A post was split to a new topic: Whoops error after install
White Page on http:///admin/config.php
Voice mail not recognizing touch tones
Yeah - i kind of figured that. I checked device settings in advanced settings and SIP DTMF is set to rfc2833. I also tried the other options with no luck. I am not sure if there is somewhere else to check or how to go about troubleshooting it.
Keith
Softphones, External Clients, VPN, Etc
1.If we want our users (less than 10) to be able to log into softphone software at home or on mobile phones, can this be done with just setting up firewall/routing rules or is VPN required?
Could you? It depends on if your server is publicly accessible or not. If the phones can reach the PBX directly or via routing from a public address then you wouldn't need VPN, if your PBX is only accessible privately, then you probably would need a VPN to get your phones on a network that can reach the PBX. I would recommend consulting your network/security engineer to determine what you should do.
2.Should we allow users to be able to log into their own desk extension with softphone? And are there any problems with logging into a softphone with the same extension as a user's desk phone? As in, multiple clients on the same extension... Or, do we need to get into setting up separate extensions for softphone use. The problem I see with that is the voicemail is attached to the extension and not the user. For ease, the extension passwords would be easier to remember and not the auto generated Secret.
To enable login in to multiple endpoints, edit the extension and increase the number of contacts. My recommendation would be to test the setup to ensure it meets all of your workflow and security specifications.
3.If we want Sangoma phones to be installed off-site (homes), should we get the FreePBX VPN add-on or can this work with router VPN?
You should be able to do it with either option. The best approach depends on your network topology and resource requirements. A collaboration with your network/security engineer to determine what you should do.
Multiple random incoming calls
I just set up my inbound route and I keep getting call after call the caller ID shows ext not even on my ext range. below is a snippet of a call I just received.
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [89011441873877602@from-sip-external:1] NoOp("SIP/71.223.91.149-000001c4", "Received incoming SIP connection from unknown peer to 89011441873877602") in new stack
-- Executing [89011441873877602@from-sip-external:2] Set("SIP/71.223.91.149-000001c4", "DID=89011441873877602") in new stack
-- Executing [89011441873877602@from-sip-external:3] Goto("SIP/71.223.91.149-000001c4", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/71.223.91.149-000001c4", "1?setlanguage:checkanon") in new stack
-- Goto (from-sip-external,s,2)
-- Executing [s@from-sip-external:2] Set("SIP/71.223.91.149-000001c4", "CHANNEL(language)=en") in new stack
-- Executing [s@from-sip-external:3] GotoIf("SIP/71.223.91.149-000001c4", "0?noanonymous") in new stack
-- Executing [s@from-sip-external:4] Goto("SIP/71.223.91.149-000001c4", "from-trunk,89011441873877602,1") in new stack
-- Goto (from-trunk,89011441873877602,1)
-- Executing [89011441873877602@from-trunk:1] NoOp("SIP/71.223.91.149-000001c4", "Catch-All DID Match - Found 89011441873877602 - You probably want a DID for this.") in new stack
-- Executing [89011441873877602@from-trunk:2] Log("SIP/71.223.91.149-000001c4", "WARNING,Friendly Scanner from 64.91.235.188") in new stack
[2017-10-09 09:00:20] WARNING[108312][C-000000e8]: Ext. 89011441873877602:2 @ from-trunk: Friendly Scanner from 64.91.235.188
-- Executing [89011441873877602@from-trunk:3] Set("SIP/71.223.91.149-000001c4", "_FROMDID=89011441873877602") in new stack
-- Executing [89011441873877602@from-trunk:4] Goto("SIP/71.223.91.149-000001c4", "ext-did,s,1") in new stack
-- Goto (ext-did,s,1)
-- Executing [s@ext-did:1] Set("SIP/71.223.91.149-000001c4", "__DIRECTION=INBOUND") in new stack
-- Executing [s@ext-did:2] Gosub("SIP/71.223.91.149-000001c4", "sub-record-check,s,1(in,s,dontcare)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("SIP/71.223.91.149-000001c4", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("SIP/71.223.91.149-000001c4", "_RECSTATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("SIP/71.223.91.149-000001c4", "NOW=1507564820") in new stack
-- Executing [s@sub-record-check:4] Set("SIP/71.223.91.149-000001c4", "__DAY=09") in new stack
-- Executing [s@sub-record-check:5] Set("SIP/71.223.91.149-000001c4", "__MONTH=10") in new stack
-- Executing [s@sub-record-check:6] Set("SIP/71.223.91.149-000001c4", "__YEAR=2017") in new stack
-- Executing [s@sub-record-check:7] Set("SIP/71.223.91.149-000001c4", "__TIMESTR=20171009-090020") in new stack
-- Executing [s@sub-record-check:8] Set("SIP/71.223.91.149-000001c4", "__FROMEXTEN=unknown") in new stack
-- Executing [s@sub-record-check:9] Set("SIP/71.223.91.149-000001c4", "_MONFMT=wav") in new stack
-- Executing [s@sub-record-check:10] NoOp("SIP/71.223.91.149-000001c4", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("SIP/71.223.91.149-000001c4", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("SIP/71.223.91.149-000001c4", "2?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("SIP/71.223.91.149-000001c4", "1?sub-record-check,in,1") in new stack
-- Goto (sub-record-check,in,1)
-- Executing [in@sub-record-check:1] NoOp("SIP/71.223.91.149-000001c4", "Inbound Recording Check to s") in new stack
-- Executing [in@sub-record-check:2] Set("SIP/71.223.91.149-000001c4", "FROMEXTEN=unknown") in new stack
-- Executing [in@sub-record-check:3] ExecIf("SIP/71.223.91.149-000001c4", "5?Set(FROMEXTEN=10001)") in new stack
-- Executing [in@sub-record-check:4] Gosub("SIP/71.223.91.149-000001c4", "recordcheck,1(dontcare,in,s)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("SIP/71.223.91.149-000001c4", "Starting recording check against dontcare") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("SIP/71.223.91.149-000001c4", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [recordcheck@sub-record-check:3] Return("SIP/71.223.91.149-000001c4", "") in new stack
-- Executing [in@sub-record-check:5] Return("SIP/71.223.91.149-000001c4", "") in new stack
-- Executing [s@ext-did:3] Gosub("SIP/71.223.91.149-000001c4", "app-blacklist-check,s,1()") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/71.223.91.149-000001c4", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("SIP/71.223.91.149-000001c4", "CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/71.223.91.149-000001c4", "") in new stack
-- Executing [s@ext-did:4] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(_FROMDID=s)") in new stack
-- Executing [s@ext-did:5] Set("SIP/71.223.91.149-000001c4", "CDR(did)=89011441873877602") in new stack
-- Executing [s@ext-did:6] ExecIf("SIP/71.223.91.149-000001c4", "0 ?Set(CALLERID(name)=10001)") in new stack
-- Executing [s@ext-did:7] Set("SIP/71.223.91.149-000001c4", "__MOHCLASS=") in new stack
-- Executing [s@ext-did:8] Set("SIP/71.223.91.149-000001c4", "_REVERSALREJECT=FALSE") in new stack
-- Executing [s@ext-did:9] GotoIf("SIP/71.223.91.149-000001c4", "1?post-reverse-charge") in new stack
-- Goto (ext-did,s,11)
-- Executing [s@ext-did:11] NoOp("SIP/71.223.91.149-000001c4", "") in new stack
-- Executing [s@ext-did:12] Set("SIP/71.223.91.149-000001c4", "_CALLINGNAMEPRESSV=allowed_not_screened") in new stack
-- Executing [s@ext-did:13] Set("SIP/71.223.91.149-000001c4", "_CALLINGNUMPRESSV=allowed_not_screened") in new stack
-- Executing [s@ext-did:14] Set("SIP/71.223.91.149-000001c4", "CALLERID(name-pres)=allowed_not_screened") in new stack
-- Executing [s@ext-did:15] Set("SIP/71.223.91.149-000001c4", "CALLERID(num-pres)=allowed_not_screened") in new stack
-- Executing [s@ext-did:16] NoOp("SIP/71.223.91.149-000001c4", "CallerID Entry Point") in new stack
-- Executing [s@ext-did:17] Set("SIP/71.223.91.149-000001c4", "_CRMDIRECTION=INBOUND") in new stack
-- Executing [s@ext-did:18] Set("SIP/71.223.91.149-000001c4", "_CRMSOURCE=10001") in new stack
-- Executing [s@ext-did:19] Set("SIP/71.223.91.149-000001c4", "_CRMLINKEDID=1507564820.452") in new stack
-- Executing [s@ext-did:20] ExecIf("SIP/71.223.91.149-000001c4", "1?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
-- Executing [s@ext-did:21] Goto("SIP/71.223.91.149-000001c4", "from-did-direct,6000,1") in new stack
-- Goto (from-did-direct,6000,1)
-- Executing [6000@from-did-direct:1] GotoIf("SIP/71.223.91.149-000001c4", "1?ext-local,6000,1:followme-check,6000,1") in new stack
-- Goto (ext-local,6000,1)
-- Executing [6000@ext-local:1] Set("SIP/71.223.91.149-000001c4", "__RINGTIMER=15") in new stack
-- Executing [6000@ext-local:2] Macro("SIP/71.223.91.149-000001c4", "exten-vm,6000,6000,0,0,0") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/71.223.91.149-000001c4", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/71.223.91.149-000001c4", "TOUCH_MONITOR=1507564820.452") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/71.223.91.149-000001c4", "AMPUSER=10001") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/71.223.91.149-000001c4", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/71.223.91.149-000001c4", "1?Set(REALCALLERIDNUM=10001)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/71.223.91.149-000001c4", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/71.223.91.149-000001c4", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/71.223.91.149-000001c4", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/71.223.91.149-000001c4", "1?report") in new stack
-- Goto (macro-user-callerid,s,15)
-- Executing [s@macro-user-callerid:15] GotoIf("SIP/71.223.91.149-000001c4", "0?continue") in new stack
-- Executing [s@macro-user-callerid:16] ExecIf("SIP/71.223.91.149-000001c4", "1?Set(_CALLEEACCOUNCODE=)") in new stack
-- Executing [s@macro-user-callerid:17] Set("SIP/71.223.91.149-000001c4", "__TTL=6") in new stack
-- Executing [s@macro-user-callerid:18] GotoIf("SIP/71.223.91.149-000001c4", "1?continue") in new stack
-- Goto (macro-user-callerid,s,29)
-- Executing [s@macro-user-callerid:29] Set("SIP/71.223.91.149-000001c4", "CALLERID(number)=10001") in new stack
-- Executing [s@macro-user-callerid:30] Set("SIP/71.223.91.149-000001c4", "CALLERID(name)=10001") in new stack
-- Executing [s@macro-user-callerid:31] GotoIf("SIP/71.223.91.149-000001c4", "0?cnum") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/71.223.91.149-000001c4", "CDR(cnam)=10001") in new stack
-- Executing [s@macro-user-callerid:33] Set("SIP/71.223.91.149-000001c4", "CDR(cnum)=10001") in new stack
-- Executing [s@macro-user-callerid:34] Set("SIP/71.223.91.149-000001c4", "CHANNEL(language)=en") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/71.223.91.149-000001c4", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/71.223.91.149-000001c4", "__EXTTOCALL=6000") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/71.223.91.149-000001c4", "__PICKUPMARK=6000") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/71.223.91.149-000001c4", "RT=15") in new stack
[2017-10-09 09:00:20] ERROR[108312][C-000000e8]: pbx.c:4425 ast_func_read: Function PJSIP_HEADER not registered
-- Executing [s@macro-exten-vm:6] ExecIf("SIP/71.223.91.149-000001c4", "0?Macro(vm,6000,DIRECTDIAL,)") in new stack
[2017-10-09 09:00:20] ERROR[108312][C-000000e8]: pbx.c:4464 ast_func_read2: Function PJSIP_HEADER not registered
[2017-10-09 09:00:20] ERROR[108312][C-000000e8]: pbx.c:4425 ast_func_read: Function PJSIP_HEADER not registered
-- Executing [s@macro-exten-vm:7] ExecIf("SIP/71.223.91.149-000001c4", "0?MacroExit()") in new stack
[2017-10-09 09:00:20] ERROR[108312][C-000000e8]: pbx.c:4464 ast_func_read2: Function PJSIP_HEADER not registered
[2017-10-09 09:00:20] ERROR[108312][C-000000e8]: pbx.c:4425 ast_func_read: Function PJSIP_HEADER not registered
-- Executing [s@macro-exten-vm:8] ExecIf("SIP/71.223.91.149-000001c4", "0?Gosub(ext-intercom,*806000,1())") in new stack
[2017-10-09 09:00:20] ERROR[108312][C-000000e8]: pbx.c:4464 ast_func_read2: Function PJSIP_HEADER not registered
[2017-10-09 09:00:20] ERROR[108312][C-000000e8]: pbx.c:4425 ast_func_read: Function PJSIP_HEADER not registered
-- Executing [s@macro-exten-vm:9] ExecIf("SIP/71.223.91.149-000001c4", "0?MacroExit()") in new stack
[2017-10-09 09:00:20] ERROR[108312][C-000000e8]: pbx.c:4464 ast_func_read2: Function PJSIP_HEADER not registered
[2017-10-09 09:00:20] ERROR[108312][C-000000e8]: pbx.c:4425 ast_func_read: Function PJSIP_HEADER not registered
[2017-10-09 09:00:20] WARNING[108312][C-000000e8]: pbx.c:4277 func_args: Can't find trailing parenthesis for function 'DB(DEVICE/6000/dial'?
-- Executing [s@macro-exten-vm:10] ExecIf("SIP/71.223.91.149-000001c4", "0?ChanSpy(SIP/6000,q)") in new stack
[2017-10-09 09:00:20] ERROR[108312][C-000000e8]: pbx.c:4464 ast_func_read2: Function PJSIP_HEADER not registered
[2017-10-09 09:00:20] WARNING[108312][C-000000e8]: pbx.c:4277 func_args: Can't find trailing parenthesis for function 'DB(DEVICE/6000/dial'?
[2017-10-09 09:00:20] ERROR[108312][C-000000e8]: pbx.c:4425 ast_func_read: Function PJSIP_HEADER not registered
-- Executing [s@macro-exten-vm:11] ExecIf("SIP/71.223.91.149-000001c4", "0?MacroExit()") in new stack
[2017-10-09 09:00:20] ERROR[108312][C-000000e8]: pbx.c:4464 ast_func_read2: Function PJSIP_HEADER not registered
-- Executing [s@macro-exten-vm:12] Gosub("SIP/71.223.91.149-000001c4", "sub-record-check,s,1(exten,6000,dontcare)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("SIP/71.223.91.149-000001c4", "5?initialized") in new stack
-- Goto (sub-record-check,s,10)
-- Executing [s@sub-record-check:10] NoOp("SIP/71.223.91.149-000001c4", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("SIP/71.223.91.149-000001c4", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("SIP/71.223.91.149-000001c4", "5?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("SIP/71.223.91.149-000001c4", "1?sub-record-check,exten,1") in new stack
-- Goto (sub-record-check,exten,1)
-- Executing [exten@sub-record-check:1] NoOp("SIP/71.223.91.149-000001c4", "Exten Recording Check between 10001 and 6000") in new stack
-- Executing [exten@sub-record-check:2] Set("SIP/71.223.91.149-000001c4", "CALLTYPE=external") in new stack
-- Executing [exten@sub-record-check:3] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(CALLTYPE=)") in new stack
-- Executing [exten@sub-record-check:4] Set("SIP/71.223.91.149-000001c4", "CALLEE=dontcare") in new stack
-- Executing [exten@sub-record-check:5] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(CALLEE=dontcare)") in new stack
-- Executing [exten@sub-record-check:6] GotoIf("SIP/71.223.91.149-000001c4", "1?callee") in new stack
-- Goto (sub-record-check,exten,11)
-- Executing [exten@sub-record-check:11] Gosub("SIP/71.223.91.149-000001c4", "recordcheck,1(dontcare,external,6000)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("SIP/71.223.91.149-000001c4", "Starting recording check against dontcare") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("SIP/71.223.91.149-000001c4", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [recordcheck@sub-record-check:3] Return("SIP/71.223.91.149-000001c4", "") in new stack
-- Executing [exten@sub-record-check:12] Return("SIP/71.223.91.149-000001c4", "") in new stack
-- Executing [s@macro-exten-vm:13] GotoIf("SIP/71.223.91.149-000001c4", "1?macrodial") in new stack
-- Goto (macro-exten-vm,s,19)
-- Executing [s@macro-exten-vm:19] GosubIf("SIP/71.223.91.149-000001c4", "0?clrheader,1()") in new stack
-- Executing [s@macro-exten-vm:20] Macro("SIP/71.223.91.149-000001c4", "dial-one,15,Ttr,6000") in new stack
-- Executing [s@macro-dial-one:1] Set("SIP/71.223.91.149-000001c4", "DEXTEN=6000") in new stack
-- Executing [s@macro-dial-one:2] Set("SIP/71.223.91.149-000001c4", "_CRMSOURCE=10001") in new stack
-- Executing [s@macro-dial-one:3] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(EXTTOCALL=6000)") in new stack
-- Executing [s@macro-dial-one:4] Set("SIP/71.223.91.149-000001c4", "DIALSTATUS_CW=") in new stack
-- Executing [s@macro-dial-one:5] GosubIf("SIP/71.223.91.149-000001c4", "0?screen,1()") in new stack
-- Executing [s@macro-dial-one:6] GosubIf("SIP/71.223.91.149-000001c4", "0?cf,1()") in new stack
-- Executing [s@macro-dial-one:7] GotoIf("SIP/71.223.91.149-000001c4", "1?skip1") in new stack
-- Goto (macro-dial-one,s,10)
-- Executing [s@macro-dial-one:10] GotoIf("SIP/71.223.91.149-000001c4", "0?nodial") in new stack
-- Executing [s@macro-dial-one:11] GotoIf("SIP/71.223.91.149-000001c4", "0?continue") in new stack
-- Executing [s@macro-dial-one:12] Set("SIP/71.223.91.149-000001c4", "EXTHASCW=ENABLED") in new stack
-- Executing [s@macro-dial-one:13] GotoIf("SIP/71.223.91.149-000001c4", "0?next1:cwinusebusy") in new stack
-- Goto (macro-dial-one,s,25)
-- Executing [s@macro-dial-one:25] GotoIf("SIP/71.223.91.149-000001c4", "0?next3:continue") in new stack
-- Goto (macro-dial-one,s,27)
-- Executing [s@macro-dial-one:27] GotoIf("SIP/71.223.91.149-000001c4", "0?nodial") in new stack
-- Executing [s@macro-dial-one:28] GosubIf("SIP/71.223.91.149-000001c4", "1?dstring,1():dlocal,1()") in new stack
-- Executing [dstring@macro-dial-one:1] Set("SIP/71.223.91.149-000001c4", "DSTRING=") in new stack
-- Executing [dstring@macro-dial-one:2] Set("SIP/71.223.91.149-000001c4", "DEVICES=6000") in new stack
-- Executing [dstring@macro-dial-one:3] ExecIf("SIP/71.223.91.149-000001c4", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:4] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(DEVICES=000)") in new stack
-- Executing [dstring@macro-dial-one:5] Set("SIP/71.223.91.149-000001c4", "LOOPCNT=1") in new stack
-- Executing [dstring@macro-dial-one:6] Set("SIP/71.223.91.149-000001c4", "ITER=1") in new stack
-- Executing [dstring@macro-dial-one:7] Set("SIP/71.223.91.149-000001c4", "THISDIAL=SIP/6000") in new stack
-- Executing [dstring@macro-dial-one:8] GosubIf("SIP/71.223.91.149-000001c4", "1?zap2dahdi,1()") in new stack
-- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/71.223.91.149-000001c4", "0?Return()") in new stack
-- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/71.223.91.149-000001c4", "NEWDIAL=") in new stack
-- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/71.223.91.149-000001c4", "LOOPCNT2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/71.223.91.149-000001c4", "ITER2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/71.223.91.149-000001c4", "THISPART2=SIP/6000") in new stack
-- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(THISPART2=DAHDI/6000)") in new stack
-- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/71.223.91.149-000001c4", "NEWDIAL=SIP/6000&") in new stack
-- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/71.223.91.149-000001c4", "ITER2=2") in new stack
-- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/71.223.91.149-000001c4", "0?begin2") in new stack
-- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/71.223.91.149-000001c4", "THISDIAL=SIP/6000") in new stack
-- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/71.223.91.149-000001c4", "") in new stack
-- Executing [dstring@macro-dial-one:9] GotoIf("SIP/71.223.91.149-000001c4", "1?docheck") in new stack
-- Goto (macro-dial-one,dstring,14)
-- Executing [dstring@macro-dial-one:14] GotoIf("SIP/71.223.91.149-000001c4", "0?skipset") in new stack
-- Executing [dstring@macro-dial-one:15] Set("SIP/71.223.91.149-000001c4", "DSTRING=SIP/6000&") in new stack
-- Executing [dstring@macro-dial-one:16] Set("SIP/71.223.91.149-000001c4", "ITER=2") in new stack
-- Executing [dstring@macro-dial-one:17] GotoIf("SIP/71.223.91.149-000001c4", "0?begin") in new stack
-- Executing [dstring@macro-dial-one:18] ExecIf("SIP/71.223.91.149-000001c4", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:19] Set("SIP/71.223.91.149-000001c4", "DSTRING=SIP/6000") in new stack
-- Executing [dstring@macro-dial-one:20] Return("SIP/71.223.91.149-000001c4", "") in new stack
-- Executing [s@macro-dial-one:29] GotoIf("SIP/71.223.91.149-000001c4", "0?nodial") in new stack
-- Executing [s@macro-dial-one:30] GotoIf("SIP/71.223.91.149-000001c4", "0?skiptrace") in new stack
-- Executing [s@macro-dial-one:31] GosubIf("SIP/71.223.91.149-000001c4", "1?ctset,1():ctclear,1()") in new stack
-- Executing [ctset@macro-dial-one:1] Set("SIP/71.223.91.149-000001c4", "DB(CALLTRACE/6000)=10001") in new stack
-- Executing [ctset@macro-dial-one:2] Return("SIP/71.223.91.149-000001c4", "") in new stack
-- Executing [s@macro-dial-one:32] Set("SIP/71.223.91.149-000001c4", "D_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dial-one:33] NoOp("SIP/71.223.91.149-000001c4", "Blind Transfer: , Attended Transfer: , User: , Alert Info: ") in new stack
-- Executing [s@macro-dial-one:34] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(ALERT_INFO=)") in new stack
-- Executing [s@macro-dial-one:35] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(ALERT_INFO=)") in new stack
-- Executing [s@macro-dial-one:36] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(ALERT_INFO=)") in new stack
-- Executing [s@macro-dial-one:37] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
-- Executing [s@macro-dial-one:38] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
-- Executing [s@macro-dial-one:39] GosubIf("SIP/71.223.91.149-000001c4", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
-- Executing [s@macro-dial-one:40] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(CHANNEL(musicclass)=)") in new stack
-- Executing [s@macro-dial-one:41] GosubIf("SIP/71.223.91.149-000001c4", "0?qwait,1()") in new stack
-- Executing [s@macro-dial-one:42] Set("SIP/71.223.91.149-000001c4", "__CWIGNORE=") in new stack
-- Executing [s@macro-dial-one:43] Set("SIP/71.223.91.149-000001c4", "__KEEPCID=TRUE") in new stack
-- Executing [s@macro-dial-one:44] GotoIf("SIP/71.223.91.149-000001c4", "0?usegoto,1") in new stack
-- Executing [s@macro-dial-one:45] GotoIf("SIP/71.223.91.149-000001c4", "1?godial") in new stack
-- Goto (macro-dial-one,s,50)
-- Executing [s@macro-dial-one:50] Macro("SIP/71.223.91.149-000001c4", "dialout-one-predial-hook,") in new stack
-- Executing [s@macro-dialout-one-predial-hook:1] MacroExit("SIP/71.223.91.149-000001c4", "") in new stack
-- Executing [s@macro-dial-one:51] ExecIf("SIP/71.223.91.149-000001c4", "1?Set(D_OPTIONS=trI)") in new stack
-- Executing [s@macro-dial-one:52] NoOp("SIP/71.223.91.149-000001c4", "") in new stack
-- Executing [s@macro-dial-one:53] Dial("SIP/71.223.91.149-000001c4", "SIP/6000,15,trIb(func-apply-sipheaders^s^1)") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- SIP/6000-000001c5 Internal Gosub(func-apply-sipheaders,s,1) start
-- Executing [s@func-apply-sipheaders:1] ExecIf("SIP/6000-000001c5", "1?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
-- Executing [s@func-apply-sipheaders:2] NoOp("SIP/6000-000001c5", "Applying SIP Headers to channel") in new stack
-- Executing [s@func-apply-sipheaders:3] Set("SIP/6000-000001c5", "SIPHEADERKEYS=") in new stack
-- Executing [s@func-apply-sipheaders:4] While("SIP/6000-000001c5", "0") in new stack
-- Jumping to priority 7
-- Executing [s@func-apply-sipheaders:8] Return("SIP/6000-000001c5", "") in new stack
== Spawn extension (from-internal, 6000, 1) exited non-zero on 'SIP/6000-000001c5'
-- SIP/6000-000001c5 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
-- Called SIP/6000
-- Connected line update to SIP/71.223.91.149-000001c4 prevented.
-- SIP/6000-000001c5 is ringing
-- Connected line update to SIP/71.223.91.149-000001c4 prevented.
-- SIP/6000-000001c5 answered SIP/71.223.91.149-000001c4
0x7f18fc3c0d70 -- Probation passed - setting RTP source address to 192.168.0.117:61858
[2017-10-09 09:01:05] NOTICE[2166]: chan_sip.c:29270 check_rtp_timeout: Disconnecting call 'SIP/71.223.91.149-000001c4' for lack of RTP activity in 31 seconds
-- Executing [h@macro-dial-one:1] Macro("SIP/71.223.91.149-000001c4", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(CDR(recordingfile)=.wav)") in new stack
-- Executing [s@macro-hangupcall:2] GotoIf("SIP/71.223.91.149-000001c4", "1?theend") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:5] Hangup("SIP/71.223.91.149-000001c4", "") in new stack
== Spawn extension (macro-hangupcall, s, 5) exited non-zero on 'SIP/71.223.91.149-000001c4' in macro 'hangupcall'
== Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/71.223.91.149-000001c4'
-- SIP/71.223.91.149-000001c4 Internal Gosub(crm-hangup,s,1) start
-- Executing [s@crm-hangup:1] NoOp("SIP/71.223.91.149-000001c4", "Sending Hangup to CRM") in new stack
-- Executing [s@crm-hangup:2] NoOp("SIP/71.223.91.149-000001c4", "HANGUP CAUSE: 44") in new stack
-- Executing [s@crm-hangup:3] ExecIf("SIP/71.223.91.149-000001c4", "0?Set(_CRMVOICEMAIL=)") in new stack
-- Executing [s@crm-hangup:4] NoOp("SIP/71.223.91.149-000001c4", "MASTER CHANNEL: 1507564820.452 = 1507564820.452") in new stack
-- Executing [s@crm-hangup:5] GotoIf("SIP/71.223.91.149-000001c4", "0?return") in new stack
-- Executing [s@crm-hangup:6] Set("SIP/71.223.91.149-000001c4", "_CRMHANGUP=1") in new stack
-- Executing [s@crm-hangup:7] AGI("SIP/71.223.91.149-000001c4", "sangomacrm.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
-- AGI Script sangomacrm.agi completed, returning 0
-- Executing [s@crm-hangup:8] Return("SIP/71.223.91.149-000001c4", "") in new stack
== Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/71.223.91.149-000001c4'
-- SIP/71.223.91.149-000001c4 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
-- SIP/6000-000001c5 Internal Gosub(crm-hangup,s,1) start
-- Executing [s@crm-hangup:1] NoOp("SIP/6000-000001c5", "Sending Hangup to CRM") in new stack
-- Executing [s@crm-hangup:2] NoOp("SIP/6000-000001c5", "HANGUP CAUSE: 16") in new stack
-- Executing [s@crm-hangup:3] ExecIf("SIP/6000-000001c5", "0?Set(_CRMVOICEMAIL=)") in new stack
-- Executing [s@crm-hangup:4] NoOp("SIP/6000-000001c5", "MASTER CHANNEL: 1507564820.453 = 1507564820.452") in new stack
-- Executing [s@crm-hangup:5] GotoIf("SIP/6000-000001c5", "1?return") in new stack
-- Goto (crm-hangup,s,8)
-- Executing [s@crm-hangup:8] Return("SIP/6000-000001c5", "") in new stack
== Spawn extension (macro-dial-one, , 1) exited non-zero on 'SIP/6000-000001c5'
-- SIP/6000-000001c5 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
== Spawn extension (macro-dial-one, s, 53) exited non-zero on 'SIP/71.223.91.149-000001c4' in macro 'dial-one'
== Spawn extension (macro-exten-vm, s, 20) exited non-zero on 'SIP/71.223.91.149-000001c4' in macro 'exten-vm'
== Spawn extension (ext-local, 6000, 2) exited non-zero on 'SIP/71.223.91.149-000001c4'
[2017-10-09 09:01:06] WARNING[2166]: chan_sip.c:4038 retrans_pkt: Retransmission timeout reached on transmission 53b9a48cb7ff12b394a20ba758912167 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response
Inbound call terminates at 32 second
I am experiencing an issue where inbound calls are terminating at 32 seconds. I was told to enable sip debug. How do I do that?
Extension becomes unavailable
I'm brand new to Freepbx. I have setup a VPS online for a test box.
Asterisk version 13.16.0
I have setup a SIP trunk and am able to make make and receive calls.
I have purchased a Polycom vx411 phone and setup a X-Lite on my desktop for testing.
I'm finding that my polycom becomes unreachable after 1-2 minutes of being idle. Whenever I try to call it, the caller receives 3 rings and it goes to voicemail. The polycom itself never rings. If I route that same call to X-lite, X-lite rings as it should.
if I try calling the polycom from x-lite it too goes to voicemail after 3 rings.
The polycom only appears to accept incoming calls after it first boots OR after I make an outgoing call. It then appears to have 1-2 minutes of life to receive a call.
My Freepbx is on the internet, and my 2 phones are on my LAN behind a sonicwall firewall. I also tried to take the phone and plug it directly into my cable modem router to bypass the sonicwall and it does the same thing.
I've been doing a lot of reading. I double checked DND. I've recreated the extension. Rebooted the server & phone. Currently my asterisk info > peers reads like this:
Endpoint: 201/201 Unavailable 0 of inf
InAuth: 201-auth/201
Aor: 201 1
Contact: 201/sip:201@XXX.XX.XX.XXX:14884 11d67680d7 Unavail 0.000
Identify: 201-identify/201
Whoops error after install
Permission errors are almost always fixed with
fwconsole chown
Inbound call terminates at 32 second
Take a look at your Asterisk SIP settings external IP and the Chan SIP tab (found on same page) and look at your override external IP.
PHP Fatal error on check_portal.php
Thats great news. Shows me that my decision to uninstall FreePBX was a good idea. Never trust any obfuscated scripts!
Missing Default MOH
I migrated from an older version of Elastix using the conversion tool. Now my MOH doesn't work and I'm missing the "default" MOH folder. The web gui won't let me re-add it. Any suggestions?
Missing Default MOH
Try:
fwconsole ma install music
Trying to configure incoming fax-to-email
I'm trying to configure incoming fax-to-email in FreePBX, but I'm running into some issues.
What I'm trying to do is allow faxes that are sent to an incoming DID to be forwarded to the email entered into a users extension. I've set up everything using the wiki as a guide.
Fax detection is set to Dahdi (I've configured the config file correctly.)
From what I can tell from the logs, the pbx is detecting the faxes and accepting them, but not forwarding them to email.
Here is the asterik log for the incoming fax.
[2017-10-09 11:16:47] VERBOSE[2531][C-00002b64] sig_pri.c: -- Accepting call from '6182302579' to '9257065564' on channel 0/5, span 1
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:1] Set("DAHDI/i1/6182302579-165f", "__DIRECTION=INBOUND") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:2] Gosub("DAHDI/i1/6182302579-165f", "sub-record-check,s,1(in,9257065564,dontcare)") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [in@sub-record-check:1] NoOp("DAHDI/i1/6182302579-165f", "Inbound Recording Check to 9257065564") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [in@sub-record-check:4] Gosub("DAHDI/i1/6182302579-165f", "recordcheck,1(dontcare,in,9257065564)") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:3] Gosub("DAHDI/i1/6182302579-165f", "app-blacklist-check,s,1()") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:4] Set("DAHDI/i1/6182302579-165f", "_FROMDID=9257065564") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:5] Set("DAHDI/i1/6182302579-165f", "CDR(did)=9257065564") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:6] ExecIf("DAHDI/i1/6182302579-165f", "0 ?Set(CALLERID(name)=6182302579)") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:7] Set("DAHDI/i1/6182302579-165f", "__MOHCLASS=") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:8] Set("DAHDI/i1/6182302579-165f", "_REVERSALREJECT=FALSE") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:9] GotoIf("DAHDI/i1/6182302579-165f", "1?post-reverse-charge") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Goto (from-digital,9257065564,11)
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:11] NoOp("DAHDI/i1/6182302579-165f", "") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:12] Set("DAHDI/i1/6182302579-165f", "_CALLINGNAMEPRESSV=allowed_not_screened") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:13] Set("DAHDI/i1/6182302579-165f", "_CALLINGNUMPRESSV=allowed_not_screened") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:14] Set("DAHDI/i1/6182302579-165f", "CALLERID(name-pres)=allowed_not_screened") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:15] Set("DAHDI/i1/6182302579-165f", "CALLERID(num-pres)=allowed_not_screened") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:16] NoOp("DAHDI/i1/6182302579-165f", "CallerID Entry Point") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:17] Set("DAHDI/i1/6182302579-165f", "FAX_DEST=ext-fax^67^1") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:18] Set("DAHDI/i1/6182302579-165f", "FAXOPT(faxdetect)=yes") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:19] Answer("DAHDI/i1/6182302579-165f", "") in new stack
[2017-10-09 11:16:47] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [9257065564@from-digital:20] Wait("DAHDI/i1/6182302579-165f", "6") in new stack
[2017-10-09 11:17:15] VERBOSE[27399][C-00002b64] pbx.c: -- Executing [h@ext-fax:4] System("DAHDI/i1/6182302579-165f", "/var/lib/asterisk/bin/fax2mail.php --remotestationid "FaxZero.com" --user "67" --dest "9257065564" --callerid "IlVUIFNVUFBPUlQiIDw2MTgyMzAyNTc5Pg==" --file /var/spool/asterisk/fax/1507573007.26008.tif --exten "USER_HERE (67)" --delete "true" --attachformat """) in new stack
Trying to configure incoming fax-to-email
Hi!
Anything in /var/log/maillog during that time?
Good luck and have a nice day!
Nick
Issues after upgrade to FreePBX14 - LoadLicenseIfExists - Incompatible File Format
ok, dumb question, where's the ticket system?
Issues after upgrade to FreePBX14 - LoadLicenseIfExists - Incompatible File Format
Hi!
There was actually a link to it in the Troubleshooting section at the bottom of the page you followed to upgrade your system to FreePBX 14/Sangoma 7.
You must upload sngupdate and post_sngupdate to the ticket you create...
Good luck and have a nice day!
Nick
Failover ISP connection and changing the Asterisk SIP IP
I just wanted to bump this post and see if anyone had any ideas or comments to let me know if I am on the right or wrong track. I know awhile ago @avayax said he has done this a few times and it works well.
REST API documentation
Thanks I will take a look at that. Some PHP examples would be greatly appreciated.
REST API documentation
If you read the article it talks about PHP libraries.
Eth ports on Sangoma UC 40
There would be no reason to LAG the ports together in a PBXact/FreePBX. Each sip session uses about 60Kbps per session. If you calculated that at 30 calls that's about 1800Kbps or 1.8Mbps. So even if your sip sessions were using a whopping 100Kbps, 30 calls would only equate to 3.0Mbps on your 1000Mbps connection.
Which is more efficient, ChanSIP or PJSIP?
Ok will do, seems weird to not be using the 'more modern' technology though.
Thanks