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Unable to make outgoing SIP calls. Inbound works

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@BigJ wrote:

Hello.

Thanks in advance. I just rolled out a PBX setup in my office. This is my first venture in this space (PBX), so I’m a newb. My problem is that I am able to receive inbound calls, but when making outbound calls, I get the “all circuits busy” message.

Background: I set up two Cisco SPA 525 phones in my office and have created extensions for maybe 6 users. I’m on VoicePulse and that as working for I/B and O/B calls. The problem here was that Verizon would change my IP every 24 hours and I’d lose registration until I performed a manual reset. Easiest thing was to get a static IP w/ Verizon, which I did.

Adjusting the settings after that change is where the outbound calls broke, but I can’t figure out why. I get that there’s a “congestion” error, but none of the phones are being used at the time. We’re not making 5 outgoing calls at once. Of note I also added a third phone to the system, same type as the others.

I’ll paste an outgoing call. Phone numbers/ IP’s removed.

PBX Firmware: 12.7.6-1904-1.sng7

freepbxCLI>
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [TestCallOut@from-internal:1] Macro(“SIP/103-0000006a”, “user-callerid,LIMIT”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/103-0000006a”, “TOUCH_MONITOR=1561752560.109”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/103-0000006a”, “AMPUSER=103”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/103-0000006a”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“SIP/103-0000006a”, “1?Set(REALCALLERIDNUM=103)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/103-0000006a”, “AMPUSER=103”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/103-0000006a”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/103-0000006a”, “AMPUSERCIDNAME=FD3”) in new stack
– Executing [s@macro-user-callerid:8] ExecIf(“SIP/103-0000006a”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
– Executing [s@macro-user-callerid:9] GotoIf(“SIP/103-0000006a”, “0?report”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/103-0000006a”, “AMPUSERCID=103”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/103-0000006a”, “__DIAL_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-user-callerid:12] Set(“SIP/103-0000006a”, “CALLERID(all)=“FD3” <103>”) in new stack
– Executing [s@macro-user-callerid:13] GotoIf(“SIP/103-0000006a”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:14] ExecIf(“SIP/103-0000006a”, “1?Set(GROUP(concurrency_limit)=103)”) in new stack
– Executing [s@macro-user-callerid:15] ExecIf(“SIP/103-0000006a”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:16] NoOp(“SIP/103-0000006a”, “Macro Depth is 1”) in new stack
– Executing [s@macro-user-callerid:17] GotoIf(“SIP/103-0000006a”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,18)
– Executing [s@macro-user-callerid:18] GotoIf(“SIP/103-0000006a”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,37)
– Executing [s@macro-user-callerid:37] Set(“SIP/103-0000006a”, “CALLERID(number)=103”) in new stack
– Executing [s@macro-user-callerid:38] Set(“SIP/103-0000006a”, “CALLERID(name)=FD3”) in new stack
– Executing [s@macro-user-callerid:39] GotoIf(“SIP/103-0000006a”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:40] Set(“SIP/103-0000006a”, “CDR(cnam)=FD3”) in new stack
– Executing [s@macro-user-callerid:41] Set(“SIP/103-0000006a”, “CDR(cnum)=103”) in new stack
– Executing [s@macro-user-callerid:42] Set(“SIP/103-0000006a”, “CHANNEL(language)=en”) in new stack
– Executing [TestCallOut@from-internal:2] Set(“SIP/103-0000006a”, “ROUTEUSER=103”) in new stack
– Executing [TestCallOut@from-internal:3] Set(“SIP/103-0000006a”, “ROUTEUSER=103”) in new stack
– Executing [TestCallOut@from-internal:4] GotoIf(“SIP/103-0000006a”, “1?notblind”) in new stack
– Goto (from-internal,TestCallOut,7)
– Executing [TestCallOut@from-internal:7] GotoIf(“SIP/103-0000006a”, “1?restrictedroute-c4ca4238a0b923820dcc509a6f75849b,TestCallOut,2:outbound-allroutes,TestCallOut,2”) in new stack
– Goto (restrictedroute-c4ca4238a0b923820dcc509a6f75849b,TestCallOut,2)
– Executing [TestCallOut@restrictedroute-c4ca4238a0b923820dcc509a6f75849b:2] Gosub(“SIP/103-0000006a”, “sub-record-check,s,1(out,TestCallOut,dontcare)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“SIP/103-0000006a”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“SIP/103-0000006a”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“SIP/103-0000006a”, “NOW=1561752560”) in new stack
– Executing [s@sub-record-check:4] Set(“SIP/103-0000006a”, “__DAY=28”) in new stack
– Executing [s@sub-record-check:5] Set(“SIP/103-0000006a”, “__MONTH=06”) in new stack
– Executing [s@sub-record-check:6] Set(“SIP/103-0000006a”, “__YEAR=2019”) in new stack
– Executing [s@sub-record-check:7] Set(“SIP/103-0000006a”, “__TIMESTR=20190628-160920”) in new stack
– Executing [s@sub-record-check:8] Set(“SIP/103-0000006a”, “__FROMEXTEN=103”) in new stack
– Executing [s@sub-record-check:9] Set(“SIP/103-0000006a”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“SIP/103-0000006a”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“SIP/103-0000006a”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“SIP/103-0000006a”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“SIP/103-0000006a”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“SIP/103-0000006a”, “3?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“SIP/103-0000006a”, “1?sub-record-check,out,1”) in new stack
– Goto (sub-record-check,out,1)
– Executing [out@sub-record-check:1] NoOp(“SIP/103-0000006a”, “Outbound Recording Check from 103 to TestCallOut”) in new stack
– Executing [out@sub-record-check:2] Set(“SIP/103-0000006a”, “RECMODE=dontcare”) in new stack
– Executing [out@sub-record-check:3] ExecIf(“SIP/103-0000006a”, “1?Goto(routewins)”) in new stack
– Goto (sub-record-check,out,7)
– Executing [out@sub-record-check:7] Gosub(“SIP/103-0000006a”, “recordcheck,1(dontcare,out,TestCallOut)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“SIP/103-0000006a”, “Starting recording check against dontcare”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“SIP/103-0000006a”, “dontcare”) in new stack
– Goto (sub-record-check,recordcheck,3)
– Executing [recordcheck@sub-record-check:3] Return(“SIP/103-0000006a”, “”) in new stack
– Executing [out@sub-record-check:8] Return(“SIP/103-0000006a”, “”) in new stack
– Executing [TestCallOut@restrictedroute-c4ca4238a0b923820dcc509a6f75849b:3] ExecIf(“SIP/103-0000006a”, “0 ?Set(CDR(accountcode)=)”) in new stack
– Executing [TestCallOut@restrictedroute-c4ca4238a0b923820dcc509a6f75849b:4] Set(“SIP/103-0000006a”, “MOHCLASS=default”) in new stack
– Executing [TestCallOut@restrictedroute-c4ca4238a0b923820dcc509a6f75849b:5] ExecIf(“SIP/103-0000006a”, “1?Set(TRUNKCIDOVERRIDE=VoicePulseNumber)”) in new stack
– Executing [TestCallOut@restrictedroute-c4ca4238a0b923820dcc509a6f75849b:6] Set(“SIP/103-0000006a”, “_NODEST=”) in new stack
– Executing [TestCallOut@restrictedroute-c4ca4238a0b923820dcc509a6f75849b:7] Macro(“SIP/103-0000006a”, “dialout-trunk,1,1TestCallOut,off”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/103-0000006a”, “DIAL_TRUNK=1”) in new stack
– Executing [s@macro-dialout-trunk:2] ExecIf(“SIP/103-0000006a”, “0?Set(DIAL_OPTIONS=Hhtr)”) in new stack
– Executing [s@macro-dialout-trunk:3] GosubIf(“SIP/103-0000006a”, “0?sub-pincheck,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:4] ExecIf(“SIP/103-0000006a”, “0?Set(CALLERID(num)=103)”) in new stack
– Executing [s@macro-dialout-trunk:5] GotoIf(“SIP/103-0000006a”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/103-0000006a”, “DIAL_NUMBER=1TestCallOut”) in new stack
– Executing [s@macro-dialout-trunk:7] Set(“SIP/103-0000006a”, “DIAL_TRUNK_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-dialout-trunk:8] Set(“SIP/103-0000006a”, “OUTBOUND_GROUP=OUT_1”) in new stack
– Executing [s@macro-dialout-trunk:9] Set(“SIP/103-0000006a”, “DIAL_TRUNK_OPTIONS=T”) in new stack
– Executing [s@macro-dialout-trunk:10] GotoIf(“SIP/103-0000006a”, “0?nomax”) in new stack
– Executing [s@macro-dialout-trunk:11] GotoIf(“SIP/103-0000006a”, “0?chanfull”) in new stack
– Executing [s@macro-dialout-trunk:12] GotoIf(“SIP/103-0000006a”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:13] Macro(“SIP/103-0000006a”, “outbound-callerid,1”) in new stack
– Executing [s@macro-outbound-callerid:1] NoOp(“SIP/103-0000006a”, “103”) in new stack
– Executing [s@macro-outbound-callerid:2] NoOp(“SIP/103-0000006a”, “”) in new stack
– Executing [s@macro-outbound-callerid:3] NoOp(“SIP/103-0000006a”, “off”) in new stack
– Executing [s@macro-outbound-callerid:4] ExecIf(“SIP/103-0000006a”, “0?Set(CALLERPRES(name-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:5] ExecIf(“SIP/103-0000006a”, “0?Set(CALLERPRES(num-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:6] ExecIf(“SIP/103-0000006a”, “0?Set(REALCALLERIDNUM=103)”) in new stack
– Executing [s@macro-outbound-callerid:7] ExecIf(“SIP/103-0000006a”, “0?Set(AMPUSER=103)”) in new stack
– Executing [s@macro-outbound-callerid:8] GotoIf(“SIP/103-0000006a”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [s@macro-outbound-callerid:12] Set(“SIP/103-0000006a”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:13] Set(“SIP/103-0000006a”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:14] Set(“SIP/103-0000006a”, “TRUNKOUTCID=VoicePulseNumber”) in new stack
– Executing [s@macro-outbound-callerid:15] GotoIf(“SIP/103-0000006a”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,21)
– Executing [s@macro-outbound-callerid:21] ExecIf(“SIP/103-0000006a”, “1?Set(CALLERID(all)=VoicePulseNumber)”) in new stack
– Executing [s@macro-outbound-callerid:22] ExecIf(“SIP/103-0000006a”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:23] ExecIf(“SIP/103-0000006a”, “1?Set(CALLERID(all)=VoicePulseNumber)”) in new stack
– Executing [s@macro-outbound-callerid:24] ExecIf(“SIP/103-0000006a”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:25] ExecIf(“SIP/103-0000006a”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:26] Set(“SIP/103-0000006a”, “CDR(outbound_cnum)=VoicePulseNumber”) in new stack
– Executing [s@macro-outbound-callerid:27] Set(“SIP/103-0000006a”, “CDR(outbound_cnam)=”) in new stack
– Executing [s@macro-dialout-trunk:14] GosubIf(“SIP/103-0000006a”, “1?sub-flp-1,s,1()”) in new stack
– Executing [s@sub-flp-1:1] ExecIf(“SIP/103-0000006a”, “1?Return()”) in new stack
– Executing [s@macro-dialout-trunk:15] Set(“SIP/103-0000006a”, “OUTNUM=1TestCallOut”) in new stack
– Executing [s@macro-dialout-trunk:16] Set(“SIP/103-0000006a”, “custom=SIP/norman”) in new stack
– Executing [s@macro-dialout-trunk:17] ExecIf(“SIP/103-0000006a”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)”) in new stack
– Executing [s@macro-dialout-trunk:18] ExecIf(“SIP/103-0000006a”, “0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))”) in new stack
– Executing [s@macro-dialout-trunk:19] Macro(“SIP/103-0000006a”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/103-0000006a”, “”) in new stack
– Executing [s@macro-dialout-trunk:20] GotoIf(“SIP/103-0000006a”, “0?skipcrm”) in new stack
– Executing [s@macro-dialout-trunk:21] Set(“SIP/103-0000006a”, “__CRM_DIRECTION=OUTBOUND”) in new stack
– Executing [s@macro-dialout-trunk:22] Set(“SIP/103-0000006a”, “__CRM_DESTINATION=1TestCallOut”) in new stack
– Executing [s@macro-dialout-trunk:23] Set(“SIP/103-0000006a”, “__CRM_SOURCE=103”) in new stack
– Executing [s@macro-dialout-trunk:24] AGI(“SIP/103-0000006a”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <SIP/103-0000006a>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@macro-dialout-trunk:25] Set(“SIP/103-0000006a”, “CHANNEL(hangup_handler_push)=crm-hangup,s,1”) in new stack
– Executing [s@macro-dialout-trunk:26] NoOp(“SIP/103-0000006a”, “CRM Finished”) in new stack
– Executing [s@macro-dialout-trunk:27] GotoIf(“SIP/103-0000006a”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:28] ExecIf(“SIP/103-0000006a”, “1?Set(CONNECTEDLINE(num,i)=1TestCallOut)”) in new stack
– Executing [s@macro-dialout-trunk:29] ExecIf(“SIP/103-0000006a”, “1?Set(CONNECTEDLINE(name,i)=CID:VoicePulseNumber)”) in new stack
– Executing [s@macro-dialout-trunk:30] ExecIf(“SIP/103-0000006a”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)VoicePulseNumber)”) in new stack
– Executing [s@macro-dialout-trunk:31] GotoIf(“SIP/103-0000006a”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:32] Dial(“SIP/103-0000006a”, “SIP/norman/1TestCallOut,300,Tb(func-apply-sipheaders^s^1,(1))”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– SIP/norman-0000006b Internal Gosub(func-apply-sipheaders,s,1(1)) start
– Executing [s@func-apply-sipheaders:1] ExecIf(“SIP/norman-0000006b”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
– Executing [s@func-apply-sipheaders:2] NoOp(“SIP/norman-0000006b”, “Applying SIP Headers to channel SIP/norman-0000006b”) in new stack
– Executing [s@func-apply-sipheaders:3] Set(“SIP/norman-0000006b”, “TECH=SIP”) in new stack
– Executing [s@func-apply-sipheaders:4] Set(“SIP/norman-0000006b”, “SIPHEADERKEYS=”) in new stack
– Executing [s@func-apply-sipheaders:5] While(“SIP/norman-0000006b”, “0”) in new stack
– Jumping to priority 12
– Executing [s@func-apply-sipheaders:13] Return(“SIP/norman-0000006b”, “”) in new stack
== Spawn extension (from-pstn, TestCallOut, 1) exited non-zero on ‘SIP/norman-0000006b’
– SIP/norman-0000006b Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
– Called SIP/norman/1TestCallOut
[2019-06-28 16:09:20] NOTICE[14438][C-0000003f]: chan_sip.c:24042 handle_response_invite: Failed to authenticate on INVITE to ‘sip:VoicePulseNumber@MyStaticIP:5062;tag=as06dddc8f’
– SIP/norman-0000006b is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [s@macro-dialout-trunk:33] NoOp(“SIP/103-0000006a”, “Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21”) in new stack
– Executing [s@macro-dialout-trunk:34] GotoIf(“SIP/103-0000006a”, “0?continue,1:s-CONGESTION,1”) in new stack
– Goto (macro-dialout-trunk,s-CONGESTION,1)
– Executing [s-CONGESTION@macro-dialout-trunk:1] Set(“SIP/103-0000006a”, “RC=21”) in new stack
– Executing [s-CONGESTION@macro-dialout-trunk:2] Goto(“SIP/103-0000006a”, “21,1”) in new stack
– Goto (macro-dialout-trunk,21,1)
– Executing [21@macro-dialout-trunk:1] Goto(“SIP/103-0000006a”, “continue,1”) in new stack
– Goto (macro-dialout-trunk,continue,1)
– Executing [continue@macro-dialout-trunk:1] NoOp(“SIP/103-0000006a”, “TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks”) in new stack
– Executing [continue@macro-dialout-trunk:2] ExecIf(“SIP/103-0000006a”, “1?Set(CALLERID(number)=103)”) in new stack
– Executing [TestCallOut@restrictedroute-c4ca4238a0b923820dcc509a6f75849b:8] Macro(“SIP/103-0000006a”, “outisbusy,”) in new stack
– Executing [s@macro-outisbusy:1] Progress(“SIP/103-0000006a”, “”) in new stack
– Executing [s@macro-outisbusy:2] GotoIf(“SIP/103-0000006a”, “0?emergency,1”) in new stack
– Executing [s@macro-outisbusy:3] GotoIf(“SIP/103-0000006a”, “0?intracompany,1”) in new stack
– Executing [s@macro-outisbusy:4] Playback(“SIP/103-0000006a”, “all-circuits-busy-now&please-try-call-later, noanswer”) in new stack
– <SIP/103-0000006a> Playing ‘all-circuits-busy-now.ulaw’ (language ‘en’)
– <SIP/103-0000006a> Playing ‘please-try-call-later.ulaw’ (language ‘en’)
– Executing [s@macro-outisbusy:5] Congestion(“SIP/103-0000006a”, “20”) in new stack
[2019-06-28 16:09:24] WARNING[8160][C-0000003f]: channel.c:5080 ast_prod: Prodding channel ‘SIP/103-0000006a’ failed
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on ‘SIP/103-0000006a’ in macro ‘outisbusy’
== Spawn extension (restrictedroute-c4ca4238a0b923820dcc509a6f75849b, TestCallOut, 8) exited non-zero on ‘SIP/103-0000006a’
– Executing [h@restrictedroute-c4ca4238a0b923820dcc509a6f75849b:1] Hangup(“SIP/103-0000006a”, “”) in new stack
== Spawn extension (restrictedroute-c4ca4238a0b923820dcc509a6f75849b, h, 1) exited non-zero on ‘SIP/103-0000006a’
– SIP/103-0000006a Internal Gosub(crm-hangup,s,1) start
– Executing [s@crm-hangup:1] NoOp(“SIP/103-0000006a”, “Sending Hangup to CRM”) in new stack
– Executing [s@crm-hangup:2] NoOp(“SIP/103-0000006a”, “HANGUP CAUSE: 34”) in new stack
– Executing [s@crm-hangup:3] ExecIf(“SIP/103-0000006a”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [s@crm-hangup:4] NoOp(“SIP/103-0000006a”, “MASTER CHANNEL: 1561752560.109 = 1561752560.109”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“SIP/103-0000006a”, “0?return”) in new stack
– Executing [s@crm-hangup:6] Set(“SIP/103-0000006a”, “__CRM_HANGUP=1”) in new stack
– Executing [s@crm-hangup:7] AGI(“SIP/103-0000006a”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <SIP/103-0000006a>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@crm-hangup:8] Return(“SIP/103-0000006a”, “”) in new stack
== Spawn extension (restrictedroute-c4ca4238a0b923820dcc509a6f75849b, h, 1) exited non-zero on ‘SIP/103-0000006a’
– SIP/103-0000006a Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
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Random asterisk unresponsiveness

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@MacsOffice wrote:

I went back to v13.22 but this has happened again. is there anything in these log excerpts that shed light on what happens ? - its been about once a week and its about due again…

The random non responsiveness continues - This is the log at the time it occurs

25 08:28:05] VERBOSE[30192][C-00000025] pbx.c: Executing [s@ivr-1:7] Answer(“SIP/5XXXXXX3-out-00000048”, “”) in new stack
[2019-06-25 08:28:05] VERBOSE[30192][C-00000025] pbx.c: Executing [s@ivr-1:8] Wait(“SIP/5XXXXXX3-out-00000048”, “1”) in new stack
[2019-06-25 08:28:06] VERBOSE[30192][C-00000025] pbx.c: Executing [s@ivr-1:9] Set(“SIP/5XXXXXX3-out-00000048”, “IVR_MSG=custom/ivr-welcome-header&custom/ivr-valley-open-choices”) in new stack
[2019-06-25 08:28:06] VERBOSE[30192][C-00000025] pbx.c: Executing [s@ivr-1:10] Set(“SIP/5XXXXXX3-out-00000048”, “TIMEOUT(digit)=3”) in new stack
[2019-06-25 08:28:06] VERBOSE[30192][C-00000025] func_timeout.c: Digit timeout set to 3.000
[2019-06-25 08:28:06] VERBOSE[30192][C-00000025] pbx.c: Executing [s@ivr-1:11] ExecIf(“SIP/5XXXXXX3-out-00000048”, “1?Background(custom/ivr-welcome-header&custom/ivr-valley-open-choices)”) in new stack
[2019-06-25 08:28:06] VERBOSE[30192][C-00000025] file.c: <SIP/5XXXXXX3-out-00000048> Playing ‘custom/ivr-welcome-header.slin’ (language ‘en’)
[2019-06-25 08:28:17] VERBOSE[30192][C-00000025] file.c: <SIP/5XXXXXX3-out-00000048> Playing ‘custom/ivr-valley-open-choices.slin’ (language ‘en’)
[2019-06-25 08:28:34] VERBOSE[30192][C-00000025] pbx.c: Spawn extension (ivr-1, s, 11) exited non-zero on ‘SIP/5XXXXXX3-out-00000048’
[2019-06-25 08:28:34] VERBOSE[30192][C-00000025] pbx.c: Executing [h@ivr-1:1] Hangup(“SIP/5XXXXXX3-out-00000048”, “”) in new stack
[2019-06-25 08:28:34] VERBOSE[30192][C-00000025] pbx.c: Spawn extension (ivr-1, h, 1) exited non-zero on ‘SIP/5XXXXXX3-out-00000048’
[2019-06-25 08:28:34] VERBOSE[30192][C-00000025] app_stack.c: SIP/5XXXXXX3-out-00000048 Internal Gosub(crm-hangup,s,1) start
[2019-06-25 08:28:34] VERBOSE[30192][C-00000025] pbx.c: Executing [s@crm-hangup:1] NoOp(“SIP/5XXXXXX3-out-00000048”, “Sending Hangup to CRM”) in new stack
[2019-06-25 08:28:34] VERBOSE[30192][C-00000025] pbx.c: Executing [s@crm-hangup:2] NoOp(“SIP/5XXXXXX3-out-00000048”, “HANGUP CAUSE: 16”) in new stack
[2019-06-25 08:28:34] VERBOSE[30192][C-00000025] pbx.c: Executing [s@crm-hangup:3] ExecIf(“SIP/5XXXXXX3-out-00000048”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
[2019-06-25 08:28:34] VERBOSE[30192][C-00000025] pbx.c: Executing [s@crm-hangup:4] NoOp(“SIP/5XXXXXX3-out-00000048”, “MASTER CHANNEL: 1561476485.93 = 1561476485.93”) in new stack
[2019-06-25 08:28:34] VERBOSE[30192][C-00000025] pbx.c: Executing [s@crm-hangup:5] GotoIf(“SIP/5XXXXXX3-out-00000048”, “0?return”) in new stack
[2019-06-25 08:28:34] VERBOSE[30192][C-00000025] pbx.c: Executing [s@crm-hangup:6] Set(“SIP/5XXXXXX3-out-00000048”, “__CRM_HANGUP=1”) in new stack
[2019-06-25 08:28:34] VERBOSE[30192][C-00000025] pbx.c: Executing [s@crm-hangup:7] AGI(“SIP/5XXXXXX3-out-00000048”, “sangomacrm.agi”) in new stack
[2019-06-25 08:28:34] VERBOSE[30192][C-00000025] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2019-06-25 08:58:08] VERBOSE[2677] loader.c: Reloading module ‘app_voicemail.so’ (Comedian Mail (Voicemail System))
[2019-06-25 08:58:08] WARNING[2677] app_voicemail.c: maxsilence should be less than minsecs or you may get empty messages
[2019-06-25 09:50:23] VERBOSE[11288] netsock2.c: Using SIP RTP TOS bits 184
[2019-06-25 09:50:23] VERBOSE[11288] netsock2.c: Using SIP RTP CoS mark 5
[2019-06-25 09:51:50] VERBOSE[11499] netsock2.c: Using SIP RTP TOS bits 184
[2019-06-25 09:51:50] VERBOSE[11499] netsock2.c: Using SIP RTP CoS mark 5
[2019-06-25 09:52:09] VERBOSE[11649] netsock2.c: Using SIP RTP TOS bits 184
[2019-06-25 09:52:09] VERBOSE[11649] netsock2.c: Using SIP RTP CoS mark 5
[2019-06-25 09:52:43] VERBOSE[11682] netsock2.c: Using SIP RTP TOS bits 184

Issued Reboot -h

[2019-06-25 10:06:07] VERBOSE[13981] netsock2.c: Using SIP RTP CoS mark 5
[2019-06-25 10:07:59] VERBOSE[14220] netsock2.c: Using SIP RTP TOS bits 184
[2019-06-25 10:07:59] VERBOSE[14220] netsock2.c: Using SIP RTP CoS mark 5
[2019-06-25 11:29:40] VERBOSE[15699] asterisk.c: Remote UNIX connection
[2019-06-25 11:30:20] VERBOSE[28243] asterisk.c: Remote UNIX connection disconnected
[2019-06-25 11:30:34] VERBOSE[15664] asterisk.c: Asterisk uncleanly ending (0).
[2019-06-25 11:30:34] VERBOSE[15664] asterisk.c: Executing last minute cleanups
[2019-06-25 11:30:34] VERBOSE[15664] res_musiconhold.c: Destroying musiconhold processes
[2019-06-25 11:30:35] VERBOSE[15664] manager.c: Manager unregistered action DBGet
[2019-06-25 11:30:35] VERBOSE[15664] manager.c: Manager unregistered action DBPut
[2019-06-25 11:30:35] VERBOSE[15664] manager.c: Manager unregistered action DBDel
[2019-06-25 11:30:35] VERBOSE[15664] manager.c: Manager unregistered action DBDelTree
[2019-06-25 11:32:20] Asterisk 13.22.0 built by mockbuild @ jenkins7 on a x86_64 running Linux on 2018-07-25 22:30:39 UTC
[2019-06-25 11:32:20] VERBOSE[5889] message.c: Message handler ‘dialplan’ registered.

And what is this message you have to be kidding?
[2019-06-25 11:32:20] WARNING[5889] presencestate.c: No provider found for label CustomPresence
[2019-06-25 11:32:20] WARNING[5889] presencestate.c: No provider found for label CustomPresence
[2019-06-25 11:32:20] WARNING[5889] presencestate.c: No provider found for label CustomPresence
[2019-06-25 11:32:20] WARNING[5889] pbx.c: Unable to register extension ‘*8670’ priority -1 in ‘park-hints’, already in use
[2019-06-25 11:32:20] WARNING[5889] pbx_config.c: Unable to register extension at line 4505 of /etc/asterisk/extensions_additional.conf
[2019-06-25 11:32:20] WARNING[5889] pbx.c: Extension ‘s’ priority 13 in ‘macro-user-logon’, label ‘gotpass’ already in use at priority 9
[2019-06-25 11:32:20] ERROR[5889] pbx.c: You have to be kidding – add exten ‘’ to context clean? Figure out a name and call me back. Action ignored.

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Queue only fails to failover location

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@Michael1194 wrote:

Hello, everybody,

I’ve been working on completing my FreePBX installation for a few hours now.
Currently I have packed it so far that it arrives directly at my end device without the queue.

However, if I have switched on the queue, the call does not go to my end devices, but directly to my mailbox, which I have configured as failover.

My queue configuration is as follows:

Queue Agents:
Static Agents: 100,0 and 101,0

If you need more configuration points, please let me know.

Regards from Germany

Michael

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After 15 upgrade cannot upgrade endpoint manager 13.0.71

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@bob_dt wrote:

I’m confused . . . I have upgraded from 14 to 15 and now FreePBX is insisting that ‘endpoint manger’ needs to be upgraded. Any attempt to upgrade fails. An attempt to uninstall EM suggests that ‘Restapps’ module needs to be disabled and it is disabled, according to dashboard but, no love! Will not uninstall.

I have run “fwconsole ma refreshsignatures”. And treid again . . . no love.

I think (could be wrong) that I need to uninstall and reinstall endpoint manager 13.0.71?

Not sure how to proceed?

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Being in an IVR, how I can "force" via an external action to move to another?

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@lmon wrote:

Hi all,
I posted several questions setting up an IVR … ( for reference : IVR - Execute Shell Script / PHP file ) .
I would need now to have an external action ( external to Freepbx / asterisk ) to make an IVR move to another one. I found that there is a REST API available ( https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573) but I don’t know which one ( AGI ou ARI ) and how to perform the following :

the user already went through several steps through IVRs… one announcement is saying that the caller must wait for an external validation…
From another system, completely unliked, someone performs an API call which will make the user currently on the IVR workflow waiting, moving to a next step.
any hints /suggestions?
Thanks

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Zero touch configuration

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@antonis77 wrote:

Hi,

I want to setup zero touch configuration for S206 and S500 phones.
I ve registered my phones to portal using the internal IP since my phones are all local.

When i reboot the phone it shows a deployment id which belongs to a test pbx and not the production PBX server.

Does it matter that both of them are in the same LAN ?

Thanks

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Asterisk ARI

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@AnthKay92 wrote:

We have two extensions provisioned on our PBX. They are able to call each other fine with no issues.

We also have an application that needs to do the following using Asterisk ARI:
Existing Functionality in App (CTI) –

The application, should make phone calls (Inbound and Outbound) using CTI functionality.

Inbound : When if there is any inbound call, popups the customer related information based on the incoming phone number, in the Client application.

Outbound : When TSO (telesales operator), clicks on call option in TESS Client, CTI application will place call to the customer.

In Asterisk, we need to achieve the same functionality. As observed, in the ARI, we are not receiving the call events.

We have tested in the .NET ARI as well as with “wscat” command (the details are available in the Asterisk official site: https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers)

Following are the step by step testing details (steps available in https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers)

  1. Executed the wscat command in Node.Js prompt.

wscat -c "ws://10.38.240.1:8088/ari/events?api_key=3rd_Party_App:1d8c16f0eebafafceb7e0f6ab181e3f5&app=Tess_Asterisk"

  1. In Asterisk, the new WebSocket connection and a message telling us that our Stasis application has been created

  2. Dialled 2001 extension from SIP phone, but none of the events are displayed in the wscat window.

In wscat we should see the StasisStart even indicating that a channel has entered into the stasis application, however nothing does.

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All calls going to failover destination

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@Rusent wrote:

Hi, we ran some updates on our distro server to bring FreePBX up to the latest 10.13.66-22 version, however are now having some queue issues.

We have a number of queues, all which have a “rescue” queue set as the Fail Over Destination. We are finding that since the update all incoming calls go to that failover destination. I tried setting one of the queues to point to another queue but then it just loops and fails.

We had this exact issue last time we installed updates but unfortunately don’t have a documented fix (we changed some timeout settings and restarted the server, but I am not sure if anything specifically resolved the issue.) We have a number of agents signed in on all of the queues so can;t see why anything should be going to the failover destination.

Has anybody experienced the issue, or does anybody have any idea why this may be happening? Thanks for any help!

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Difficulties with external calls

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@jefersonp22 wrote:

I’m having trouble making external calls
on the first attempt what appears in the
CLI: Setting global variable ‘SIPDOMAIN’ to ‘10.0.60.175’
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
– Executing [0999103460@from-internal:1] Macro(“PJSIP/8003-00000e4e”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“PJSIP/8003-00000e4e”, “TOUCH_MONITOR=1559573187.3960”) in new stack
– Executing [s@macro-user-callerid:2] Set(“PJSIP/8003-00000e4e”, “AMPUSER=8003”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“PJSIP/8003-00000e4e”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“PJSIP/8003-00000e4e”, “1?Set(REALCALLERIDNUM=8003)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“PJSIP/8003-00000e4e”, “AMPUSER=8003”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“PJSIP/8003-00000e4e”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“PJSIP/8003-00000e4e”, “AMPUSERCIDNAME=8003”) in new stack
– Executing [s@macro-user-callerid:8] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
– Executing [s@macro-user-callerid:9] GotoIf(“PJSIP/8003-00000e4e”, “0?report”) in new stack
– Executing [s@macro-user-callerid:10] Set(“PJSIP/8003-00000e4e”, “AMPUSERCID=8003”) in new stack
– Executing [s@macro-user-callerid:11] Set(“PJSIP/8003-00000e4e”, “__DIAL_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-user-callerid:12] Set(“PJSIP/8003-00000e4e”, “CALLERID(all)=“8003” <8003>”) in new stack
– Executing [s@macro-user-callerid:13] GotoIf(“PJSIP/8003-00000e4e”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:14] ExecIf(“PJSIP/8003-00000e4e”, “1?Set(GROUP(concurrency_limit)=8003)”) in new stack
– Executing [s@macro-user-callerid:15] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:16] NoOp(“PJSIP/8003-00000e4e”, “Macro Depth is 1”) in new stack
– Executing [s@macro-user-callerid:17] GotoIf(“PJSIP/8003-00000e4e”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,18)
– Executing [s@macro-user-callerid:18] GotoIf(“PJSIP/8003-00000e4e”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,37)
– Executing [s@macro-user-callerid:37] Set(“PJSIP/8003-00000e4e”, “CALLERID(number)=8003”) in new stack
– Executing [s@macro-user-callerid:38] Set(“PJSIP/8003-00000e4e”, “CALLERID(name)=8003”) in new stack
– Executing [s@macro-user-callerid:39] GotoIf(“PJSIP/8003-00000e4e”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:40] Set(“PJSIP/8003-00000e4e”, “CDR(cnam)=8003”) in new stack
– Executing [s@macro-user-callerid:41] Set(“PJSIP/8003-00000e4e”, “CDR(cnum)=8003”) in new stack
– Executing [s@macro-user-callerid:42] Set(“PJSIP/8003-00000e4e”, “CHANNEL(language)=en”) in new stack
– Executing [0999103460@from-internal:2] Gosub(“PJSIP/8003-00000e4e”, “sub-record-check,s,1(out,0999103460,dontcare)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“PJSIP/8003-00000e4e”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“PJSIP/8003-00000e4e”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“PJSIP/8003-00000e4e”, “NOW=1559573187”) in new stack
– Executing [s@sub-record-check:4] Set(“PJSIP/8003-00000e4e”, “__DAY=03”) in new stack
– Executing [s@sub-record-check:5] Set(“PJSIP/8003-00000e4e”, “__MONTH=06”) in new stack
– Executing [s@sub-record-check:6] Set(“PJSIP/8003-00000e4e”, “__YEAR=2019”) in new stack
– Executing [s@sub-record-check:7] Set(“PJSIP/8003-00000e4e”, “__TIMESTR=20190603-144627”) in new stack
– Executing [s@sub-record-check:8] Set(“PJSIP/8003-00000e4e”, “__FROMEXTEN=8003”) in new stack
– Executing [s@sub-record-check:9] Set(“PJSIP/8003-00000e4e”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“PJSIP/8003-00000e4e”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“PJSIP/8003-00000e4e”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“PJSIP/8003-00000e4e”, “3?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“PJSIP/8003-00000e4e”, “1?sub-record-check,out,1”) in new stack
– Goto (sub-record-check,out,1)
– Executing [out@sub-record-check:1] NoOp(“PJSIP/8003-00000e4e”, “Outbound Recording Check from 8003 to 0999103460”) in new stack
– Executing [out@sub-record-check:2] Set(“PJSIP/8003-00000e4e”, “RECMODE=dontcare”) in new stack
– Executing [out@sub-record-check:3] ExecIf(“PJSIP/8003-00000e4e”, “1?Goto(routewins)”) in new stack
– Goto (sub-record-check,out,7)
– Executing [out@sub-record-check:7] Gosub(“PJSIP/8003-00000e4e”, “recordcheck,1(dontcare,out,0999103460)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“PJSIP/8003-00000e4e”, “Starting recording check against dontcare”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“PJSIP/8003-00000e4e”, “dontcare”) in new stack
– Goto (sub-record-check,recordcheck,3)
– Executing [recordcheck@sub-record-check:3] Return(“PJSIP/8003-00000e4e”, “”) in new stack
– Executing [out@sub-record-check:8] Return(“PJSIP/8003-00000e4e”, “”) in new stack
– Executing [0999103460@from-internal:3] ExecIf(“PJSIP/8003-00000e4e”, “0 ?Set(CDR(accountcode)=)”) in new stack
– Executing [0999103460@from-internal:4] Set(“PJSIP/8003-00000e4e”, “MOHCLASS=default”) in new stack
– Executing [0999103460@from-internal:5] Set(“PJSIP/8003-00000e4e”, “_NODEST=”) in new stack
– Executing [0999103460@from-internal:6] Macro(“PJSIP/8003-00000e4e”, “dialout-trunk,1,0999103460,off”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“PJSIP/8003-00000e4e”, “DIAL_TRUNK=1”) in new stack
– Executing [s@macro-dialout-trunk:2] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(DIAL_OPTIONS=Hhtr)”) in new stack
– Executing [s@macro-dialout-trunk:3] GosubIf(“PJSIP/8003-00000e4e”, “0?sub-pincheck,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:4] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(CALLERID(num)=8003)”) in new stack
– Executing [s@macro-dialout-trunk:5] GotoIf(“PJSIP/8003-00000e4e”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“PJSIP/8003-00000e4e”, “DIAL_NUMBER=0999103460”) in new stack
– Executing [s@macro-dialout-trunk:7] Set(“PJSIP/8003-00000e4e”, “DIAL_TRUNK_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-dialout-trunk:8] Set(“PJSIP/8003-00000e4e”, “OUTBOUND_GROUP=OUT_1”) in new stack
– Executing [s@macro-dialout-trunk:9] Set(“PJSIP/8003-00000e4e”, “DIAL_TRUNK_OPTIONS=T”) in new stack
– Executing [s@macro-dialout-trunk:10] GotoIf(“PJSIP/8003-00000e4e”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,12)
– Executing [s@macro-dialout-trunk:12] GotoIf(“PJSIP/8003-00000e4e”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:13] Macro(“PJSIP/8003-00000e4e”, “outbound-callerid,1”) in new stack
– Executing [s@macro-outbound-callerid:1] NoOp(“PJSIP/8003-00000e4e”, “8003”) in new stack
– Executing [s@macro-outbound-callerid:2] NoOp(“PJSIP/8003-00000e4e”, “”) in new stack
– Executing [s@macro-outbound-callerid:3] NoOp(“PJSIP/8003-00000e4e”, “off”) in new stack
– Executing [s@macro-outbound-callerid:4] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(CALLERPRES(name-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:5] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(CALLERPRES(num-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:6] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(REALCALLERIDNUM=8003)”) in new stack
– Executing [s@macro-outbound-callerid:7] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(AMPUSER=8003)”) in new stack
– Executing [s@macro-outbound-callerid:8] GotoIf(“PJSIP/8003-00000e4e”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [s@macro-outbound-callerid:12] Set(“PJSIP/8003-00000e4e”, “USEROUTCID=8003”) in new stack
– Executing [s@macro-outbound-callerid:13] Set(“PJSIP/8003-00000e4e”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:14] Set(“PJSIP/8003-00000e4e”, “TRUNKOUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:15] GotoIf(“PJSIP/8003-00000e4e”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,21)
– Executing [s@macro-outbound-callerid:21] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:22] ExecIf(“PJSIP/8003-00000e4e”, “1?Set(CALLERID(all)=8003)”) in new stack
– Executing [s@macro-outbound-callerid:23] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:24] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:25] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:26] Set(“PJSIP/8003-00000e4e”, “CDR(outbound_cnum)=8003”) in new stack
– Executing [s@macro-outbound-callerid:27] Set(“PJSIP/8003-00000e4e”, “CDR(outbound_cnam)=”) in new stack
– Executing [s@macro-dialout-trunk:14] GosubIf(“PJSIP/8003-00000e4e”, “0?sub-flp-1,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:15] Set(“PJSIP/8003-00000e4e”, “OUTNUM=0999103460”) in new stack
– Executing [s@macro-dialout-trunk:16] Set(“PJSIP/8003-00000e4e”, “custom=SIP/6350”) in new stack
– Executing [s@macro-dialout-trunk:17] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)”) in new stack
– Executing [s@macro-dialout-trunk:18] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))”) in new stack
– Executing [s@macro-dialout-trunk:19] Macro(“PJSIP/8003-00000e4e”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“PJSIP/8003-00000e4e”, “”) in new stack
– Executing [s@macro-dialout-trunk:20] GotoIf(“PJSIP/8003-00000e4e”, “0?skipcrm”) in new stack
– Executing [s@macro-dialout-trunk:21] Set(“PJSIP/8003-00000e4e”, “__CRM_DIRECTION=OUTBOUND”) in new stack
– Executing [s@macro-dialout-trunk:22] Set(“PJSIP/8003-00000e4e”, “__CRM_DESTINATION=0999103460”) in new stack
– Executing [s@macro-dialout-trunk:23] Set(“PJSIP/8003-00000e4e”, “__CRM_SOURCE=8003”) in new stack
– Executing [s@macro-dialout-trunk:24] AGI(“PJSIP/8003-00000e4e”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <PJSIP/8003-00000e4e>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@macro-dialout-trunk:25] Set(“PJSIP/8003-00000e4e”, “CHANNEL(hangup_handler_push)=crm-hangup,s,1”) in new stack
– Executing [s@macro-dialout-trunk:26] NoOp(“PJSIP/8003-00000e4e”, “CRM Finished”) in new stack
– Executing [s@macro-dialout-trunk:27] GotoIf(“PJSIP/8003-00000e4e”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:28] ExecIf(“PJSIP/8003-00000e4e”, “1?Set(CONNECTEDLINE(num,i)=0999103460)”) in new stack
– Executing [s@macro-dialout-trunk:29] ExecIf(“PJSIP/8003-00000e4e”, “1?Set(CONNECTEDLINE(name,i)=CID:8003)”) in new stack
– Executing [s@macro-dialout-trunk:30] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)8003)”) in new stack
– Executing [s@macro-dialout-trunk:31] GotoIf(“PJSIP/8003-00000e4e”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:32] Dial(“PJSIP/8003-00000e4e”, “SIP/6350/0999103460@6350,300,Tb(func-apply-sipheaders^s^1,(1))”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– SIP/6350-00000128 Internal Gosub(func-apply-sipheaders,s,1(1)) start
– Executing [s@func-apply-sipheaders:1] ExecIf(“SIP/6350-00000128”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
– Executing [s@func-apply-sipheaders:2] NoOp(“SIP/6350-00000128”, “Applying SIP Headers to channel SIP/6350-00000128”) in new stack
– Executing [s@func-apply-sipheaders:3] Set(“SIP/6350-00000128”, “TECH=SIP”) in new stack
– Executing [s@func-apply-sipheaders:4] Set(“SIP/6350-00000128”, “SIPHEADERKEYS=”) in new stack
– Executing [s@func-apply-sipheaders:5] While(“SIP/6350-00000128”, “0”) in new stack
– Jumping to priority 12
– Executing [s@func-apply-sipheaders:13] Return(“SIP/6350-00000128”, “”) in new stack
== Spawn extension (from-pstn, 0999103460, 1) exited non-zero on ‘SIP/6350-00000128’
– SIP/6350-00000128 Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
– Called SIP/6350/0999103460@6350
– SIP/6350-00000128 is ringing
[2019-06-03 14:46:36] WARNING[13270]: res_pjsip_registrar.c:941 find_registrar_aor: AOR ‘8009’ not found for endpoint ‘interno_adm’
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-dialout-trunk:33] NoOp(“PJSIP/8003-00000e4e”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 127”) in new stack
– Executing [s@macro-dialout-trunk:34] GotoIf(“PJSIP/8003-00000e4e”, “0?continue,1:s-CHANUNAVAIL,1”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set(“PJSIP/8003-00000e4e”, “RC=127”) in new stack
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto(“PJSIP/8003-00000e4e”, “127,1”) in new stack
– Goto (macro-dialout-trunk,127,1)
– Executing [127@macro-dialout-trunk:1] Goto(“PJSIP/8003-00000e4e”, “continue,1”) in new stack
– Goto (macro-dialout-trunk,continue,1)
– Executing [continue@macro-dialout-trunk:1] NoOp(“PJSIP/8003-00000e4e”, “TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 127 - failing through to other trunks”) in new stack
– Executing [continue@macro-dialout-trunk:2] ExecIf(“PJSIP/8003-00000e4e”, “1?Set(CALLERID(number)=8003)”) in new stack
– Executing [0999103460@from-internal:7] Macro(“PJSIP/8003-00000e4e”, “outisbusy,”) in new stack
– Executing [s@macro-outisbusy:1] Progress(“PJSIP/8003-00000e4e”, “”) in new stack
– Executing [s@macro-outisbusy:2] GotoIf(“PJSIP/8003-00000e4e”, “0?emergency,1”) in new stack
– Executing [s@macro-outisbusy:3] GotoIf(“PJSIP/8003-00000e4e”, “0?intracompany,1”) in new stack
– Executing [s@macro-outisbusy:4] Playback(“PJSIP/8003-00000e4e”, “all-circuits-busy-now&please-try-call-later, noanswer”) in new stack
– <PJSIP/8003-00000e4e> Playing ‘all-circuits-busy-now.g722’ (language ‘en’)
> 0x7fc61004f6c0 – Strict RTP learning after remote address set to: 10.0.91.148:11914
> 0x7fc61004f6c0 – Strict RTP switching to RTP target address 10.0.91.148:11914 as source
– <PJSIP/8003-00000e4e> Playing ‘please-try-call-later.ulaw’ (language ‘en’)
– Executing [s@macro-outisbusy:5] Congestion(“PJSIP/8003-00000e4e”, “20”) in new stack
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on ‘PJSIP/8003-00000e4e’ in macro ‘outisbusy’
== Spawn extension (from-internal, 0999103460, 7) exited non-zero on ‘PJSIP/8003-00000e4e’
– Executing [h@from-internal:1] Macro(“PJSIP/8003-00000e4e”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/8003-00000e4e”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“PJSIP/8003-00000e4e”, " montior file= ") in new stack
– Executing [s@macro-hangupcall:5] GotoIf(“PJSIP/8003-00000e4e”, “1?skipagi”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [s@macro-hangupcall:7] Hangup(“PJSIP/8003-00000e4e”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/8003-00000e4e’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/8003-00000e4e’
– PJSIP/8003-00000e4e Internal Gosub(crm-hangup,s,1) start
– Executing [s@crm-hangup:1] NoOp(“PJSIP/8003-00000e4e”, “Sending Hangup to CRM”) in new stack
– Executing [s@crm-hangup:2] NoOp(“PJSIP/8003-00000e4e”, “HANGUP CAUSE: 34”) in new stack
– Executing [s@crm-hangup:3] ExecIf(“PJSIP/8003-00000e4e”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [s@crm-hangup:4] NoOp(“PJSIP/8003-00000e4e”, “MASTER CHANNEL: 1559573187.3960 = 1559573187.3960”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“PJSIP/8003-00000e4e”, “0?return”) in new stack
– Executing [s@crm-hangup:6] Set(“PJSIP/8003-00000e4e”, “__CRM_HANGUP=1”) in new stack
– Executing [s@crm-hangup:7] AGI(“PJSIP/8003-00000e4e”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <PJSIP/8003-00000e4e>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@crm-hangup:8] Return(“PJSIP/8003-00000e4e”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/8003-00000e4e’
– PJSIP/8003-00000e4e Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

And the call drops

On the second attempt
The call is completed successfully.

Someone could tell me why I can not complete call on first attempt ?

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Cannot Update Anything

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@brodiemac wrote:

When I try to update modules, I get the error “Warning: Cannot connect to online repository(s) (http mirror1 dot freepbx dot org,http mirror2 dot freepbx dot org). Online modules are not available.” When I try to update the system, I get the error "System Update Details
Current System Update Status:Idle
Last Online Check Status:3 minutes ago (Yum Error)
Last System Update:Unknown (System updates not run since last reboot)
Updates Available:Unable to run ‘yum check-updates’, can’t check for updates

I’m running version 14.0.11

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How Zulu logs in with certificates on the Intranet

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@zhoubowen wrote:

Hello:

I want to test Zulu on PC in our LAN environment, but he needs FQDN. Can we use LAN IP address?

Can we test it in LAN environment?

Thank you!

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Https://community.freepbx.org/t/calendar-module-problems/47173

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@Eugene_85 wrote:

I’ve got a freepbx FreePBX 14.0.13.4 and i’ve got a troubles with local calendar.


l was waiting resolve of the problem in latest versions, but nothing has changed.

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Endpoint manager

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@antonis77 wrote:

Hi everyone,

In the endpoint manager in the sangoma template ive setup an s500 phone with different BLF buttons.

In the 100 extension for example i ve seen that there is a BLF button for this phone

In the 101 extension for example i ve seen that there is a BLF button for this phone too,

Is there a way to not showing a BLF for the phone itself ?

I want all the others to 've a BLF button for other phones but not a BLF for itself ?

Is it possible ?

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Inbound Routes, Not Routing

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@bonifacefj wrote:

I have recently installed a FreePBX server and set it up with a trunk.

All was working well until I added a new extension and associated inbound route.

I have a block of numbers ending with 00 - 09. Currently only three are in use, 00, 01 & 09. The inbound routes I have setup are below.

  • 00 Routes to default IVR using the DID rule as _X.00

  • 01 Routes to a ring group, using the DID rule as _X.01

  • 09 Routes to a second IVR using the DID rule as _X.09

As I said prior to setting the newest extention and route (09) the two other routes worked as expected, but now calls to any of the three DID’s routes to my default IVR. The DID’s are definitely being presented as expected and are shown in the call logs.

I’ll also clarify that I only have one trunk set up and only have the three inbound routes set up with no others yet.

Any guidance would be appreciated!

Fred

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Stripping "+" from CID presented to phone?

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@bobf wrote:

We are using FreePBX 13.0.195.4 with Twilio as our inbound and outbound SIP trunking provider. Everything is working well EXCEPT for an annoyance that the inbound (to us) caller ID has “+1” at the start and the “redial” function on our phones (Cisco CP-8841-3PCC) seems to balk at redialing numbers that start with “+”.

How do we strip that “+” before it gets to the phone?

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Play announcement on ALL inbound routes then go to normal routing

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@Kafluke wrote:

I have about 400 inbound routes that need to play an announcement before being routed normally. “For quality and training purposes, this call my be monitored.”

I started working through the list but I don’t want to have to make 400 separate announcements.

I tried adding this to my “extensions_custom.conf” file and then changing my inbound trunk context to “context=inbound-play-msg-to-all” but it didn’t work. Didn’t seem to affect any incoming calls at all.

[inbound-play-msg-to-all]
exten => s,1,NoOp(Lets play a message for all incoming messages)
same => s,Playback(quality_and_training_advisement_pro.wav)
same => s,Goto(ext-did,${EXTEN},1)
same => s,Hangup

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Managing updates to FPBX systems in groups or rings of systems

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@bmartindcs wrote:

We manage a growing number of FPBX systems. All systems are using the Sangoma distro, and all systems have standardized virtual machine hardware in the datacenters we host from and all that works fine (finally, it took a LOT of time to get that stuff all ironed out).

The main challenge still bugging us is how to handle/manage updates to the OS, and the updates to Freepbx modules.

Of course we can yum update the OS related stuff or use the System Updates in GUI, and then use the module admin to do the module updates, but I want to keep all updates on all systems consistent, as some module updates invariably introduce bugs or feature changes that would blindside us or the end users, so we don’t leave “auto-update” on for that reason either. I would like to approach this with “release rings” methodology ideally.

We’d have our inner ring with a lab system that we do initial testing with. Those updates are then “approved” by us to our next ring, which would consist of our own production PBX. After X time and we bless those updates at our ring, the next ring of a core group of clients can get them and after some more time the outer ring of clients can get those.

Currently when doing updates you can only update to “now” and not specifically install updates up to a certain level/date. I think the easiest way to define the “lines in the sand” update-wise would be to allow updates that existed as of X date rather than trying to police each modules individual versions

As an example, on one system I’ll use the FPBX Framework Module as an example. As of right this second it’s on 14.0.11. “check for updates” returns that as of right now the update will take me to 14.0.13.4. If I look at release notes there has been several iterations between 14.0.11 and 14.0.13.4, but we can’t select any of those as what we want to update to. We can only jump to the most current which is 14.0.13.4. The “Previous” feature allows me to roll back to updates that existed BEFORE my currently installed version. I could theoretically update to 14.0.13.4, then use the “previous” function to go down to a specific one, but that would be a ton of manual work on one system, let alone many.

Any ideas on how we can accomplish this?

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Issue with directory and issue with phone

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@jhelms wrote:

My first issue is with my phone directory. It is set up on nginx and it worked for a while then all of a sudden it stopped working and gives the error “xml format error or invalid application”…

My second issue is we have 8 sites using the VOIP system, one site will not show caller id for incoming calls, it only shows “anonymous” unless it is within our system. Any call from a landline or cell phone shows anonymous and extension dialing shows which it should as they are all linked together.

Does anyone have any tips or ideas, I recently was handed this when our main person left the company so I am not all the way up to speed with this stuff. Any and all help would be appreciated!

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UCP broken due to corrupted JS files

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@dominic wrote:

Every so often, I try to log into the UCP and I get cloud with a gear (the loading indicator) that never goes away. I check the JS console and see some errors. It turns out that one or more of the JS files is corrupted. About half way through the file there is a bunch of random garbage that doesn’t belong there. I check the file on the server disk, and it’s fine. So, somehow it’s being corrupted in transit.

Today when it happened I noticed that some of the random garbage actually looked like PHP code. Has anyone seen anything like this? Any ideas on what’s causing it? The only solution I have found is a full server reboot.

Example corrupted file

Here is a section of this file: /ucp/assets/js/compiled/main/jsphpg_a84f8885e18c7255b6baebc581baad46.js?load_version=v14.0.3.3

The PHP code that’s showed up and definitely doesn’t belong looks like it’s from libphonenumber-for-php. That library is used in many locations, for example: /var/www/html/admin/modules/vqplus/vendor/giggsey/libphonenumber-for-php/src/geocoding/data/zh/86155.php

function(t){if(null==t[0]||null==t[1]||null==t[2])return[null,null,null,t[3]];var e,i,s=t[0]/255,n=t[1]/255,o=t[2]/255,a=t[3],r=Math.max(s,n,o),l=Math.min(s,n,o),h=r-l,c=r+l,u=.5*c;return e=l===r?0:s===r?60*(n-o)/h+360:n===r?60*(o-s)/h+120:60*(s-n)/h+240,i=0===h?0:.5>=u?h/c:h/(2-c),[Math.round(e)%360,i,u,null==a?1:a]
},c.hsla.from=function(t){if(null==t[0]||null==t[1]||null==t[2])return[null,null,nu  861552657 => '剿ž—çœå‰æž—市',
  861552658 => '剿ž—çœå››å¹³å¸‚',
  861552659 => '剿ž—çœå››å¹³å¸‚',
  861552660 => '剿ž—çœé€šåŒ–市',
  861552661 => '剿ž—çœé€šåŒ–市',
  861552662 => '剿ž—çœé€šåŒ–市',
  861552663 => '剿ž—çœé•¿æ˜¥å¸‚',
  861552664 => '剿ž—çœé•¿æ˜¥å¸‚',
  861552665 => '剿ž—çœé•¿æ˜¥å¸‚',
  861552666 => '剿ž—çœé•¿æ˜¥å¸‚',
  861552667 => '剿ž—çœé•¿æ˜¥å¸‚',
  861552668 => '剿ž—çœç™½åŸŽå¸‚',
  861552669 => '剿ž—çœç™½åŸŽå¸‚',
  86155267 => '剿ž—çœå»¶è¾¹æœé²œæ—自治州',
  86155268 => '剿ž—çœé•¿æ˜¥å¸‚',
  861552690 => '剿ž—çœè¾½æºå¸‚',
  861552691 => '剿ž—çœè¾½æºå¸‚',
  861552692 => '剿ž—çœç™½å±±å¸‚',
  861552693 => '剿ž—çœç™½å±±å¸‚',
  861552694 => '剿ž—çœç™½åŸŽå¸‚',
  861552695 => '剿ž—çœç™½åŸŽå¸‚',
  861552696 => '剿ž—çœç™½å±±å¸‚',
  861552697 => '剿ž—çœæ¾åŽŸå¸‚',
  861552698 => '剿ž—çœæ¾åŽŸå¸‚',
  861552699 => '剿ž—çœæ¾åŽŸå¸‚',
  8615527 => 'æ¹–åŒ—çœæ­¦æ±‰å¸‚',
  86155280 => 'å››å·çœæˆéƒ½å¸‚',
  86155281 => 'å››å·çœæˆéƒ½å¸‚',
  86155282 => 'å››å·çœæˆéƒ½å¸‚',
  86155283 => 'å››å·çœæˆéƒ½å¸‚',
  86155284 => 'å››å·çœæˆéƒ½å¸‚',
  86155285 => 'å››å·çœç»µé˜³å¸‚',
  861552860 => 'å››å·çœå—充市',
  861552861 => 'å››å·çœå—充市',
  861552862 => 'å››å·çœé›…安市',
  861552863 => 'å››å·çœé›…安市',
  861552864 => 'å››å·çœé›…安市',
  861552865 => 'å››å·çœé›…安市',
  861552866 => 'å››å·çœå·´ä¸­å¸‚',
  861552867 => 'å››å·çœå·´ä¸­å¸‚',
  861552868 => 'å››å·çœèµ„阳市',
  861552869 => 'å››å·çœèµ„阳市',
  861552870 => 'å››å·çœå®œå®¾å¸‚',
  861552871 => 'å››å·çœå®œå®¾å¸‚',
  861552872 => 'å››å·çœå®œå®¾å¸‚',
  861552873 => 'å››å·çœå®œå®¾å¸‚',
  861552874 => 'å››å·çœå®œå®¾å¸‚',
  861552875 => 'å››å·çœä¹å±±å¸‚',
  861552876 => 'å››å·çœä¹å±±å¸‚',
  861552877 => 'å››å·çœä¹å±±å¸‚',
  861552878 => 'å››å·çœä¹å±±å¸‚',
  861552879 => 'å››å·çœæˆéƒ½å¸‚',
  861552880 => 'å››å·çœå—充市',
  861552881 => 'å››å·çœå—充市',
  861552882 => 'å››å·çœå—充市',
  861552883 => 'å››å·çœå—充市',
  861552884 => 'å››å·çœå—充市',
  861552885 => 'å››å·çœå—充市',
  861552886 => 'å››å·çœèµ„阳市',
  861552887 => 'å››å·çœè¾¾å·žå¸‚',
  861552888 => 'å››å·çœè¾¾å·žå¸‚',
  861552889 => 'å››å·çœç”˜å­œè—æ—自治州',
  861552890 => 'å››å·çœé›…安市',
  861552891 => 'å››å·çœé›…安市',
  861552892 => 'å››å·çœå·´ä¸­å¸‚',
  861552893 => 'å››å·çœå·´ä¸­å¸‚',
  861552894 => 'å››å·çœå·´ä¸­å¸‚',
  861552895 => 'å››å·çœå®œå®¾å¸‚',
  861552896 => 'å››å·çœå®œå®¾å¸‚',
  861552897 => 'å››å·çœå®œå®¾å¸‚',
  861552898 => 'å››å·çœå®œå®¾å¸‚',
  861552899 => 'å››å·çœå®œå®¾å¸‚',
  86155290 => '陕西çœè¥¿å®‰å¸‚',
  861552910 => '陕西çœå’¸é˜³å¸‚',
  861552911 => '陕西çœå»¶å®‰å¸‚',
  861552912 => 'é™•è¥¿çœæ¦†æž—市',
  861552913 => 'é™•è¥¿çœæ¸­å—市',
  861552914 => '陕西çœå•†æ´›å¸‚',
  861552915 => '陕西çœå®‰åº·å¸‚',
  861552916 => 'é™•è¥¿çœæ±‰ä¸­å¸‚',
  861552917 => '陕西çœå®é¸¡å¸‚',
  861552918 => '陕西çœå®é¸¡å¸‚',
  861552919 => '陕西çœé“œå·å¸‚',
  86155292 => '陕西çœè¥¿å®‰å¸‚',
  86155293 => '陕西çœè¥¿å®‰å¸‚',
  86155294 => '陕西çœè¥¿å®‰å¸‚',
  86155295 => '陕西çœè¥¿å®‰å¸‚',
  86155296 => '陕西çœè¥¿å®‰å¸‚',
  86155297 => 'é™•è¥¿çœæ¦†æž—市',
  86155298 => 'é™•è¥¿çœæ¦†æž—市',
  86155299 => 'é™•è¥¿çœæ¦†æž—市',
  86155300 => '河北çœé‚¯éƒ¸å¸‚',
  86155301 => '河北çœçŸ³å®¶åº„市',
  86155302 => '河北çœä¿å®šå¸‚',
  861553030 => '河北çœå¼ å®¶å£å¸‚',
  861553031 => '河北çœå¼ å®¶å£å¸‚',
  861553032 => '河北çœå¼ å®¶å£å¸‚',
  861553033 => '河北çœå¼ å®¶å£å¸‚',
  861553034 => '河北çœå¼ å®¶å£å¸‚',
  861553035 => '河北çœå”山市',
  861553036 => '河北çœå¼ å®¶å£å¸‚',
  861553037 => '河北çœå¼ å®¶å£å¸‚',
  861553038 => '河北çœå¼ å®¶å£å¸‚',
  861553039 => '河北çœå¼ å®¶å£å¸‚',
  86155304 => 'æ²³åŒ—çœæ²§å·žå¸‚',
  86155305 => '河北çœå”山市',
  86155306 => '河北çœå»ŠåŠå¸‚',
  86155307 => 'æ²³åŒ—çœæ²§å·žå¸‚',
  86155308 => '河北çœå”山市',
  86155309 => '河北çœé‚¢å°å¸‚lass:function(e,i,s,n,o){return t.effects.animateClass.call(

Update

I tried reinstalling ucp using: fwconsole ma downloadinstall ucp

I got a new corruption, this time with python code being dumped into the js file. My new theory is that somehow random chunks of recent memory are being injected into the files that apache is serving. Now I’m concerned that this might be some kind of a security breach. Thoughts?

p.prototype.initServer=function(b,c,d){var e,f=this,g={},i={searchText:this.searchText,sortName:this.options.sortName,sortOrder:this.options.sortOrder};this.options.pagination&&(i.pageSize=this.options.pageSize===this.options.formatAllRows()?this.options.totalRows:this.options.pageSize,i.pageNumber=this.options.pageNumber),(d||this.options.url||th= head
            parent = None
            q = self.import_it(head, qname, parent)
            if q: return q, tail
        raise ImportError, "No module named '%s'" % qname

    def load_tail(self, q, tail):
        m = q
        while tail:
            i = tail.find('.')
            if i < 0: i = len(tail)
            head, tail = tail[:i], tail[i+1:]
            mname = "%s.%s" % (m.__name__, head)
            m = self.import_it(head, mname, m)
            if not m:
                raise ImportError, "No module named '%s'" % mname
        return m

    def ensure_fromlist(self, m, fromlist, recursive=0):
        for sub in fromlist:
            if sub == "*":
                if not recursive:
                    try:
                        all = m.__all__
                    except AttributeError:
                        pass
                    else:
                        self.ensure_fromlist(m, all, 1)
                continue
            if sub != "*" and not hasattr(m, sub):
                subname = "%s.%s" % (m.__name__, sub)
                submod = self.import_it(sub, subname, m)
                if not submod:
                    raise ImportError, "No module named '%s'" % subname

    def import_it(self, partname, fqname, parent, force_load=0):
        if not partname:
            # completely empty module name should only happen in
            # 'from . import' or __import__("")
            return parent
        if not force_load:
            try:
                return self.modules[fqname]
            except KeyError:
                pass
        try:
            path = parent and parent.__path__
        except AttributeError:
            return None
        partname = str(partname)
        stuff = self.loader.find_module(partname, path)
        if not stuff:
            return None
        fqname = str(fqname)
        m = self.loader.load_module(fqname, stuff)
        if parent:
            setattr(parent, partname, m)
        return m

    def reload(self, module):
        name = str(module.__name__)
        if '.' not in name:
            return self.import_it(name, name, None, force_load=1)
        i = name.rfind('.')
        pname = name[:i]
        parent = self.modules[pname]
        return self.import_it(name[i+1:], name, parent, force_load=1)


default_importer = None
current_importer = None

def install(importer = None):
    global current_importer
    current_importer = importer or default_importer or ModuleImporter()
    current_importer.install()

def uninstall():
    global current_importer
    current_importer.uninstall()
������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������eld="%s"]',l[e]));f.length>1&&(f=a(m[d[0].cellIndex])),f.find(".fht-cell").width(d.innerWidth())})

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Supervisor don't see playback recording in freePBX UCP dashboard

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0

@zhamepace0605 wrote:

My IT Head surely change the default location where call recordings are being saved, as a result Team Leads can no longer view playback button in UCP, they cannot do call listening.

I believe that UCP will get recording data from the default location.

Question: How do we change or tell UCP to get the recording from a new location that my IT Head define. see snip image, someone said I have to look for a file and edit it there.

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