I get the “There was an error. See the console log for more details.”

Where is this log located and what might be causing the error message?

Console%20Error%20Message

Thanks in advance for any help that can be provided.

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Chanspy help

May 8, 2018, 7:33 am
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@chasemixon wrote:

I have been asked to configure chanspy for a local user, I have done a couple of searches but everything is really old for the results. I using a distro and at the top of the extensions.conf file it plainly says “DO NOT MODIFY” this file so can someone explain to me how chanspy works? I put this in my Extensions_custom.conf file and I read where I need to put the include in the extensions.conf file right now it has a # sign in front of it, I’m guessing that means ignore this statement, like a “REM” in batch files? the goal is to allow one user to listen to another user while she is being trained, so he would like to be able to listen, and whisper.
I found this code and put it in the extension_custom.conf. but I sure would like to understand what it means… :slight_smile:

[ext-local-custom]

;listen
exten => 556,1,Macro(user-callerid)
exten => 556,n,Authenticate(1234)
exten => 556,n,Read(SPYNUM,agent-newlocation)
exten => 556,n,ChanSpy(SIP/${SPYNUM),q)
exten => 556,n,Hangup

;whisper
exten => 557,1,Macro(user-callerid)
exten => 557,n,Authenticate(1234)
exten => 557,n,Read(SPYNUM,agent-newlocation)
exten => 557,n,ChanSpy(SIP/${SPYNUM),qw)
exten => 557,n,Hangup

;barge
exten => 558,1,Macro(user-callerid)
exten => 558,n,Authenticate(1234)
exten => 558,n,Read(SPYNUM,agent-newlocation)
exten => 558,n,ChanSpy(SIP/${SPYNUM),qB)
exten => 558,n,Hangup

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Queue Pause - Reason Codes

May 8, 2018, 8:15 am
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@comtech wrote:

Does anyone have any experience in introducing different reason codes for queue pausing for agents? How did you implement this?

I know there are solutions like FOP2, that have this functionality built in, but I am wary as support for that app seems very spotty.

We have three pause codes (After Call Work, Training, Break) we would like to report out on. Any ideas?

Thanks!

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Use FreePBX for Cellphone Voicemail - One DID, Multiple Cellphons

May 8, 2018, 9:04 am
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@tuffcalc wrote:

Hi,

Does anyone know if this scenario is possible?

  1. Have on DID as the “access” number;
  2. Change conditional call forwarding on multiple cellphones to point to the “DID access number”.
  3. When a call is forwarded from the cellphone to the DID (e.g., if no answer or out of service), FreePBX through an inbound route knows which voicemail box to route the call to.

Thanks

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How to Diagnose Strange Failure

May 8, 2018, 10:03 am
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@dominic wrote:

I have a system running SNG7 Distro. Maybe once a week I will wind up in a situation where none of my phones can register. If I SSH in and running fwconsole reload it just hangs forever. If I run fwconsole restart, it tries to till asterisk for 30 seconds, then it tries to force kill it, then it says “asterisk is still running and we can’t stop it.” After that I do a full system reboot and everything runs normally for a while.

I can’t for the life of me figure out what is causing this. It always seems to happen in the middle of the night. Looking at /var/log/asterisk/full during the problem it looks like asterisk was just restarted (I see the build and version information, then some VERBOSE level messages about finding config files). Nothing shows up in the log when I attempt to register the phones.

Any ideas on what could be causing this or how I should go about debugging it?

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Hold music

May 8, 2018, 10:29 am
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Easy question = What is the name and location of the Astrisk phone book

May 8, 2018, 12:01 pm
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@gregorywest wrote:

Would like to be able to download the phonebook so that I can put it into another system in real time (or at least batch) do not want to have to do an export all the time.

Can anyone help? Just need that name and location of the Astrick phone book,

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Auto update scheduler not working

May 8, 2018, 12:24 pm
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@netphoneusa wrote:

I have set up auto update for modules and system on the scheduler tab and it never actually performs the update. Am I missing something?

I have the latest distro installed

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