@philip_rhoades wrote:

People,

I have been following Asterix from since before I retired and now I don’t have clients that might make use of a multi-user PBX but I have a few possible uses of the tech for my own non-profit organisations etc.

I installed the downloaded iso happily in a VM except for setting up a free SIP which required me to be in N America (I am in Australia).

So the first step for me is to set up a soft phone on my Fedora Linux workstation that can talk to the FPBX on the VM and just chat to someone . . somewhere else in the country or the world . . - is there another conveniet SIP provider than I can use?

Secondly, I would like to connect my single line PSTN connection to the workstation somehow - I presume this will require hardware of some sort to plug into either a USB port or a PCI slot?

Eventually I want to be able to use the setup as a digital answering machine for the PSTN line.

Thanks,
Phil.

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@edwinr wrote:

i got alot in my cdr where destination was log like ‘s [from-internal]’ instead of phone numbers or extensions.Please advise.
FreePBX 12.0.76.6

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@dsirota wrote:

Trying to run some updates to FreePBX
Version 15.0.16.52
Asterisk version 16.9.0

Everytime I run yum update I get this error message for each file after it runs a transaction check:

file /etc/asterisk/amd.conf from install of asterisk16-configs-16.9.0-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64

I’ve searched around here and can’t seem to find a working solution for this. Any help would be appreciated!

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@TechHome wrote:

I cant use the outbound routes. Every time I start a call I get call again later. That are the logs:

freepbx*CLI>
  == Setting global variable 'SIPDOMAIN' to '192.168.1.165'
  == 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 [05312504948@from-internal:1] Macro("PJSIP/11-00000012", "user-callerid,LIMIT,EXTERNAL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("PJSIP/11-00000012", "TOUCH_MONITOR=1590306339.18") in new stack
-- Executing [s@macro-user-callerid:2] Set("PJSIP/11-00000012", "AMPUSER=11") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("PJSIP/11-00000012", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("PJSIP/11-00000012", "1?Set(REALCALLERIDNUM=11)") in new stack
-- Executing [s@macro-user-callerid:5] Set("PJSIP/11-00000012", "AMPUSER=11") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("PJSIP/11-00000012", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("PJSIP/11-00000012", "AMPUSERCIDNAME=Marlon Otto") in new stack
-- Executing [s@macro-user-callerid:8] ExecIf("PJSIP/11-00000012", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
-- Executing [s@macro-user-callerid:9] GotoIf("PJSIP/11-00000012", "0?report") in new stack
-- Executing [s@macro-user-callerid:10] Set("PJSIP/11-00000012", "AMPUSERCID=11") in new stack
-- Executing [s@macro-user-callerid:11] Set("PJSIP/11-00000012", "__DIAL_OPTIONS=HhTtr") in new stack
-- Executing [s@macro-user-callerid:12] Set("PJSIP/11-00000012", "CALLERID(all)="Marlon Otto" <11>") in new stack
-- Executing [s@macro-user-callerid:13] ExecIf("PJSIP/11-00000012", "0?Set(CALLERID(all)=EXTERNAL)") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("PJSIP/11-00000012", "0?limit") in new stack
-- Executing [s@macro-user-callerid:15] ExecIf("PJSIP/11-00000012", "1?Set(GROUP(concurrency_limit)=11)") in new stack
-- Executing [s@macro-user-callerid:16] ExecIf("PJSIP/11-00000012", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:17] NoOp("PJSIP/11-00000012", "Macro Depth is 1") in new stack
-- Executing [s@macro-user-callerid:18] GotoIf("PJSIP/11-00000012", "1?report2:macroerror") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] GotoIf("PJSIP/11-00000012", "1?continue") in new stack
-- Goto (macro-user-callerid,s,38)
-- Executing [s@macro-user-callerid:38] Set("PJSIP/11-00000012", "CALLERID(number)=11") in new stack
-- Executing [s@macro-user-callerid:39] Set("PJSIP/11-00000012", "CALLERID(name)=Marlon Otto") in new stack
-- Executing [s@macro-user-callerid:40] GotoIf("PJSIP/11-00000012", "0?cnum") in new stack
-- Executing [s@macro-user-callerid:41] Set("PJSIP/11-00000012", "CDR(cnam)=Marlon Otto") in new stack
-- Executing [s@macro-user-callerid:42] Set("PJSIP/11-00000012", "CDR(cnum)=11") in new stack
-- Executing [s@macro-user-callerid:43] Set("PJSIP/11-00000012", "CHANNEL(language)=de_DE") in new stack
-- Executing [05312504948@from-internal:2] Gosub("PJSIP/11-00000012", "sub-record-check,s,1(out,05312504948,dontcare)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("PJSIP/11-00000012", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("PJSIP/11-00000012", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("PJSIP/11-00000012", "NOW=1590306339") in new stack
-- Executing [s@sub-record-check:4] Set("PJSIP/11-00000012", "__DAY=24") in new stack
-- Executing [s@sub-record-check:5] Set("PJSIP/11-00000012", "__MONTH=05") in new stack
-- Executing [s@sub-record-check:6] Set("PJSIP/11-00000012", "__YEAR=2020") in new stack
-- Executing [s@sub-record-check:7] Set("PJSIP/11-00000012", "__TIMESTR=20200524-094539") in new stack
-- Executing [s@sub-record-check:8] Set("PJSIP/11-00000012", "__FROMEXTEN=11") in new stack
-- Executing [s@sub-record-check:9] Set("PJSIP/11-00000012", "__MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:10] NoOp("PJSIP/11-00000012", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("PJSIP/11-00000012", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("PJSIP/11-00000012", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("PJSIP/11-00000012", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("PJSIP/11-00000012", "3?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("PJSIP/11-00000012", "1?sub-record-check,out,1") in new stack
-- Goto (sub-record-check,out,1)
-- Executing [out@sub-record-check:1] NoOp("PJSIP/11-00000012", "Outbound Recording Check from 11 to 05312504948") in new stack
-- Executing [out@sub-record-check:2] Set("PJSIP/11-00000012", "RECMODE=dontcare") in new stack
-- Executing [out@sub-record-check:3] ExecIf("PJSIP/11-00000012", "1?Goto(routewins)") in new stack
-- Goto (sub-record-check,out,7)
-- Executing [out@sub-record-check:7] Gosub("PJSIP/11-00000012", "recordcheck,1(dontcare,out,05312504948)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/11-00000012", "Starting recording check against dontcare") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("PJSIP/11-00000012", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [recordcheck@sub-record-check:3] Return("PJSIP/11-00000012", "") in new stack
-- Executing [out@sub-record-check:8] Return("PJSIP/11-00000012", "") in new stack
-- Executing [05312504948@from-internal:3] ExecIf("PJSIP/11-00000012", "0 ?Set(CDR(accountcode)=)") in new stack
-- Executing [05312504948@from-internal:4] Set("PJSIP/11-00000012", "MOHCLASS=default") in new stack
-- Executing [05312504948@from-internal:5] Set("PJSIP/11-00000012", "_NODEST=") in new stack
-- Executing [05312504948@from-internal:6] Macro("PJSIP/11-00000012", "dialout-trunk,1,05312504948,,off") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("PJSIP/11-00000012", "DIAL_TRUNK=1") in new stack
-- Executing [s@macro-dialout-trunk:2] ExecIf("PJSIP/11-00000012", "0?Set(DIAL_OPTIONS=Hhtr)") in new stack
-- Executing [s@macro-dialout-trunk:3] GosubIf("PJSIP/11-00000012", "0?sub-pincheck,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:4] ExecIf("PJSIP/11-00000012", "0?Set(CALLERID(num)=11)") in new stack
-- Executing [s@macro-dialout-trunk:5] GotoIf("PJSIP/11-00000012", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("PJSIP/11-00000012", "DIAL_NUMBER=05312504948") in new stack
-- Executing [s@macro-dialout-trunk:7] Set("PJSIP/11-00000012", "DIAL_TRUNK_OPTIONS=HhTtr") in new stack
-- Executing [s@macro-dialout-trunk:8] Set("PJSIP/11-00000012", "OUTBOUND_GROUP=OUT_1") in new stack
-- Executing [s@macro-dialout-trunk:9] Set("PJSIP/11-00000012", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dialout-trunk:10] GotoIf("PJSIP/11-00000012", "0?nomax") in new stack
-- Executing [s@macro-dialout-trunk:11] GotoIf("PJSIP/11-00000012", "0?chanfull") in new stack
-- Executing [s@macro-dialout-trunk:12] GotoIf("PJSIP/11-00000012", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:13] Macro("PJSIP/11-00000012", "outbound-callerid,1") in new stack
-- Executing [s@macro-outbound-callerid:1] NoOp("PJSIP/11-00000012", "11") in new stack
-- Executing [s@macro-outbound-callerid:2] NoOp("PJSIP/11-00000012", "") in new stack
-- Executing [s@macro-outbound-callerid:3] NoOp("PJSIP/11-00000012", "off") in new stack
-- Executing [s@macro-outbound-callerid:4] ExecIf("PJSIP/11-00000012", "0?Set(CALLERPRES(name-pres)=)") in new stack
-- Executing [s@macro-outbound-callerid:5] ExecIf("PJSIP/11-00000012", "0?Set(CALLERPRES(num-pres)=)") in new stack
-- Executing [s@macro-outbound-callerid:6] ExecIf("PJSIP/11-00000012", "0?Set(REALCALLERIDNUM=11)") in new stack
-- Executing [s@macro-outbound-callerid:7] ExecIf("PJSIP/11-00000012", "0?Set(AMPUSER=11)") in new stack
-- Executing [s@macro-outbound-callerid:8] GotoIf("PJSIP/11-00000012", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] Set("PJSIP/11-00000012", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:13] Set("PJSIP/11-00000012", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:14] Set("PJSIP/11-00000012", "TRUNKOUTCID=+495316157762") in new stack
-- Executing [s@macro-outbound-callerid:15] GotoIf("PJSIP/11-00000012", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,21)
-- Executing [s@macro-outbound-callerid:21] ExecIf("PJSIP/11-00000012", "1?Set(CALLERID(all)=+495316157762)") in new stack
-- Executing [s@macro-outbound-callerid:22] ExecIf("PJSIP/11-00000012", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:23] ExecIf("PJSIP/11-00000012", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:24] ExecIf("PJSIP/11-00000012", "0?Set(CALLERID(all)=11)") in new stack
-- Executing [s@macro-outbound-callerid:25] Set("PJSIP/11-00000012", "TIOHIDE=no") in new stack
-- Executing [s@macro-outbound-callerid:26] ExecIf("PJSIP/11-00000012", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:27] ExecIf("PJSIP/11-00000012", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:28] ExecIf("PJSIP/11-00000012", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:29] ExecIf("PJSIP/11-00000012", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:30] Set("PJSIP/11-00000012", "CDR(outbound_cnum)=+495316157762") in new stack
-- Executing [s@macro-outbound-callerid:31] Set("PJSIP/11-00000012", "CDR(outbound_cnam)=") in new stack
-- Executing [s@macro-dialout-trunk:14] GosubIf("PJSIP/11-00000012", "0?sub-flp-1,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:15] Set("PJSIP/11-00000012", "OUTNUM=05312504948") in new stack
-- Executing [s@macro-dialout-trunk:16] Set("PJSIP/11-00000012", "custom=PJSIP") in new stack
-- Executing [s@macro-dialout-trunk:17] ExecIf("PJSIP/11-00000012", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Ttr)") in new stack
-- Executing [s@macro-dialout-trunk:18] ExecIf("PJSIP/11-00000012", "0?Set(DIAL_TRUNK_OPTIONS=TtrM(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:19] Macro("PJSIP/11-00000012", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("PJSIP/11-00000012", "") in new stack
-- Executing [s@macro-dialout-trunk:20] GotoIf("PJSIP/11-00000012", "0?skipcrm") in new stack
-- Executing [s@macro-dialout-trunk:21] Set("PJSIP/11-00000012", "__CRM_DIRECTION=OUTBOUND") in new stack
-- Executing [s@macro-dialout-trunk:22] Set("PJSIP/11-00000012", "__CRM_DESTINATION=05312504948") in new stack
-- Executing [s@macro-dialout-trunk:23] Set("PJSIP/11-00000012", "__CRM_SOURCE=11") in new stack
-- Executing [s@macro-dialout-trunk:24] AGI("PJSIP/11-00000012", "agi://127.0.0.1/sangomacrm.agi") in new stack
-- <PJSIP/11-00000012>AGI Script agi://127.0.0.1/sangomacrm.agi completed, returning 0
-- Executing [s@macro-dialout-trunk:25] Set("PJSIP/11-00000012", "CHANNEL(hangup_handler_push)=crm-hangup,s,1") in new stack
-- Executing [s@macro-dialout-trunk:26] NoOp("PJSIP/11-00000012", "CRM Finished") in new stack
-- Executing [s@macro-dialout-trunk:27] GotoIf("PJSIP/11-00000012", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:28] ExecIf("PJSIP/11-00000012", "1?Set(CONNECTEDLINE(num,i)=05312504948)") in new stack
-- Executing [s@macro-dialout-trunk:29] GotoIf("PJSIP/11-00000012", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:30] ExecIf("PJSIP/11-00000012", "0?Set(DIAL_TRUNK_OPTIONS=tr)") in new stack
-- Executing [s@macro-dialout-trunk:31] Dial("PJSIP/11-00000012", "PJSIP/05312504948@Vodafone-SIP-Trunk,300,Ttrb(func-apply-sipheaders^s^1,(1))") in new stack
-- PJSIP/Vodafone-SIP-Trunk-00000013 Internal Gosub(func-apply-sipheaders,s,1(1)) start
-- Executing [s@func-apply-sipheaders:1] ExecIf("PJSIP/Vodafone-SIP-Trunk-00000013", "0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
-- Executing [s@func-apply-sipheaders:2] NoOp("PJSIP/Vodafone-SIP-Trunk-00000013", "Applying SIP Headers to channel PJSIP/Vodafone-SIP-Trunk-00000013") in new stack
-- Executing [s@func-apply-sipheaders:3] Set("PJSIP/Vodafone-SIP-Trunk-00000013", "TECH=PJSIP") in new stack
-- Executing [s@func-apply-sipheaders:4] Set("PJSIP/Vodafone-SIP-Trunk-00000013", "SIPHEADERKEYS=") in new stack
-- Executing [s@func-apply-sipheaders:5] While("PJSIP/Vodafone-SIP-Trunk-00000013", "0") in new stack
-- Jumping to priority 13
-- Executing [s@func-apply-sipheaders:14] Return("PJSIP/Vodafone-SIP-Trunk-00000013", "") in new stack
  == Spawn extension (from-pstn-pciheader, 05312504948, 1) exited non-zero on 'PJSIP/Vodafone-SIP-Trunk-00000013'
-- PJSIP/Vodafone-SIP-Trunk-00000013 Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
-- Called PJSIP/05312504948@Vodafone-SIP-Trunk
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:32] NoOp("PJSIP/11-00000012", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
-- Executing [s@macro-dialout-trunk:33] GotoIf("PJSIP/11-00000012", "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/11-00000012", "RC=21") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("PJSIP/11-00000012", "21,1") in new stack
-- Goto (macro-dialout-trunk,21,1)
-- Executing [21@macro-dialout-trunk:1] Goto("PJSIP/11-00000012", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("PJSIP/11-00000012", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] ExecIf("PJSIP/11-00000012", "1?Set(CALLERID(number)=11)") in new stack
-- Executing [05312504948@from-internal:7] Macro("PJSIP/11-00000012", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("PJSIP/11-00000012", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("PJSIP/11-00000012", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("PJSIP/11-00000012", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("PJSIP/11-00000012", "all-circuits-busy-now&please-try-call-later, noanswer") in new stack
   > 0x7fed98025df0 -- Strict RTP learning after remote address set to: 172.18.11.218:4098
-- <PJSIP/11-00000012> Playing 'all-circuits-busy-now.g722' (language 'de_DE')
   > 0x7fed98025df0 -- Strict RTP learning after remote address set to: 172.18.11.218:4098
   > 0x7fed98025df0 -- Strict RTP qualifying stream type: audio
   > 0x7fed98025df0 -- Strict RTP switching source address to 192.168.1.151:4098
-- <PJSIP/11-00000012> Playing 'please-try-call-later.g722' (language 'de_DE')
-- Executing [s@macro-outisbusy:5] Congestion("PJSIP/11-00000012", "20") in new stack
  == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'PJSIP/11-00000012' in macro 'outisbusy'
  == Spawn extension (from-internal, 05312504948, 7) exited non-zero on 'PJSIP/11-00000012'
-- Executing [h@from-internal:1] Macro("PJSIP/11-00000012", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/11-00000012", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/11-00000012", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] NoOp("PJSIP/11-00000012", " montior file= ") in new stack
-- Executing [s@macro-hangupcall:5] GotoIf("PJSIP/11-00000012", "1?skipagi") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] Hangup("PJSIP/11-00000012", "") in new stack
  == Spawn extension (macro-hangupcall, s, 7) exited non-zero on 'PJSIP/11-00000012' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/11-00000012'
-- PJSIP/11-00000012 Internal Gosub(crm-hangup,s,1) start
-- Executing [s@crm-hangup:1] NoOp("PJSIP/11-00000012", "Sending Hangup to CRM") in new stack
-- Executing [s@crm-hangup:2] NoOp("PJSIP/11-00000012", "HANGUP CAUSE: 34") in new stack
-- Executing [s@crm-hangup:3] ExecIf("PJSIP/11-00000012", "0?Set(__CRM_VOICEMAIL=)") in new stack
-- Executing [s@crm-hangup:4] NoOp("PJSIP/11-00000012", "MASTER CHANNEL: 1590306339.18 = 1590306339.18") in new stack
-- Executing [s@crm-hangup:5] GotoIf("PJSIP/11-00000012", "0?return") in new stack
-- Executing [s@crm-hangup:6] Set("PJSIP/11-00000012", "__CRM_HANGUP=1") in new stack
-- Executing [s@crm-hangup:7] AGI("PJSIP/11-00000012", "agi://127.0.0.1/sangomacrm.agi") in new stack
-- <PJSIP/11-00000012>AGI Script agi://127.0.0.1/sangomacrm.agi completed, returning 0
-- Executing [s@crm-hangup:8] Return("PJSIP/11-00000012", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/11-00000012'
-- PJSIP/11-00000012 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

Thanks for help
~Marlon

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Participants: 1

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@dckmedia wrote:

I have chan_mobile installed with FreePBX 15.0.16.49 and Asterisk 16.9.0. Im using iphone SE connected via Bluetooth adapter and can make and receive calls.

Problem is calls from "NO CALLER ID or Private number " to iphone, it will not ring my sip phones.

it this something to do with my iphone SE or is this something to do with my incoming dial plans.

root@pbx:~# asterisk -r
Asterisk 16.9.0, Copyright (C) 1999 - 2018, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for detail                                                                                        s.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Running as user 'asterisk'
Running under group 'asterisk'
Connected to Asterisk 16.9.0 currently running on pbx (pid = 1511)
[2020-05-24 17:58:55] DEBUG[3989]: chan_mobile.c:3926 do_monitor_phone: [IPHONE-SE] read +CIEV: 3,1
[2020-05-24 17:58:55] DEBUG[3989]: chan_mobile.c:3609 handle_response_ciev: [IPHONE-SE] incoming call, waiting for caller id
[2020-05-24 17:58:55] DEBUG[3985]: chan_mobile.c:1913 sco_accept: Incoming Audio Connection from device E0:C7:67:6A:D2:C1 MTU is 64
[2020-05-24 17:58:55] DEBUG[3985]: chan_mobile.c:1939 sco_accept: incoming audio connection for pvt without owner
[2020-05-24 17:58:57] DEBUG[1601]: res_pjsip_registrar.c:1279 check_expiration_thread: Woke up at 1590307137  Interval: 30
[2020-05-24 17:58:57] DEBUG[1601]: res_pjsip_registrar.c:1286 check_expiration_thread: Expiring 0 contacts
[2020-05-24 17:58:58] DEBUG[3989]: chan_mobile.c:3926 do_monitor_phone: [IPHONE-SE] read RING
[2020-05-24 17:58:58] DEBUG[3989]: chan_mobile.c:3688 handle_response_ring: [IPHONE-SE] got ring while waiting for caller id
[2020-05-24 17:59:01] DEBUG[3989]: chan_mobile.c:3926 do_monitor_phone: [IPHONE-SE] read RING
[2020-05-24 17:59:01] DEBUG[3989]: chan_mobile.c:3688 handle_response_ring: [IPHONE-SE] got ring while waiting for caller id
[2020-05-24 17:59:01] DEBUG[5953]: manager.c:6608 process_message: Running action 'Login'
[2020-05-24 17:59:02] DEBUG[5953]: manager.c:6608 process_message: Running action 'Command'
[2020-05-24 17:59:02] DEBUG[5953]: manager.c:6608 process_message: Running action 'Command'
[2020-05-24 17:59:02] DEBUG[5953]: manager.c:6608 process_message: Running action 'Command'
[2020-05-24 17:59:02] DEBUG[5953]: manager.c:6608 process_message: Running action 'Command'
[2020-05-24 17:59:02] DEBUG[5953]: manager.c:6608 process_message: Running action 'Command'
[2020-05-24 17:59:02] DEBUG[5953]: manager.c:6608 process_message: Running action 'Command'
[2020-05-24 17:59:02] DEBUG[5953]: manager.c:6608 process_message: Running action 'Command'
[2020-05-24 17:59:04] DEBUG[3989]: chan_mobile.c:3926 do_monitor_phone: [IPHONE-SE] read RING
[2020-05-24 17:59:04] DEBUG[3989]: chan_mobile.c:3688 handle_response_ring: [IPHONE-SE] got ring while waiting for caller id
[2020-05-24 17:59:07] DEBUG[3989]: chan_mobile.c:3926 do_monitor_phone: [IPHONE-SE] read RING
[2020-05-24 17:59:07] DEBUG[3989]: chan_mobile.c:3688 handle_response_ring: [IPHONE-SE] got ring while waiting for caller id
[2020-05-24 17:59:10] DEBUG[3989]: chan_mobile.c:3926 do_monitor_phone: [IPHONE-SE] read +CIEV: 3,0

Posts: 1

Participants: 1

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@StephanK wrote:

> [2020-05-23 08:01:26] WARNING[2546] chan_sip.c: sip_xmit of 0xb7407968 (len 627) to 159.65.251.173:31964 returned -1: Address family not supported by protocol

I have thousands of these in my logfiles. Any idea?

Posts: 3

Participants: 2

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@Prodna wrote:

Hi;

We dynamically add the extensions to the queue with “AMI”. We accept one call for each extension.
However, in a way that we cannot understand, sometimes the call drops as soon as it transfers to the call extension. Asteriks hangup cause comes as “16-Normal clearing”.
(Asterisk Version: 16.6.2)

What could be the reason for you?

queue.conf

[100001]
announce_frequency=0
announce_holdtime=no
announce_position=no
autofill=yes
autopause=no
autopausebusy=no
autopausedelay=0
autopauseunavail=no
joinempty=no
leavewhenempty=penalty,paused,invalid,unavailable,inuse,ringing
maxlen=0
memberdelay=0
min_announce_frequency=15
monitor_join=yes
musicclass=calmasesi
penaltymemberslimit=0
periodic_announce_frequency=0
queue_callswaiting=silence/1
queue_thereare=silence/1
queue_youarenext=silence/1
reportholdtime=no
retry=1
ringinuse=no
servicelevel=60
strategy=rrmemory
timeout=15
timeoutpriority=conf
timeoutrestart=no
weight=0
wrapuptime=5
context=

pjsip_endpoint.conf

[147] ;extension
type=endpoint
aors=147
auth=147-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=ulaw,alaw,g723,g729,g726,g722,gsm,ilbc,opus
context=RCOPOUT_A
callerid=7@admin2 <147>
dtmf_mode=rfc4733
aggregate_mwi=yes
use_avpf=yes
rtcp_mux=yes
bundle=no
ice_support=yes
media_use_received_transport=yes
trust_id_inbound=yes
send_connected_line=yes
media_encryption=dtls
timers=yes
media_encryption_optimistic=no
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en
one_touch_recording=on
record_on_feature=apprecord
record_off_feature=apprecord
media_encryption=dtls
dtls_verify=fingerprint
dtls_setup=actpass
dtls_rekey=0
dtls_cert_file=/etc/asterisk/keys/default.crt
dtls_private_key=/etc/asterisk/keys/default.key

cdr_logs

Time Event CNAM CNUM ANI DID AMA exten context App channel UserDefType EventExtra CEL Table
Sat, 23 May 2020 18:07 CHAN_START DEFAULT DIAL RCOUT_A Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 CHAN_START DEFAULT DIAL RCOUT_A Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 CHAN_START DEFAULT s from-pstn PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 ANSWER DIAL DEFAULT DIAL from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 ANSWER 49345***** DEFAULT DIAL RCOUT_A Dial Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 ANSWER DIAL 49345***** DEFAULT DIAL RCOUT_A AppDial2 Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 BRIDGE_ENTER DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 BRIDGE_ENTER 49345***** DEFAULT DIAL RCOUT_A Dial Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 APP_START DIAL 49345***** DEFAULT ANSWERED RCOUT_A Queue Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 CHAN_START 1@master3 4 DEFAULT s RCOPOUT_A PJSIP/4-0000006f
Sat, 23 May 2020 18:07 ANSWER 1@master3 4 4 DEFAULT ANSWERED RCOPOUT_A AppQueue PJSIP/4-0000006f
Sat, 23 May 2020 18:07 APP_START 1@master3 4 4 DEFAULT s RCOUT_A_KUYRUK_UYE MixMonitor PJSIP/4-0000006f
Sat, 23 May 2020 18:07 APP_END 1@master3 4 4 DEFAULT s RCOUT_A_KUYRUK_UYE MixMonitor PJSIP/4-0000006f
Sat, 23 May 2020 18:07 BRIDGE_ENTER 1@master3 4 4 DEFAULT ANSWERED RCOPOUT_A AppQueue PJSIP/4-0000006f
Sat, 23 May 2020 18:07 BRIDGE_ENTER DIAL 49345***** DEFAULT ANSWERED RCOUT_A Queue Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 BRIDGE_EXIT DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 BRIDGE_EXIT DIAL 49345***** DEFAULT ANSWERED RCOUT_A Queue Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 BRIDGE_ENTER DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 LOCAL_OPTIMIZE DIAL 49345***** DEFAULT ANSWERED RCOUT_A Queue Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 BRIDGE_EXIT 1@master3 4 DEFAULT DIAL RCOUT_A Dial Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 HANGUP 1@master3 4 DEFAULT DIAL RCOUT_A Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 CHAN_END 1@master3 4 DEFAULT DIAL RCOUT_A Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 APP_END DIAL 49345***** DEFAULT ANSWERED RCOUT_A Queue Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 HANGUP DIAL 49345***** DEFAULT ANSWERED RCOUT_A AppDial2 Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 CHAN_END DIAL 49345***** DEFAULT ANSWERED RCOUT_A AppDial2 Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 BRIDGE_EXIT DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 BRIDGE_EXIT 1@master3 4 4 DEFAULT ANSWERED RCOPOUT_A AppQueue PJSIP/4-0000006f
Sat, 23 May 2020 18:07 HANGUP 1@master3 4 4 DEFAULT ANSWERED RCOPOUT_A AppQueue PJSIP/4-0000006f
Sat, 23 May 2020 18:07 CHAN_END 1@master3 4 4 DEFAULT ANSWERED RCOPOUT_A AppQueue PJSIP/4-0000006f
Sat, 23 May 2020 18:07 HANGUP DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Time Event CNAM CNUM ANI DID AMA exten context App channel UserDefType EventExtra CEL Table
Sat, 23 May 2020 18:07 CHAN_END DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 LINKEDID_END DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Related Call Detail Records
Call Date Recording System CallerID Outbound CallerID DID App Destination Disposition Duration Userfield Account CDR Table CDR Graph
Sat, 23 May 2020 18:07 1.590.250.051.295 DIAL Return ANSWERED 00:00 89098
Sat, 23 May 2020 18:07 o DIAL Queue ANSWERED ANSWERED 00:11 89098
Sat, 23 May 2020 18:07 o 4 Dial DIAL ANSWERED 00:11 89098

asterisk_logs

<— Received SIP response (956 bytes) from UDP:87.238.XXX.XX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;branch=z9hG4bKPj39a93181-2955-47ac-85bb-76d174751ea2;rport=8000
Record-Route: sip:87.238.XXX.XX;lr;ep
Contact: sip:87.238.XXX.XX:5074
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13652 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Content-Type: application/sdp
Server: PortaSIP
Content-Length: 392

v=0
o=PortaSIP 2130776891631118155 1 IN IP4 87.238.XXX.XX
s=Phone Call via hiQ9200 SIPCA
t=0 0
m=audio 41492 RTP/AVP 8 0 18 101
c=IN IP4 87.238.XXX.XX
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:0 PCMU/8000
a=fmtp:0 vad=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sqn: 0
a=cdsc: 1 image udptl t38
a=sendrecv
a=ptime:20

-- PJSIP/t_1_14-0000006d answered Local/DIAL@RCOUT_A-0000005c;2
-- Local/DIAL@RCOUT_A-0000005c;1 answered
-- Executing [ANSWERED@RCOUT_A:1] Answer("Local/DIAL@RCOUT_A-0000005c;1", "") in new stack

<— Transmitting SIP request (450 bytes) to UDP:87.238.XXX.XX:5060 —>
ACK sip:87.238.XXX.XX:5074 SIP/2.0
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;rport;branch=z9hG4bKPj866061f5-366f-415d-9c15-6181c45325f3
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13652 ACK
Route: sip:87.238.XXX.XX;lr;ep
Max-Forwards: 70
User-Agent: FPBX-14.0.13.26(16.6.2)
Content-Length: 0

-- Executing [ANSWERED@RCOUT_A:2] Set("Local/DIAL@RCOUT_A-0000005c;1", "RCUID=o-100016-41-182520") in new stack
-- Executing [ANSWERED@RCOUT_A:3] Set("Local/DIAL@RCOUT_A-0000005c;1", "RCTIP=o") in new stack
-- Executing [ANSWERED@RCOUT_A:4] Set("Local/DIAL@RCOUT_A-0000005c;1", "RCKID=100016") in new stack
-- Executing [ANSWERED@RCOUT_A:5] Set("Local/DIAL@RCOUT_A-0000005c;1", "RCAID=182520") in new stack
-- Channel PJSIP/t_1_14-0000006d joined 'simple_bridge' basic-bridge <e1d70cb4-56b5-46b4-8cdf-03f2f6d5e42d>
-- Channel Local/DIAL@RCOUT_A-0000005c;2 joined 'simple_bridge' basic-bridge <e1d70cb4-56b5-46b4-8cdf-03f2f6d5e42d>
-- Executing [ANSWERED@RCOUT_A:6] Verbose("Local/DIAL@RCOUT_A-0000005c;1", "1, RCOUT 3 PJSIP/0001498454*****@t_1_14 - o-100016-41-182520 - Local/DIAL@RCOUT_A-0000005c;1 - 2020-05-23 18:07:42") in new stack

RCOUT 3 PJSIP/0001498454*****@t_1_14 - o-100016-41-182520 - Local/DIAL@RCOUT_A-0000005c;1 - 2020-05-23 18:07:42
– Executing [ANSWERED@RCOUT_A:7] Gosub(“Local/DIAL@RCOUT_A-0000005c;1”, “RCAMD,s,1(100016,182520,0)”) in new stack
– Executing [s@RCAMD:1] Answer(“Local/DIAL@RCOUT_A-0000005c;1”, “”) in new stack
– Executing [s@RCAMD:2] Verbose(“Local/DIAL@RCOUT_A-0000005c;1”, “1,AMD BASLADI 100016-182520-0”) in new stack
AMD BASLADI 100016-182520-0
– Executing [s@RCAMD:3] Set(“Local/DIAL@RCOUT_A-0000005c;1”, “DURUM=ANSWER”) in new stack
– Executing [s@RCAMD:4] GotoIf(“Local/DIAL@RCOUT_A-0000005c;1”, “1?son”) in new stack
– Goto (RCAMD,s,14)
– Executing [s@RCAMD:14] Return(“Local/DIAL@RCOUT_A-0000005c;1”, “ANSWER”) in new stack
– Executing [ANSWERED@RCOUT_A:8] Set(“Local/DIAL@RCOUT_A-0000005c;1”, “RCSTATU=ANSWER”) in new stack
– Executing [ANSWERED@RCOUT_A:9] Set(“Local/DIAL@RCOUT_A-0000005c;1”, “GELEN=1”) in new stack
– Executing [ANSWERED@RCOUT_A:10] GotoIf(“Local/DIAL@RCOUT_A-0000005c;1”, “1?atla”) in new stack
– Goto (RCOUT_A,ANSWERED,14)
– Executing [ANSWERED@RCOUT_A:14] NoOp(“Local/DIAL@RCOUT_A-0000005c;1”, “”) in new stack
– Executing [ANSWERED@RCOUT_A:15] GotoIf(“Local/DIAL@RCOUT_A-0000005c;1”, “0?kapat”) in new stack
– Executing [ANSWERED@RCOUT_A:16] Queue(“Local/DIAL@RCOUT_A-0000005c;1”, “100016,tn,10,RCOUT_A_KUYRUK_UYE”) in new stack
– Started music on hold, class ‘calmasesi’, on channel ‘Local/DIAL@RCOUT_A-0000005c;1’
– Called PJSIP/4
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
== DTLS ECDH initialized (automatic), faster PFS enabled
<— Transmitting SIP request (1692 bytes) to WSS:178.246.XXX.XX:18439 —>
INVITE sip:vujiu9og@178.246.XXX.XX:18439;transport=ws SIP/2.0
Via: SIP/2.0/WSS 85.95.XXX.XX:8002;rport;branch=z9hG4bKPjfdfe26c6-e26f-4198-a7ee-2b40b65a9d13;alias
From: sip:DIAL@pbxtest.xxxxx.com;tag=e2f7c171-1f8f-4a93-a8c2-628292ac491c
To: sip:vujiu9og@178.246.XXX.XX
Contact: sip:asterisk@pbxtest.xxxxx.com:5060;transport=ws
Call-ID: e87b6de1-05dc-45ee-8d4d-1d7da862a6d5
CSeq: 1327 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: sip:DIAL@pbxtest.xxxxx.com
Max-Forwards: 70
User-Agent: FPBX-14.0.13.26(16.6.2)
Content-Type: application/sdp
Content-Length: 927

v=0
o=- 1690486252 1690486252 IN IP4 85.95.XXX.XX
s=Asterisk
c=IN IP4 85.95.XXX.XX
t=0 0
m=audio 24360 UDP/TLS/RTP/SAVPF 0 8 4 18 111 9 3 97 107 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 49:42:49:29:BE:E1:65:AD:1B:80:7E:E1:75:5D:5F:E3:63:FC:8D:D7:32:D6:94:98:BC:4C:33:D0:7B:1D:36:0B
a=ice-ufrag:310c33e03aa30526508942d35514483e
a=ice-pwd:292f7726449b35512adb649f416e5aa0
a=candidate:Hbfbf9d57 1 UDP 2130706431 fe80::e050:1636:aa00:252e 24360 typ host
a=candidate:H555ff211 1 UDP 2130706431 85.95.XXX.XX 24360 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux

<— Received SIP response (370 bytes) from WSS:178.246.XXX.XX:18439 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 85.95.XXX.XX:8002;rport;branch=z9hG4bKPjfdfe26c6-e26f-4198-a7ee-2b40b65a9d13;alias
From: sip:DIAL@pbxtest.xxxxx.com;tag=e2f7c171-1f8f-4a93-a8c2-628292ac491c
To: sip:vujiu9og@178.246.XXX.XX
CSeq: 1327 INVITE
Call-ID: e87b6de1-05dc-45ee-8d4d-1d7da862a6d5
Supported: outbound
User-Agent: SIP.js/0.14.4
Content-Length: 0

<— Received SIP response (436 bytes) from WSS:178.246.XXX.XX:18439 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 85.95.XXX.XX:8002;rport;branch=z9hG4bKPjfdfe26c6-e26f-4198-a7ee-2b40b65a9d13;alias
From: sip:DIAL@pbxtest.xxxxx.com;tag=e2f7c171-1f8f-4a93-a8c2-628292ac491c
To: sip:vujiu9og@178.246.XXX.XX;tag=mlglunc7ma
CSeq: 1327 INVITE
Call-ID: e87b6de1-05dc-45ee-8d4d-1d7da862a6d5
Supported: outbound
User-Agent: SIP.js/0.14.4
Contact: sip:vujiu9og@192.0.X.XXX;transport=ws
Content-Length: 0

-- PJSIP/4-0000006f is ringing
-- PJSIP/4-0000006f is ringing

<— Received SIP response (1901 bytes) from WSS:178.246.XXX.XX:18439 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS 85.95.XXX.XX:8002;rport;branch=z9hG4bKPjfdfe26c6-e26f-4198-a7ee-2b40b65a9d13;alias
From: sip:DIAL@pbxtest.xxxxx.com;tag=e2f7c171-1f8f-4a93-a8c2-628292ac491c
To: sip:vujiu9og@178.246.XXX.XX;tag=mlglunc7ma
CSeq: 1327 INVITE
Call-ID: e87b6de1-05dc-45ee-8d4d-1d7da862a6d5
Supported: outbound
User-Agent: SIP.js/0.14.4
Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,SUBSCRIBE
Contact: sip:vujiu9og@192.0.X.XXX;transport=ws
Content-Type: application/sdp
Content-Length: 1356

v=0
o=- 4567787090042083750 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS 1gQ0FLUOf9GslA86Y7uBOZZ3ecACM0dkMbNW
m=audio 18512 UDP/TLS/RTP/SAVPF 0 8 9 107 101
c=IN IP4 178.246.XXX.XX
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1798856119 1 udp 2122260223 192.168.43.236 59130 typ host generation 0 network-id 1 network-cost 10
a=candidate:2813682212 1 udp 1686052607 178.246.XXX.XX 18512 typ srflx raddr 192.168.43.236 rport 59130 generation 0 network-id 1 network-cost 10
a=candidate:633053511 1 tcp 1518280447 192.168.43.236 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:weD2
a=ice-pwd:AJqFH6qGz+P6Yrdo45YWg6GB
a=ice-options:trickle
a=fingerprint:sha-256 3B:8D:ED:F8:C7:56:E8:95:F9:3D:78:5B:15:F2:40:19:37:9F:21:B7:7D:2E:6C:99:34:6D:9D:52:D2:B9:5B:84
a=setup:active
a=mid:0
a=sendrecv
a=msid:1gQ0FLUOf9GslA86Y7uBOZZ3ecACM0dkMbNW dd127927-5900-4267-aef5-b85b92d1d8cd
a=rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 minptime=10;useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=ssrc:397357961 cname:eq6+5jnRwp9EfZ6s
a=ssrc:397357961 msid:1gQ0FLUOf9GslA86Y7uBOZZ3ecACM0dkMbNW dd127927-5900-4267-aef5-b85b92d1d8cd
a=ssrc:397357961 mslabel:1gQ0FLUOf9GslA86Y7uBOZZ3ecACM0dkMbNW
a=ssrc:397357961 label:dd127927-5900-4267-aef5-b85b92d1d8cd

<— Transmitting SIP request (430 bytes) to WSS:178.246.XXX.XX:18439 —>
ACK sip:vujiu9og@178.246.XXX.XX:18439;transport=ws SIP/2.0
Via: SIP/2.0/WSS 85.95.XXX.XX:8002;rport;branch=z9hG4bKPj531dc48c-82c9-4c77-9e3b-a4a8c5bcb073;alias
From: sip:DIAL@pbxtest.xxxxx.com;tag=e2f7c171-1f8f-4a93-a8c2-628292ac491c
To: sip:vujiu9og@178.246.XXX.XX;tag=mlglunc7ma
Call-ID: e87b6de1-05dc-45ee-8d4d-1d7da862a6d5
CSeq: 1327 ACK
Max-Forwards: 70
User-Agent: FPBX-14.0.13.26(16.6.2)
Content-Length: 0

-- PJSIP/4-0000006f answered Local/DIAL@RCOUT_A-0000005c;1
-- Stopped music on hold on Local/DIAL@RCOUT_A-0000005c;1
-- PJSIP/4-0000006f Internal Gosub(RCOUT_A_KUYRUK_UYE,s,1) start
-- Executing [s@RCOUT_A_KUYRUK_UYE:1] NoOp("PJSIP/4-0000006f", "") in new stack
-- Executing [s@RCOUT_A_KUYRUK_UYE:2] Verbose("PJSIP/4-0000006f", "1, RCOUT 6  - o-100016-41-182520 - PJSIP/4-0000006f - RCOUT_A_KUYRUK_UYE - PJSIP/4-0000006f - s") in new stack

RCOUT 6 - o-100016-41-182520 - PJSIP/4-0000006f - RCOUT_A_KUYRUK_UYE - PJSIP/4-0000006f - s
– Executing [s@RCOUT_A_KUYRUK_UYE:3] Set(“PJSIP/4-0000006f”, “RCUID=o-100016-41-182520”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:4] Set(“PJSIP/4-0000006f”, “RCKID=100016”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:5] Set(“PJSIP/4-0000006f”, “RCDTID=41”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:6] Set(“PJSIP/4-0000006f”, “RCAID=182520”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:7] Set(“PJSIP/4-0000006f”, “RECDOSYA=/var/spool/asterisk/recording/182520-”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:8] Set(“PJSIP/4-0000006f”, “GELEN=1”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:9] GotoIf(“PJSIP/4-0000006f”, “1?atla”) in new stack
– Goto (RCOUT_A_KUYRUK_UYE,s,13)
– Executing [s@RCOUT_A_KUYRUK_UYE:13] NoOp(“PJSIP/4-0000006f”, “”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:14] UserEvent(“PJSIP/4-0000006f”, “RCDialBegin_Op,RCAID:182520,RCDTID:41,RCUYE:4,RCDNID:89098”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:15] MixMonitor(“PJSIP/4-0000006f”, “/var/spool/asterisk/recording/182520-sistem.wav,a”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:16] Return(“PJSIP/4-0000006f”, “”) in new stack
== Begin MixMonitor Recording PJSIP/4-0000006f
== Spawn extension (RCOPOUT_A, ANSWERED, 1) exited non-zero on ‘PJSIP/4-0000006f’
– PJSIP/4-0000006f Internal Gosub(RCOUT_A_KUYRUK_UYE,s,1) complete GOSUB_RETVAL=
– Channel PJSIP/4-0000006f joined ‘simple_bridge’ basic-bridge <0c7ae476-3a83-4f55-8d77-d7271181acca>
– Channel Local/DIAL@RCOUT_A-0000005c;1 joined ‘simple_bridge’ basic-bridge <0c7ae476-3a83-4f55-8d77-d7271181acca>
– Channel PJSIP/t_1_14-0000006d left ‘simple_bridge’ basic-bridge
– Channel Local/DIAL@RCOUT_A-0000005c;1 left ‘simple_bridge’ basic-bridge <0c7ae476-3a83-4f55-8d77-d7271181acca>
– Channel PJSIP/t_1_14-0000006d swapped with Local/DIAL@RCOUT_A-0000005c;1 into ‘simple_bridge’ basic-bridge <0c7ae476-3a83-4f55-8d77-d7271181acca>
– Channel Local/DIAL@RCOUT_A-0000005c;2 left ‘simple_bridge’ basic-bridge
== Spawn extension (RCOUT_A, DIAL, 8) exited non-zero on ‘Local/DIAL@RCOUT_A-0000005c;2’
<— Transmitting SIP request (1035 bytes) to UDP:87.238.XXX.XX:5060 —>
INVITE sip:87.238.XXX.XX:5074 SIP/2.0
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;rport;branch=z9hG4bKPj17aae872-b13f-4b1e-ab0c-9f04657c2d91
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
Contact: sip:asterisk@85.95.XXX.XX:8000
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13653 INVITE
Route: sip:87.238.XXX.XX;lr;ep
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-14.0.13.26(16.6.2)
Content-Type: application/sdp
Content-Length: 308

v=0
o=- 1908267525 1908267526 IN IP4 85.95.XXX.XX
s=Asterisk
c=IN IP4 85.95.XXX.XX
t=0 0
m=audio 23890 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

== Spawn extension (RCOUT_A, ANSWERED, 16) exited non-zero on ‘Local/DIAL@RCOUT_A-0000005c;1’
<— Received SIP response (352 bytes) from UDP:87.238.XXX.XX:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;rport=8000;branch=z9hG4bKPj17aae872-b13f-4b1e-ab0c-9f04657c2d91
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13653 INVITE
Content-Length: 0

<— Received SIP response (915 bytes) from UDP:87.238.XXX.XX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;branch=z9hG4bKPj17aae872-b13f-4b1e-ab0c-9f04657c2d91;rport=8000
Contact: sip:87.238.XXX.XX:5074
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13653 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Content-Type: application/sdp
Server: PortaSIP
Content-Length: 392

v=0
o=PortaSIP 2130776891631118155 2 IN IP4 87.238.XXX.XX
s=Phone Call via hiQ9200 SIPCA
t=0 0
m=audio 41492 RTP/AVP 0 8 18 101
c=IN IP4 87.238.XXX.XX
a=rtpmap:0 PCMU/8000
a=fmtp:0 vad=no
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sqn: 0
a=cdsc: 1 image udptl t38
a=sendrecv
a=ptime:20

<— Transmitting SIP request (450 bytes) to UDP:87.238.XXX.XX:5060 —>
ACK sip:87.238.XXX.XX:5074 SIP/2.0
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;rport;branch=z9hG4bKPj12e9a84e-da11-4215-87e6-8ab565bdf41b
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13653 ACK
Route: sip:87.238.XXX.XX;lr;ep
Max-Forwards: 70
User-Agent: FPBX-14.0.13.26(16.6.2)
Content-Length: 0

<— Received SIP request (731 bytes) from UDP:87.238.XXX.XX:5060 —>
BYE sip:asterisk@85.95.XXX.XX:8000 SIP/2.0
Via: SIP/2.0/UDP 87.238.XXX.XX:5060;branch=z9hG4bK-524287-1—683e14e478b76ff31034c2a7e9ee6db3;rport
Via: SIP/2.0/UDP 87.238.XXX.XX:5074;branch=z9hG4bK-ga4exivxulbugknp;rport=5074
Max-Forwards: 69
Contact: sip:87.238.XXX.XX:5074
To: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
From: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 555 BYE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
User-Agent: PortaSIP
h323-conf-id: 1503134234-742797846-1530232655-1293871401
cisco-GUID: 1503134234-742797846-1530232655-1293871401
Content-Length: 0

<— Transmitting SIP response (487 bytes) to UDP:87.238.XXX.XX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.238.XXX.XX:5060;rport=5060;received=87.238.XXX.XX;branch=z9hG4bK-524287-1—683e14e478b76ff31034c2a7e9ee6db3
Via: SIP/2.0/UDP 87.238.XXX.XX:5074;rport=5074;branch=z9hG4bK-ga4exivxulbugknp
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
From: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
To: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
CSeq: 555 BYE
Server: FPBX-14.0.13.26(16.6.2)
Content-Length: 0

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@StephanK wrote:

Is there a way to add a file to the backup with FreePBX 15?

It was possible earlier, but I can’t find anything on 15.

I want to add my postfix main.conf and extension_custom.conf.

Thanks!

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@lwintour wrote:

Hi there,
My Polycom phone will register to the server, but is unable to make or receive calls. My bria soft phone registers and can make an receive call happily, but the Polycom wont. Can someone please help me?

Please note: I am in the AEST (Australian Eastern Standard Time) timezone, so I may be a few hours to respond to any posts.

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@kfeen wrote:

Ok the title might not be the best but let me explain. Right now I have one server at my office that’s running one instance of FreePBX that only one remote client needs to connect to. All I do is have ports 5060 and 10000-20000 forwarded to the LAN IP that is assigned to that server, the clients phones point to my WAN and it works great. All of my other clients have the servers at their offices so no real networking config is required.

In the very near future I plan on moving all of my clients FreePBX instances into my office for various reasons. I will be using Proxmox to run VMs that contain each FreePBX instance. My question is how would I direct each one of my clients to the LAN IP address that is assigned to their FreePBX instance if all I have is one WAN? I’m using a network bridge to connect Proxmox’s internal LAN to the rest of my network. I’m not that great at networking so the help is much appreciated.

I know that this is more of a networking question rather than a FreePBX question but I thought this would be the best place to get a direct answer.

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@nickjenkey wrote:

Today I went to the admin page to modify an inbound routes settings and in the connectivity dropdown the inbound route and outbound route options are not there.

I restarted the system, ran a system update and still no menu items. The phones still work on our trunk including the priority inbound route I set up for my cellphone.

All the other menu Items in other tabs seem to be there

Any Ideas?

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@WB3FFV wrote:

I am running a current PBXact server with about 500 active extensions, which most are connected using PJsip. It’s running asterisk 16.9.0 for the release, and all modules are current. I am seeing the following errors in the logging, and I am wondering if something needs to be adjusted up for the extension count we have. It’s on a pretty rocking server, so not a problem at all to allow it to use more of the resources on the server.

Here are the errors I am seeing:

[2020-05-26 16:48:57] WARNING[26365][C-00000569] taskprocessor.c: The ‘stasis/m:cache_pattern:1/channel:all-00000f52’ task processor queue reached 500 scheduled tasks again.
[2020-05-26 16:48:57] WARNING[26367][C-00000569] taskprocessor.c: The ‘stasis/m:cache_pattern:1/channel:all-00000f52’ task processor queue reached 500 scheduled tasks again.
[2020-05-26 16:52:20] WARNING[29631][C-00000572] taskprocessor.c: The ‘stasis/m:cache_pattern:1/channel:all-00000f52’ task processor queue reached 500 scheduled tasks again.
[2020-05-26 17:02:05] WARNING[4350][C-00000580] taskprocessor.c: The ‘stasis/m:cache_pattern:1/channel:all-00000f52’ task processor queue reached 500 scheduled tasks again.
[2020-05-26 17:08:53] WARNING[9307][C-0000058e] taskprocessor.c: The ‘stasis/p:channel:all-000028ef’ task processor queue reached 500 scheduled tasks.
[2020-05-26 17:15:20] WARNING[13284][C-00000598] taskprocessor.c: The ‘stasis/m:cache_pattern:1/channel:all-00000f52’ task processor queue reached 500 scheduled tasks again.
[2020-05-26 17:23:52] WARNING[15986][C-000005a3] taskprocessor.c: The ‘stasis/m:cache_pattern:1/channel:all-00000f52’ task processor queue reached 500 scheduled tasks again.
[2020-05-26 17:32:49] WARNING[17784][C-000005a9] taskprocessor.c: The ‘stasis/p:channel:all-00002923’ task processor queue reached 500 scheduled tasks.
[2020-05-26 17:32:49] WARNING[17784][C-000005a9] taskprocessor.c: The ‘stasis/m:cache_pattern:1/channel:all-00000f52’ task processor queue reached 500 scheduled tasks again.
[2020-05-26 17:39:21] WARNING[20522][C-000005af] taskprocessor.c: The ‘stasis/m:channel:all-00000f53’ task processor queue reached 500 scheduled tasks again.
[2020-05-26 17:40:21] WARNING[18975][C-000005af] taskprocessor.c: The ‘stasis/m:cache_pattern:1/channel:all-00000f52’ task processor queue reached 500 scheduled tasks again.

If anyone can offer any clues, or point me in the right direction it would be great…

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@chrischevy wrote:

I have a question that is more network related than FreePBX related.

One of our clients has a router that is provided and managed by the “mothership”. They have no access to it and the cannot ask for any modifications (very tedious everytime). However, the company policy allows them to have a second router with a second internet connection. They want to use this second connection for VPN access and to connect outside IP phones (the FreeePBX is local)

I’m pretty sure I can setup something for them, but I have a question regarding the “gateway”.

Let’s say I have a second Internet connection with a fixed IP, configured in the second router. This router has a LAN interface connected in the LAN network. I know that all traffic originating from the devices on the LAN will go to their default gateway, which is the “locked down” main router. BUT, if I have traffic incoming in the second Internet connection, through the second router (like a VPN connection or a SIP phone registration, through port forwarding), once this traffic reaches it’s destination on the LAN, will the “replying” traffic go back the way it came to the second router or will it be directed to the default gateway (main router), thus breaking everything ? I’m pretty sure that the traffic will go back the same way it came in, but I need to be sure and I cannot really test it for now.
Note: Adding a route to the main router to redirect some traffic to the second router is not possible.

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@FPBX135 wrote:

Hello, I have a few cisco 7941s and 7942s set up on FreePBX 15.0.16.52. The phones are connected to the server. I can access the voicemail, *98 using the voicemail button on the phone, perfectly fine. But if I try to enter any other Feature Codes or extension numbers, I just get a tone and it doesn’t work. Even when I type *98 I just get the tone. I only know a little about FreePBX so I don’t know why this wouldn’t work. What should I do?
Thank you!

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@f1systems wrote:

I was adding and removing users and extensions and now lost outbound calling. This has not been my day.

https://pastebin.freepbx.org/view/8128d018#L208

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@Pentium5 wrote:

Hi all,
I have a fresh Asterisk 16 + FreePBX 15 install on a CentOS 8.1 (not a distro).
The issue is that dashboard is loading too long, getting stuck at 75%:

image.png1525×276 17.2 KB

When timeout expires (50 s), it finally loads, but with empty section “System overview”:

image.png1748×341 37.4 KB

Apache logs contains the following lines:

[Thu May 14 16:41:52.110192 2020] [proxy_fcgi:error] [pid 8643:tid 140221907703552] (70007)The timeout specified has expired: [client 10.33.3.248:54675] AH01075: Error dispatching request to : (polling), referer: http://10.10.1.149/admin/config.php
[Thu May 14 17:04:12.601101 2020] [proxy_fcgi:error] [pid 8643:tid 140222033770240] (70007)The timeout specified has expired: [client 10.33.3.248:56184] AH01075: Error dispatching request to : (polling), referer: http://10.10.1.149/admin/config.php?display=index

PHP version:

[root@hostname ~]# php -v
PHP 7.2.11 (cli) (built: Oct  9 2018 15:09:36) ( NTS )
Copyright (c) 1997-2018 The PHP Group
Zend Engine v3.2.0, Copyright (c) 1998-2018 Zend Technologies

Based on similar topics and my observations, I tried the following actions, but none of them helped:

• Disable RSS feeds
• Upgrade dashboard module (its current version is 15.0.5)
• Provide correct hostname in /etc/hosts

Apart from this issue, the system seem to be operating properly; provisioning to Asterisk also works.
Could you please guide me what else should I check / correct?

Thanks.

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@JessicaRabbit wrote:

FreePBX 15.0.16.52

Audio lost after 45 minutes. Has happened a number of times. Users loosing confidence.
Strange looking log. Following call being established then 45 minutes later next entries look to me like a new call arriving but user doesn’t remember any call, user just hung up. Am not very good with logs.

Large log. Can someone look at log? How do I submit?

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@Spaze wrote:

I am mighrating from bare Asterisk to FreePBX 15 (Asterisk 16.6.2 and the UI firmware 12.7.6-2002-2.sng7). I successfully imported about 1400 extensions and added ~180 queues with static agents and noticed that opening a queue/extension settings takes long time - about 15-20 seconds.
Checking the Asterisk CLI, I found that the Web UI is pushing several thousands manager commands to retrieve all the extensions, queues and other user settings from Asterisk DB, causing the web UI stuck and wait for the database poll to be done.
Does anybody know how to disable that? I am thinking about rolling back to 14th version which has the same number of users but is working fine.

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@posi211 wrote:

Did anyone see the new website? FreePBX.org

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@josephchrz wrote:

Hello i don’t remember what port i set to get into the web interface from my freepbx. i have changed it from the standard port 80 to something else. I have tired to look all over the internet for something i can look in the command line of putty. But i can not find nothing. Can someone please help me to figure out where i can find it at please?

Joseph

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