@1reason wrote:

When an office user dials a number to call someone using an outgoing truck, sometimes instead of hearing the expected ringing of the other party, the incoming IVR greeting is played, sometimes, the call is simply terminated.

It doesn’t happen every time, albeit it happens almost every day now (maybe 50% or more lately) What to do?

Here’s my log from the last attempted call that simply terminated:

[2018-12-28 16:19:22] VERBOSE[16445] pbx_variables.c: Setting global variable 'SIPDOMAIN' to '192.168.1.115'
[2018-12-28 16:19:22] VERBOSE[16445] netsock2.c: Using SIP RTP Audio TOS bits 184
[2018-12-28 16:19:22] VERBOSE[16445] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
[2018-12-28 16:19:22] VERBOSE[16445] netsock2.c: Using SIP RTP Audio CoS mark 5
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [828XXXX@from-internal:1] Macro("PJSIP/2210-000002a3", "user-callerid,LIMIT,EXTERNAL,") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:1] Set("PJSIP/2210-000002a3", "TOUCH_MONITOR=1546013962.759") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:2] Set("PJSIP/2210-000002a3", "AMPUSER=2210") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:3] GotoIf("PJSIP/2210-000002a3", "0?report") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:4] ExecIf("PJSIP/2210-000002a3", "1?Set(REALCALLERIDNUM=2210)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:5] Set("PJSIP/2210-000002a3", "AMPUSER=2210") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:6] GotoIf("PJSIP/2210-000002a3", "0?limit") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:7] Set("PJSIP/2210-000002a3", "AMPUSERCIDNAME=Robert Golf Outgoing") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:8] ExecIf("PJSIP/2210-000002a3", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:9] GotoIf("PJSIP/2210-000002a3", "0?report") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:10] Set("PJSIP/2210-000002a3", "AMPUSERCID=2210") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:11] Set("PJSIP/2210-000002a3", "__DIAL_OPTIONS=HhTtr") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:12] Set("PJSIP/2210-000002a3", "CALLERID(all)="Robert Golf Outgoing" <2210>") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:13] GotoIf("PJSIP/2210-000002a3", "0?limit") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:14] ExecIf("PJSIP/2210-000002a3", "1?Set(GROUP(concurrency_limit)=2210)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:15] ExecIf("PJSIP/2210-000002a3", "0?Set(CHANNEL(language)=)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:16] NoOp("PJSIP/2210-000002a3", "Macro Depth is 1") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:17] GotoIf("PJSIP/2210-000002a3", "1?report2:macroerror") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx_builtins.c: Goto (macro-user-callerid,s,18)
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:18] GotoIf("PJSIP/2210-000002a3", "1?continue") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx_builtins.c: Goto (macro-user-callerid,s,37)
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:37] Set("PJSIP/2210-000002a3", "CALLERID(number)=2210") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:38] Set("PJSIP/2210-000002a3", "CALLERID(name)=Robert Golf Outgoing") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:39] GotoIf("PJSIP/2210-000002a3", "0?cnum") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:40] Set("PJSIP/2210-000002a3", "CDR(cnam)=Robert Golf Outgoing") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:41] Set("PJSIP/2210-000002a3", "CDR(cnum)=2210") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-user-callerid:42] Set("PJSIP/2210-000002a3", "CHANNEL(language)=en") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [828XXXX@from-internal:2] Gosub("PJSIP/2210-000002a3", "sub-record-check,s,1(out,828XXXX,force)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@sub-record-check:1] GotoIf("PJSIP/2210-000002a3", "0?initialized") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@sub-record-check:2] Set("PJSIP/2210-000002a3", "__REC_STATUS=INITIALIZED") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@sub-record-check:3] Set("PJSIP/2210-000002a3", "NOW=1546013962") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@sub-record-check:4] Set("PJSIP/2210-000002a3", "__DAY=28") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@sub-record-check:5] Set("PJSIP/2210-000002a3", "__MONTH=12") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@sub-record-check:6] Set("PJSIP/2210-000002a3", "__YEAR=2018") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@sub-record-check:7] Set("PJSIP/2210-000002a3", "__TIMESTR=20181228-161922") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@sub-record-check:8] Set("PJSIP/2210-000002a3", "__FROMEXTEN=2210") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@sub-record-check:9] Set("PJSIP/2210-000002a3", "__MON_FMT=WAV") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@sub-record-check:10] NoOp("PJSIP/2210-000002a3", "Recordings initialized") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@sub-record-check:11] ExecIf("PJSIP/2210-000002a3", "0?Set(ARG3=dontcare)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@sub-record-check:12] Set("PJSIP/2210-000002a3", "REC_POLICY_MODE_SAVE=") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@sub-record-check:13] ExecIf("PJSIP/2210-000002a3", "0?Set(REC_STATUS=NO)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@sub-record-check:14] GotoIf("PJSIP/2210-000002a3", "3?checkaction") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx_builtins.c: Goto (sub-record-check,s,17)
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@sub-record-check:17] GotoIf("PJSIP/2210-000002a3", "1?sub-record-check,out,1") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx_builtins.c: Goto (sub-record-check,out,1)
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [out@sub-record-check:1] NoOp("PJSIP/2210-000002a3", "Outbound Recording Check from 2210 to 828XXXX") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [out@sub-record-check:2] Set("PJSIP/2210-000002a3", "RECMODE=dontcare") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [out@sub-record-check:3] ExecIf("PJSIP/2210-000002a3", "1?Goto(routewins)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx_builtins.c: Goto (sub-record-check,out,7)
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [out@sub-record-check:7] Gosub("PJSIP/2210-000002a3", "recordcheck,1(force,out,828XXXX)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/2210-000002a3", "Starting recording check against force") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [recordcheck@sub-record-check:2] Goto("PJSIP/2210-000002a3", "force") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx_builtins.c: Goto (sub-record-check,recordcheck,5)
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [recordcheck@sub-record-check:5] Set("PJSIP/2210-000002a3", "__REC_POLICY_MODE=FORCE") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [recordcheck@sub-record-check:6] GotoIf("PJSIP/2210-000002a3", "1?startrec") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx_builtins.c: Goto (sub-record-check,recordcheck,16)
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [recordcheck@sub-record-check:16] NoOp("PJSIP/2210-000002a3", "Starting recording: out, 828XXXX") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [recordcheck@sub-record-check:17] Set("PJSIP/2210-000002a3", "__CALLFILENAME=out-828XXXX-2210-20181228-161922-1546013962.759") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [recordcheck@sub-record-check:18] MixMonitor("PJSIP/2210-000002a3", "2018/12/28/out-828XXXX-2210-20181228-161922-1546013962.759.WAV,abi(LOCAL_MIXMON_ID),") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [recordcheck@sub-record-check:19] Set("PJSIP/2210-000002a3", "__MIXMON_ID=0x7f5bf002ce50") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [recordcheck@sub-record-check:20] Set("PJSIP/2210-000002a3", "__RECORD_ID=PJSIP/2210-000002a3") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [recordcheck@sub-record-check:21] Set("PJSIP/2210-000002a3", "__REC_STATUS=RECORDING") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [recordcheck@sub-record-check:22] Set("PJSIP/2210-000002a3", "CDR(recordingfile)=out-828XXXX-2210-20181228-161922-1546013962.759.WAV") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [recordcheck@sub-record-check:23] Return("PJSIP/2210-000002a3", "") in new stack
[2018-12-28 16:19:22] VERBOSE[29831][C-00000110] app_mixmonitor.c: Begin MixMonitor Recording PJSIP/2210-000002a3
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [out@sub-record-check:8] Return("PJSIP/2210-000002a3", "") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [828XXXX@from-internal:3] ExecIf("PJSIP/2210-000002a3", "0 ?Set(CDR(accountcode)=)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [828XXXX@from-internal:4] Set("PJSIP/2210-000002a3", "ROUTE_CIDSAVE="Robert Golf Outgoing" <2210>") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [828XXXX@from-internal:5] Set("PJSIP/2210-000002a3", "MOHCLASS=default") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [828XXXX@from-internal:6] ExecIf("PJSIP/2210-000002a3", "1?Set(TRUNKCIDOVERRIDE=<7155989924>)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [828XXXX@from-internal:7] Set("PJSIP/2210-000002a3", "_NODEST=") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [828XXXX@from-internal:8] Macro("PJSIP/2210-000002a3", "dialout-trunk,2,1715828XXXX,,on") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:1] Set("PJSIP/2210-000002a3", "DIAL_TRUNK=2") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:2] UserEvent("PJSIP/2210-000002a3", "zulu-outbound-call,from:2210,to:1715828XXXX") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:3] ExecIf("PJSIP/2210-000002a3", "0?Set(DIAL_OPTIONS=Hhtr)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:4] GosubIf("PJSIP/2210-000002a3", "0?sub-pincheck,s,1()") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:5] ExecIf("PJSIP/2210-000002a3", "0?Set(CALLERID(num)=2210)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:6] GotoIf("PJSIP/2210-000002a3", "0?disabletrunk,1") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:7] Set("PJSIP/2210-000002a3", "DIAL_NUMBER=1715828XXXX") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:8] Set("PJSIP/2210-000002a3", "DIAL_TRUNK_OPTIONS=HhTtr") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:9] Set("PJSIP/2210-000002a3", "OUTBOUND_GROUP=OUT_2") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:10] Set("PJSIP/2210-000002a3", "DIAL_TRUNK_OPTIONS=T") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:11] GotoIf("PJSIP/2210-000002a3", "0?nomax") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:12] GotoIf("PJSIP/2210-000002a3", "0?chanfull") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:13] GotoIf("PJSIP/2210-000002a3", "0?skipoutcid") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:14] Macro("PJSIP/2210-000002a3", "outbound-callerid,2") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:1] NoOp("PJSIP/2210-000002a3", "2210") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:2] NoOp("PJSIP/2210-000002a3", "") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:3] NoOp("PJSIP/2210-000002a3", "off") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:4] ExecIf("PJSIP/2210-000002a3", "0?Set(CALLERPRES(name-pres)=)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:5] ExecIf("PJSIP/2210-000002a3", "0?Set(CALLERPRES(num-pres)=)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:6] ExecIf("PJSIP/2210-000002a3", "0?Set(REALCALLERIDNUM=2210)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:7] ExecIf("PJSIP/2210-000002a3", "0?Set(AMPUSER=2210)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:8] GotoIf("PJSIP/2210-000002a3", "1?normcid") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx_builtins.c: Goto (macro-outbound-callerid,s,12)
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:12] Set("PJSIP/2210-000002a3", "USEROUTCID=") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:13] Set("PJSIP/2210-000002a3", "EMERGENCYCID=") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:14] Set("PJSIP/2210-000002a3", "TRUNKOUTCID=7155989924") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:15] GotoIf("PJSIP/2210-000002a3", "1?trunkcid") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx_builtins.c: Goto (macro-outbound-callerid,s,20)
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:20] ExecIf("PJSIP/2210-000002a3", "1?Set(CALLERID(all)=7155989924)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:21] ExecIf("PJSIP/2210-000002a3", "0?Set(CALLERID(all)=)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:22] ExecIf("PJSIP/2210-000002a3", "1?Set(CALLERID(all)=<7155989924>)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:23] ExecIf("PJSIP/2210-000002a3", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:24] ExecIf("PJSIP/2210-000002a3", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:25] Set("PJSIP/2210-000002a3", "CDR(outbound_cnum)=7155989924") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-outbound-callerid:26] Set("PJSIP/2210-000002a3", "CDR(outbound_cnam)=") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:15] GosubIf("PJSIP/2210-000002a3", "0?sub-flp-2,s,1()") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:16] Set("PJSIP/2210-000002a3", "OUTNUM=1715828XXXX") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:17] Set("PJSIP/2210-000002a3", "custom=PJSIP") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:18] ExecIf("PJSIP/2210-000002a3", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:19] ExecIf("PJSIP/2210-000002a3", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:20] Macro("PJSIP/2210-000002a3", "dialout-trunk-predial-hook,") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("PJSIP/2210-000002a3", "") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:21] GotoIf("PJSIP/2210-000002a3", "0?skipcrm") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:22] Set("PJSIP/2210-000002a3", "__CRM_DIRECTION=OUTBOUND") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:23] Set("PJSIP/2210-000002a3", "__CRM_DESTINATION=1715828XXXX") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:24] Set("PJSIP/2210-000002a3", "__CRM_SOURCE=2210") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:25] AGI("PJSIP/2210-000002a3", "sangomacrm.agi") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] res_agi.c: sangomacrm.agi: LINKEDID: 1546013962.759
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] res_agi.c: sangomacrm.agi: SOURCE: 2210
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] res_agi.c: sangomacrm.agi: DESTINATION: 1715828XXXX
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] res_agi.c: sangomacrm.agi: DIRECTION: OUTBOUND
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] res_agi.c: sangomacrm.agi: EXTTOCALL:
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] res_agi.c: sangomacrm.agi: START
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] res_agi.c: sangomacrm.agi: SCRIPT: php /var/www/html/admin/modules/sangomacrm/importOne.php 'eyJ1dWlkIjoiMTU0NjAxMzk2Mi43NTkiLCJzb3VyY2UiOiIyMjEwIiwiZGVzdGluYXRpb24iOiIxNzE1ODI4MTAxMCIsImRpcmVjdGlvbiI6Ik9VVEJPVU5EIiwidHlwZSI6IlNUQVJUIiwienVsdV9yYXdfdHlwZSI6IiIsInp1bHVfdHlwZSI6IiIsInp1bHVfdXJsIjoiIiwiZXh0dG9jYWxsIjoiIiwiY251bSI6IjcxNTU5ODk5MjQiLCJjbmFtIjoiIiwiY2FsbHBvcCI6ZmFsc2UsInZvaWNlbWFpbCI6IiJ9' > /dev/null 2>&1 &
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] res_agi.c: <PJSIP/2210-000002a3>AGI Script sangomacrm.agi completed, returning 0
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:26] Set("PJSIP/2210-000002a3", "CHANNEL(hangup_handler_push)=crm-hangup,s,1") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:27] NoOp("PJSIP/2210-000002a3", "CRM Finished") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:28] GotoIf("PJSIP/2210-000002a3", "0?bypass,1") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:29] ExecIf("PJSIP/2210-000002a3", "1?Set(CONNECTEDLINE(num,i)=1715828XXXX)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:30] ExecIf("PJSIP/2210-000002a3", "1?Set(CONNECTEDLINE(name,i)=CID:7155989924)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:31] ExecIf("PJSIP/2210-000002a3", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)7155989924)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:32] GotoIf("PJSIP/2210-000002a3", "0?customtrunk") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-dialout-trunk:33] Dial("PJSIP/2210-000002a3", "PJSIP/1715828XXXX@Alcazar_Networks_outbound,300,Tb(func-apply-sipheaders^s^1,(2))") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] app_stack.c: PJSIP/Alcazar_Networks_outbound-000002a4 Internal Gosub(func-apply-sipheaders,s,1(2)) start
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@func-apply-sipheaders:1] ExecIf("PJSIP/Alcazar_Networks_outbound-000002a4", "0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@func-apply-sipheaders:2] UserEvent("PJSIP/Alcazar_Networks_outbound-000002a4", "zulu-call-b,type:func-apply-sipheaders,to:,from:2210") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@func-apply-sipheaders:3] NoOp("PJSIP/Alcazar_Networks_outbound-000002a4", "Applying SIP Headers to channel 2") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@func-apply-sipheaders:4] Set("PJSIP/Alcazar_Networks_outbound-000002a4", "SIPHEADERKEYS=") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@func-apply-sipheaders:5] ExecIf("PJSIP/Alcazar_Networks_outbound-000002a4", "0?Set(Rheader=1)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@func-apply-sipheaders:6] While("PJSIP/Alcazar_Networks_outbound-000002a4", "0") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] app_while.c: Jumping to priority 10
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@func-apply-sipheaders:11] ExecIf("PJSIP/Alcazar_Networks_outbound-000002a4", "0?SIPRemoveHeader(Alert-Info:)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@func-apply-sipheaders:12] ExecIf("PJSIP/Alcazar_Networks_outbound-000002a4", "0?Set(PJSIP_HEADER(remove,Alert-Info)=)") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] pbx.c: Executing [s@func-apply-sipheaders:13] Return("PJSIP/Alcazar_Networks_outbound-000002a4", "") in new stack
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] app_stack.c: Spawn extension (outbound, 828XXXX, 1) exited non-zero on 'PJSIP/Alcazar_Networks_outbound-000002a4'
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] app_stack.c: PJSIP/Alcazar_Networks_outbound-000002a4 Internal Gosub(func-apply-sipheaders,s,1(2)) complete GOSUB_RETVAL=
[2018-12-28 16:19:22] VERBOSE[29830][C-00000110] app_dial.c: Called PJSIP/1715828XXXX@Alcazar_Networks_outbound
[2018-12-28 16:19:25] VERBOSE[29830][C-00000110] app_dial.c: PJSIP/Alcazar_Networks_outbound-000002a4 is making progress passing it to PJSIP/2210-000002a3
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] app_macro.c: Spawn extension (macro-dialout-trunk, s, 33) exited non-zero on 'PJSIP/2210-000002a3' in macro 'dialout-trunk'
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] pbx.c: Spawn extension (from-internal, 828XXXX, 8) exited non-zero on 'PJSIP/2210-000002a3'
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] pbx.c: Executing [h@from-internal:1] Macro("PJSIP/2210-000002a3", "hangupcall") in new stack
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("PJSIP/2210-000002a3", "1?theend") in new stack
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("PJSIP/2210-000002a3", "0?Set(CDR(recordingfile)=)") in new stack
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-hangupcall:4] NoOp("PJSIP/2210-000002a3", "PJSIP/Alcazar_Networks_outbound-000002a4 monior file= /var/spool/asterisk/monitor/2018/12/28/out-828XXXX-2210-20181228-161922-1546013962.759.WAV") in new stack
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-hangupcall:5] AGI("PJSIP/2210-000002a3", "attendedtransfer-rec-restart.php,PJSIP/Alcazar_Networks_outbound-000002a4,/var/spool/asterisk/monitor/2018/12/28/out-828XXXX-2210-20181228-161922-1546013962.759.WAV") in new stack
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] res_agi.c: <PJSIP/2210-000002a3>AGI Script attendedtransfer-rec-restart.php completed, returning 0
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] pbx.c: Executing [s@macro-hangupcall:6] Hangup("PJSIP/2210-000002a3", "") in new stack
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'PJSIP/2210-000002a3' in macro 'hangupcall'
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/2210-000002a3'
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] app_stack.c: PJSIP/2210-000002a3 Internal Gosub(crm-hangup,s,1) start
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] pbx.c: Executing [s@crm-hangup:1] NoOp("PJSIP/2210-000002a3", "Sending Hangup to CRM") in new stack
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] pbx.c: Executing [s@crm-hangup:2] NoOp("PJSIP/2210-000002a3", "HANGUP CAUSE: 127") in new stack
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] pbx.c: Executing [s@crm-hangup:3] ExecIf("PJSIP/2210-000002a3", "0?Set(__CRM_VOICEMAIL=)") in new stack
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] pbx.c: Executing [s@crm-hangup:4] NoOp("PJSIP/2210-000002a3", "MASTER CHANNEL: 1546013962.759 = 1546013962.759") in new stack
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] pbx.c: Executing [s@crm-hangup:5] GotoIf("PJSIP/2210-000002a3", "0?return") in new stack
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] pbx.c: Executing [s@crm-hangup:6] Set("PJSIP/2210-000002a3", "__CRM_HANGUP=1") in new stack
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] pbx.c: Executing [s@crm-hangup:7] AGI("PJSIP/2210-000002a3", "sangomacrm.agi") in new stack
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] res_agi.c: sangomacrm.agi: LINKEDID: 1546013962.759
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] res_agi.c: sangomacrm.agi: SOURCE: 2210
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] res_agi.c: sangomacrm.agi: DESTINATION: 1715828XXXX
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] res_agi.c: sangomacrm.agi: DIRECTION: OUTBOUND
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] res_agi.c: sangomacrm.agi: EXTTOCALL:
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] res_agi.c: sangomacrm.agi: START
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] res_agi.c: sangomacrm.agi: SCRIPT: php /var/www/html/admin/modules/sangomacrm/importOne.php 'eyJ1dWlkIjoiMTU0NjAxMzk2Mi43NTkiLCJzb3VyY2UiOiIyMjEwIiwiZGVzdGluYXRpb24iOiIxNzE1ODI4MTAxMCIsImRpcmVjdGlvbiI6Ik9VVEJPVU5EIiwidHlwZSI6IkVORCIsInp1bHVfcmF3X3R5cGUiOiIiLCJ6dWx1X3R5cGUiOiIiLCJ6dWx1X3VybCI6IiIsImV4dHRvY2FsbCI6IiIsImNudW0iOiI3MTU1OTg5OTI0IiwiY25hbSI6IiIsImNhbGxwb3AiOmZhbHNlLCJ2b2ljZW1haWwiOiIifQ==' > /dev/null 2>&1 &
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] res_agi.c: <PJSIP/2210-000002a3>AGI Script sangomacrm.agi completed, returning 0
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] pbx.c: Executing [s@crm-hangup:8] Return("PJSIP/2210-000002a3", "") in new stack
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/2210-000002a3'
[2018-12-28 16:19:41] VERBOSE[29830][C-00000110] app_stack.c: PJSIP/2210-000002a3 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
[2018-12-28 16:19:41] VERBOSE[29831][C-00000110] app_mixmonitor.c: MixMonitor close filestream (mixed)
[2018-12-28 16:19:41] VERBOSE[29831][C-00000110] app_mixmonitor.c: End MixMonitor Recording PJSIP/2210-000002a3
[2018-12-28 16:19:47] WARNING[16445] res_pjsip_registrar.c: Endpoint 'anonymous' has no configured AORs

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@gregorywest wrote:

Here is a different and more pressing issue. One my PBX which is on a WAY overpowered VM Server using only local HDD for storage I am getting a fair amount of audio breakups on both internal and external calls.

Two questions, is this more likely an issue with the VM Server, or more likely a QoS problem on the 1GB network.

If it is a QoS problem, are there any good easy to understand QoS documents to explain how to implement. I do know how to use vLans if that is the best way to solve the problem.

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@DarkQuark wrote:

Greetings, I am attempting to do something I am not sure is possible. My goal is to use FreePBX as an autodialer for notifications coming from another platform.(not for marketing robocalls)

My thought process is to utilize call files. I have done this successfully to play prerecorded sound files. While I had issues with VM detection that seemed to work for the most part.

However what I really need is to somehow utilize TTS to read out text placed in the call file (places there programmatically by another platform).

I cannot find any syntax to do this but the info out there for call files seems scant.

Any help or suggestions (in any direction) is appreciated. Thank you.

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@dannyprecise wrote:

A user hung up a call and can make outbound calls and get calls from other extensions.

However it’s showing they are still on the outbound call they made 2 hours ago and hung up on.

Since it thinks they are still on a call, they cannot take any calls from the queue they are associated with

Is there aa way to clear the users call somehow to force it to update that they are off the call?

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@ColoradoRuss wrote:

I am posting this because this issue drove me nuts for days and I finally solved it. Hopefully this post will help someone else…

I have a couple hundred pjsip extensions. There were no obvious issues with the phone config nor the extensions, but I kept none of the extensions could complete a call, even from extension to extension.

I disconnected all phones but one and ran wireshark on the ip of the phone: “tshark host 192.168.1.x”

I was repeatedly getting 401 Unauthorized replies to all of my attempted REGISTER requests.

Ran debug on asterisk:
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
CLI> core set verbose 5
CLI> core set debug 5
CLI> module reload logger

My phones were trying to register two test extensions that I set up, 11 and 12. I saw in the matrix running across my screen that it was referencing a different extension, call it 99. I saw references to “99-identify” - clearly WRONG!

Some to find out, asterisk tries to identify an endpoint in an order that causes it to miscategorize my phones extensions.
https://blogs.asterisk.org/2018/02/07/identifying-endpoint-pjsip/

Solution was to enter the following line in my pjsip_custom_post.conf file and restarting
https://issues.freepbx.org/browse/FREEPBX-17803
endpoint_identifier_order=auth_username,username,ip

The default identify order is apparently not what I was expecting.
CLI> pjsip show identifiers
Identifier Names:
name not specified
ip
username
anonymous
header
auth_username

Im sure some of the others may be able to add some suggestions to this, but I hope this helps someone.

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@cometa_s wrote:

**Hello, i want to record about 10 different IVR recordings and have each play in a random for every incoming call.

Any idea? Easy way

thanks.**

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@jessy5765 wrote:

We just rolled out Zulu to one of my clients and everything is working very well. The SSL certificate is installed and voice is working now that ICE settings are in there, however…

Every couple days they are calling me and saying when they try to sign in its just showing them scroll bars and they never connect on the Zulu desktop software

I go in and see that a few users have 10-20 connections to their extension, even though i try to clear it, they never clear and i have to restart the system to clear them. Upon restart everything works just fine and they connect immediately

Any Ideas?.

December 10th 2018, 11:23:03 am - debug: AccountManager - constructor
December 10th 2018, 11:23:03 am - info: Using the WebAudio API to generate sounds
December 10th 2018, 11:23:03 am - debug: avatars - type - @@redux/INIT
December 10th 2018, 11:23:03 am - debug: avatars - type - @@redux/PROBE_UNKNOWN_ACTION_c.5.8.7.0.r
December 10th 2018, 11:23:04 am - debug: avatars - type - @@redux/INIT
December 10th 2018, 11:23:04 am - debug: Containers - LoginControl - needsLogin: true
December 10th 2018, 11:23:04 am - debug: avatars - type - persist/REHYDRATE
December 10th 2018, 11:23:04 am - debug: Reducers - config - persist/REHYDRATE
December 10th 2018, 11:23:04 am - debug: Containers - LoginControl - needsLogin: true
December 10th 2018, 11:23:04 am - debug: Reducers - app - AUTH_LOGIN_REQUEST
December 10th 2018, 11:23:04 am - debug: avatars - type - AUTH_LOGIN_REQUEST
December 10th 2018, 11:23:04 am - debug: WebSocketConnection - constructor - 0
December 10th 2018, 11:23:04 am - debug: PBXAccount - constructor - 43089783-a299-45b3-8037-bb7ad1528b08 - 0
December 10th 2018, 11:23:04 am - debug: AccountManager - add - 43089783-a299-45b3-8037-bb7ad1528b08
December 10th 2018, 11:23:04 am - debug: WebSocketConnection - connect
December 10th 2018, 11:23:04 am - debug: WebSocketConnection - connect - Connecting to - wss://pbx.msihcare.com:8002/
December 10th 2018, 11:23:04 am - debug: PBXAccount - _onConnecting - 43089783-a299-45b3-8037-bb7ad1528b08
December 10th 2018, 11:23:04 am - debug: Actions - pbxaccount - pbxAccountConnecting - 43089783-a299-45b3-8037-bb7ad1528b08
December 10th 2018, 11:23:04 am - debug: Reducers - auth - PBXACCOUNT_CONNECTING
December 10th 2018, 11:23:04 am - debug: avatars - type - PBXACCOUNT_CONNECTING
December 10th 2018, 11:23:05 am - debug: WebSocketConnection - connect - onopen
December 10th 2018, 11:23:05 am - debug: PBXAccount - _onOpened - 43089783-a299-45b3-8037-bb7ad1528b08
December 10th 2018, 11:23:05 am - debug: Actions - pbxaccount - pbxAccountOpen - 43089783-a299-45b3-8037-bb7ad1528b08
December 10th 2018, 11:23:05 am - debug: avatars - type - PBXACCOUNT_OPEN
December 10th 2018, 11:23:05 am - debug: WebSocketConnection - login - clientversion: 3.0.0 - clienttype: electron-windows
December 10th 2018, 11:23:05 am - debug: PBXAccount - _onAuthenticating - 43089783-a299-45b3-8037-bb7ad1528b08
December 10th 2018, 11:23:05 am - debug: Actions - pbxaccount - pbxAccountAuthenticating - 43089783-a299-45b3-8037-bb7ad1528b08
December 10th 2018, 11:23:05 am - debug: avatars - type - PBXACCOUNT_AUTHENTICATING
December 10th 2018, 11:23:07 am - debug: cancelLogin - 43089783-a299-45b3-8037-bb7ad1528b08
December 10th 2018, 11:23:07 am - debug: AccountManager - get - 43089783-a299-45b3-8037-bb7ad1528b08
December 10th 2018, 11:23:07 am - debug: PBXAccount - close - Removing listeners… - 43089783-a299-45b3-8037-bb7ad1528b08
December 10th 2018, 11:23:07 am - debug: WebSocketConnection - close
December 10th 2018, 11:23:07 am - debug: Reducers - app - AUTH_LOGIN_CANCEL
December 10th 2018, 11:23:07 am - debug: avatars - type - AUTH_LOGIN_CANCEL
December 10th 2018, 11:23:18 am - debug: Containers - Login - doLogin - 4099 - pbx.clientdomain.com - - true
December 10th 2018, 11:23:18 am - debug: Actions - auth - login - 4099 - pbx.clientdomain.com - 8002
December 10th 2018, 11:23:19 am - debug: Reducers - app - AUTH_LOGIN_REQUEST
December 10th 2018, 11:23:19 am - debug: avatars - type - AUTH_LOGIN_REQUEST
December 10th 2018, 11:23:19 am - debug: WebSocketConnection - constructor - 0
December 10th 2018, 11:23:19 am - debug: PBXAccount - constructor - ceaf3edf-c204-42b2-a21b-ae483646a537 - 0
December 10th 2018, 11:23:19 am - debug: AccountManager - add - ceaf3edf-c204-42b2-a21b-ae483646a537
December 10th 2018, 11:23:19 am - debug: WebSocketConnection - connect
December 10th 2018, 11:23:19 am - debug: WebSocketConnection - connect - Connecting to - wss://pbx.msihcare.com:8002/
December 10th 2018, 11:23:19 am - debug: PBXAccount - _onConnecting - ceaf3edf-c204-42b2-a21b-ae483646a537
December 10th 2018, 11:23:19 am - debug: Actions - pbxaccount - pbxAccountConnecting - ceaf3edf-c204-42b2-a21b-ae483646a537
December 10th 2018, 11:23:19 am - debug: Reducers - auth - PBXACCOUNT_CONNECTING
December 10th 2018, 11:23:19 am - debug: avatars - type - PBXACCOUNT_CONNECTING
December 10th 2018, 11:23:19 am - debug: WebSocketConnection - connect - onopen
December 10th 2018, 11:23:19 am - debug: PBXAccount - _onOpened - ceaf3edf-c204-42b2-a21b-ae483646a537
December 10th 2018, 11:23:19 am - debug: Actions - pbxaccount - pbxAccountOpen - ceaf3edf-c204-42b2-a21b-ae483646a537
December 10th 2018, 11:23:19 am - debug: avatars - type - PBXACCOUNT_OPEN
December 10th 2018, 11:23:19 am - debug: WebSocketConnection - login - clientversion: 3.0.0 - clienttype: electron-windows
December 10th 2018, 11:23:19 am - debug: PBXAccount - _onAuthenticating - ceaf3edf-c204-42b2-a21b-ae483646a537
December 10th 2018, 11:23:19 am - debug: Actions - pbxaccount - pbxAccountAuthenticating - ceaf3edf-c204-42b2-a21b-ae483646a537
December 10th 2018, 11:23:19 am - debug: avatars - type - PBXACCOUNT_AUTHENTICATING

image.png1264x714 64.7 KB

These are the logs and this is the screen that just sits there scrolling. Even after reboot its happening now.

image.png1344x262 24 KB

I can SSH into the phone system, run fwconsole restart zulu and then my computer immediate gets an error message that the server is not available (after sitting in that logging in state for 30 minutes)

Once zulu starts back up i can click on connect in the Zulu Desktop app and i am in. I also tested and the phone app works just fine as well. What can keep locking the Zulu server up? Once your in its fine and works with no issues. Its like authentication is causing an issue

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@gilbernjm wrote:

hi all ,
i have a panasonic central connected to digium card , i have allowed the users on the central to make call with pstn extensions using ivr , but now i want to make the reverse , i want to allow sip client to make call with extension on the central , any idea please .

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@nickds1 wrote:

Hi - please forgive the intrusion is this is not an appropriate place for this question…

We have several buildings, linked by WiFi (Ubiquity bridges) on a single subnet. Currently, we have BT Broadband for … well… broadband (ADSL2+), and a single POTS line (also BT - bundled with the broadband) to a master box, on which there is a Panasonic KX DECT system - this is all to the main house. We are in “a rural situation” in the UK.

I’d like to extend the DECT (or a similar system) to the other buildings (my workshops etc.) - it has been suggested that one option might be to run FreePBX on an RPi and to use VoIP, but I’m way out of my comfort zone at this point. We need to extend the voice system as mobiles don’t work very well here (yes, there are still “dead zones” in the countryside).

Being an engineer, simple is good. The other occupants of the property are not technical, so… simple is good. They understand DECT - If there is a simple way to extend that over an IP link, that’d be great.

Before anyone suggests just using DECT repeaters, we’ve tried that - the distances are too large and there are annoying things like trees in the way…

I’d really appreciate thoughts on this - is going VoIP OTT? Is there a simpler solution? I do understand networking to a reasonable level (happy setting up Cisco switches, Ubiquity kit & firewalls etc.)

Many thanks

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@mmagary wrote:

All calls going to one extension. I have 4 port digium FXO card all ports setup on different groups 0,1,2,3. Believe issue to be DID as outbound calls go out proper ports. Is anyone familiar with DID for routing through the inbound. We don’t currently believe the POTS provider is pushing DID.

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@aluatis wrote:

Hello,

I am new to FreePBX. I have tried searching the forums but cannot seem to get the right settings set on my system.

I have a trunk with Twilio for outbound service. Dialing NANP numbers right now works fine. Dialing international numbers is giving the following message in the logs following by “The number you have dialed is not in service.”

-- Executing [1@macro-dialout-trunk:1] Goto("SIP/17976-00000052", "s-INVALIDNMBR,1") in new stack

First, I am in US so the format I am dialing is 011+country code+area code+number. For this test, I am using the Hilton London number of +442074938000. It is being dialed on my handset as the following 011442074938000.
Second, dialing this number on a standard POTS line dials fine using the above number as I wrote it.
Third, My Outbound Routes are as follows:

prepend,prefix,“match pattern”,callerid
,011.,
,1800NXXXXXX,
,1833NXXXXXX,
,1844NXXXXXX,
,1855NXXXXXX,
,1866NXXXXXX,
,1877NXXXXXX,
,1888NXXXXXX,
,1NXXNXXXXXX,
1,NXXNXXXXXX,

Lastly, Twilio requires ALL dialed calls to be in E.164 format. This was fairly easy to accomplish by adding a + in the Outbound Dial Prefix field under Dialed Number Manipulation Field.
This is where I think I have a problem. No matter what I enter for trying to rewrite the 011 rule, it always sends the call to Twilio as follows:

== Spawn extension (from-trunk-sip-TwilioTrunk, 011442074938000, 1) exited non-zero on ‘SIP/TwilioTrunk-0000005b’
– SIP/TwilioTrunk-0000005b Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
– Called SIP/TwilioTrunk/+011442074938000

I have confirmed this by going thru the “Debugger” and the logs in my account. I have added a rule for 011. putting 011 in the prefix, I have tried 011X., 011N., 011[0-9]. all of which don’t drop the 011. Is there something special about the 011 that I haven’t figured out yet? I can’t seem to figure out why some rules appear to work and others don’t.
If I add a rule to match “.” and prepend + and remove + from the Outbound Dial Prefix, it works. But if I add a rule that matches 1NXXNXXXX and prepends + it doesn’t. If I have something set in Outbound Route and I not allowed to modify it under the Trunk?

I’ve beat my ahead against the wall and was hoping someone could offer some assistance.

–
Edit: I uploaded a screenshot of my Dial Manipulation Rules. The ones for 7 and 10 digit work, the one for 011 does not.

Screenshot from 2018-12-29 20-46-59.png884x339 19.4 KB

Edit 2: Upload showing log from Twilio showing the number they received has 011 and is not formatted correctly. i (dot) imgur (dot) com/7oimK04.png (you’ll have to copy/paste)

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@comtech wrote:

FreePBX/Asterisk 14

Objective:
I would like the ability to pause an agent without having to put the current caller on hold.

Problem:
You can do this with the phone apps on Sangoma phones, but once I select pause for the agent, the active queue call that agent is on will drop a few seconds later.

Method:
I am making a new db keypair that will show each extension as 1 or null. If the extension is 1, that means they are pending needing a pause. If null, no special action (pause) is needed.

Question:
Is there an on-hook or after queue call macro or something that I can plug my dialplan into? Basically once the call hangs up, it would trigger to look at the DB table. If the DB result was 1, pause the agent, if not, follow the normal workflow.

Thanks all!

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@ColoradoRuss wrote:

Can someone explain to me the usage of the callerid field in bulk extensions? It doesnt seem to have a field in the gui.

The callerid, cid_masquerade, name, and outboundcid fields all seem to play together. Is there somewhere I can read precisely what they all do. I cant seem to figure out what the callerid and cid_masquerade do in relation to each other.

I set the callerid to “my name” <100> and cid_masquerade to 100. I want it to use internal “my name” <100> but it pulls the name from cid_masquerade. If I remove cid_masquerade, it uses the details of the actual extension. It seems callerid doesnt do anything.

Also, where is cid_masquerade in the pjsip setup? I cant find it anywhere.

Thanks and Happy New Year!

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@LTelephony wrote:

Hello,

since a few weeks we are having some Problems.

Here are some informations:

Asterisk 13.22.0
FreePBX 12.7.5-1807-1.sgn7
Phones: only Snom Phones

Situation:
We need to type a 0 as prefix before dailing out (0 0123456879)

Problem.
On the most calls that works perfect.
But sometimes when we dail 00123456789 it stands in the display an when we press “call” it switches to 0123456789 and we can’t dail out

I hope someone can help me
Thanks!

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@smokey7722 wrote:

I’ve been an affiliate partner with Vitelity for years, selling to various clients of mine (some with 1xDID, some with a few dozen). I was getting ready to onboard another new client and it seems Vitelity with their new corporate overlords, decided they no longer want to do any retail sales. Ok, thats fine, I’ll call to setup the account. Well the rep I spoke with, who also told me the affiliate program is meaningless now said he won’t setup any accounts with less than a few hundred, yes hundred, DID’s. Guess they want the commercial wholesale market only?

So given that shocking news - I need to find a new SIP Provider that offers competitive rates (unlimited isn’t required though would be welcome for some clients) and T.38 support. I’ve been reading that sipstation T.38 doesn’t work anymore? And I briefly just looked at voip.ms as a potential option as well but figured it couldn’t hurt to ask the guru’s here if they had any recommendations. The last thread I saw discussing this was a few years ago and there have certainly been some changes in the market. An affiliate program would be nice but I can live without that if thats not an option, especially since most of my clients are fairly small so theres not usually much if any kickback on the sales.

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@ewsclass66 wrote:

Hi, I am using FreePBX 14.0.5.2 installed on Ubuntu 16.04 and everything is working perfectly except from the Sound Languages module. I have downloaded and installed the en_GB module however all of the voices from the echo test to the voicemail is still in the American accent. I have tried replacing the /var/lib/asterisk/en directory with the files copied from the en_GB directory and that hasn’t resolved my issue.
When asterisk -rvvvvvvv is run and I call the voice mail, I notice this:
<PJSIP/7066-00000002> Playing ‘vm-password.slin’ (language ‘en_GB’)
It appears to be using the correct language, however I am still hearing the US accent.
thanks for any help in advance

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@rohitg76 wrote:

I am puzzled by this one. On one installation of FreePBX/Asterisk 13, in the Asterisk Info Peers report, I notice that all remote SIP peers are indicating Host as the PBX server IP address instead of reflecting their WAN IP source, from where they are connected (refer enclosed picture). The registrations and calls are working perfectly. However, since every remote connection is presenting PBX IP, Fail2Ban is going for a toss - as every hack attempt is also reflecting with this IP hence, all remote users get banned enmasse. Adding the PBX IP to whitelist will make the whole IP security futile. I am curious to find why is this happening in the first place. Have compared config-to-config line by line for all config files esp pertaining to NAT, but cannot make out the cause for this. Is it possible the Router behind which the PBX currently sits may be presenting this IP instead of the source IP?

Appreciate some guidance. Thanks.

image.png1194x446 111 KB

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@GGroce63 wrote:

I’m setting up a FreePBX server to be used as a Fax Server. We are also the Service provider for this application and are serving the FreePBX with a SIP trunk from our Meta Switch.

The Trunk is registered and I am able to receive inbound calls to the FreePBX but unable to process any outbound calls. The call does not appear to be leaving the PBX as I do not see any call record or SIP message activity on the Metaswitch.

Outbound routing appears very easy but I am apparently missing something.

Any input I can check would be appreciated/

Thanks

Gene

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Tango Frog added a comment - 5 minutes ago

Thanks for creating a report! The issue has entered the triage (open) process. That means the issue will wait in this status until the FreePBX team has an opportunity to review the issue. Our weekly triage meetings happen every Monday. Once the issue has been reviewed you may receive comments regarding the next steps towards resolution.

A good first step is for you to review the FreePBX Issue Guidelines if you haven’t already. The guidelines detail what is expected from an FreePBX issue report.

Furthermore, our records indicate that you have not filled out our CLA so if you are submitting a patch, please make sure you have reviewed and submitted a Code Submission Agreement. We will not be able to accept any patches unless you have previously filled this out.

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@jptlos wrote:

We have a ring group 101 and the “Destination if no answer” was ring group 102.
The “Destination if no answer” on each Extension of ring group 101 member was also ring group 102.

May I know what will be the priority of the system, the Extension “Destination if no answer” or Ring Groups “Destination if no answer”? Thanks.

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@person3375 wrote:

Hi,

Is there a way to change the default recording file permissions that FreePBX uses when writing the files?

I have a process that copies the files off the system then deletes the originals to save on storage space. It has it’s own user account that is part of the asterisk group. Now after doing some module updates and rebooting the system, the umask I set in the past is no longer applying to the new recording files. The issue is that new files do not have group write permissions.

This system is currently running the FreePBX distro 13.0.195.22 with the latest module updates.

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