@enrica_r wrote:

We use an extension like as a picket number. We reach us via cell phones which are entered in the follow me list. The list is used as a hunting group. So far ok.

The cell phones have a ring time limite of 25 seconds. Afterwards their voicemail is answering. Therefore we have set the ring time of follow me to 20 seconds. Afterwards next line we be called. After all tried we play an annoucement to call later again. We don't want to have an endless loop.

Often 24 seconds isn't enough time to answer the cell phone. So I want to hang-up and call again.

Because we can't set a repeat counter like "try x times". I thought to set same number twice in a list eg.
079XXXXXXX#
079XXXXXXX#
076YYYYYYY#
076YYYYYYY#

or
079XXXXXXX#
076YYYYYYY#
079XXXXXXX#
076YYYYYYY#

Unfortunately the script "dialparties.agi" says "Extension '079XXXXXX' already in the dialstring, ignoring duplicate". So also this flexible way to repeat a number isn't possible at the moment.

Isn't it possible to allow duplicates via an option switch or generally if ring strategy is "hunt"?

Thanks.

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@dotcom wrote:

Hello all,

I'm trying to dial to an emergency 3 digit number 117, but Asterisk keeps dialing the local SIP/117 extension which doesn't exists.

Since I need to add a custom X-header, I'm using the from-internal-custom context in extension_custom:

[from-internal-custom]
exten => 117,1,NoOp(Emergency call...)
exten => 117,n,SIPAddHeader(X-Postalcode: 1000)
exten => 117,n,Set(CALLERID(num)=+3222222222)
exten => 117,n,Dial(SIP/trunk_prim_out/117)
exten => 117,n,Hangup

In the logs I see:

 -- Executing [117@from-internal:1] Dial("SIP/1092-00000eb0", "SIP/117,10,tT") in new stack
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [117@from-internal:2] GotoIf("SIP/1092-00000eb0", "1?TransferFailed") in new stack
  == Spawn extension (from-internal, 117, 2) exited non-zero on 'SIP/1092-00000eb0'
    -- Executing [h@from-internal:1] Macro("SIP/1092-00000eb0", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1092-00000eb0", "1?theend") in new stack

When changing 117 to 1117 (4 digits) it is working correctly.

How can I call externally to 117 and prevent the call going to internal SIP/117?

Thanks a lot for your help!

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@Basildane wrote:

I've been on Asterisk 1.8 for many years and I'm now forcing myself to move all my stuff to 14 Distro. I have 90% of my stuff working in the new system, but there is one thing that has me completely stumped. I want to move my MeetMe stuff to ConfBridge, because this mission is to get up to the latest technology after 11 years or so on a customized Trixbox build.

So, I have a simple Application Feature *5 Quick Conference. It takes both ends of a call and transfers them into meeting 1000. We use it all the time at home to quickly share a call with family members. It works perfectly with MeetMe, but it just hangs-up when I try to use ConfBridge. (Yes, I changed the default conference tech in advanced settings).
When I use ConfBridge, the log shows "joined" and then "left" the conference, immediately, and the call terminates. Any thoughts?
I've tried many variations of ConfBridge, including moving the call out to AGI, but the results are exactly the same each time. I can direct-dial conference 1000 and it works. I just cannot transfer endpoints to it from my dialplan.

I have simplified the code down to the bare minimum to demonstrate my problem.

This is the config:
globals_custom.conf
DYNAMIC_FEATURES = qconf

features_applicationmap_custom.conf
qconf => *5,self,Macro,qconf

extensions_custom.conf
[macro-qconf]
exten => s,1,Noop(Quick Conference)
same => n,ChannelRedirect(${BRIDGEPEER},qconf-user,s,1)
;;;same => n,ConfBridge(1000) ; THIS FAILS
same => n,MeetMe(1000,A1M) ; THIS WORKS
same => n,MacroExit()

[qconf-user]
exten => s,1,Macro(user-callerid)
;;;same => n,ConfBridge(1000) ; THIS FAILS
same => n,MeetMe(1000,xw1M) ; THIS WORKS
same => n,MacroExit()

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@chaptech wrote:

Hi All,

Trying to setup something like SLA on my Freepbx box, whereas i have a desk phone and a cordless setup using the same extension. I know out of the box sla itself is not supported but i was also reading a similar thing can be achieved using pjsip as it supports multiple logins on the same extension.

So i have set this up, a w56p yealink cordless as the primary setup through EPM, then i also setup a custom extension mapping through EPM for the t46g desk phone. Both phones auto provision through the EPM, both phones register, i can call out from either phone but when i call in only the primary extension rings.

Has anyone got any clues or how to make it work so both phones also ring in? I know i could use a ring group but i also want things like the blf to show either phone busy.

cheers

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@Trezeguet17 wrote:

Hi people, im having a really annoying issue, i have my Yealink phone , with all the dnd and call forwarding functions deactivated, but when i dial the extensión , i cant receive the call it sond like if its bussy but im not

Here is the output

Executing [1106@ivr-3:1] Macro("SIP/Claro-0000a3c1", "blkvm-clr,") in new stack
-- Executing [s@macro-blkvm-clr:1] Set("SIP/Claro-0000a3c1", "SHARED(BLKVM,)=") in new stack
-- Executing [s@macro-blkvm-clr:2] Set("SIP/Claro-0000a3c1", "GOSUB_RETVAL=") in new stack
-- Executing [s@macro-blkvm-clr:3] MacroExit("SIP/Claro-0000a3c1", "") in new stack
-- Executing [1106@ivr-3:2] Set("SIP/Claro-0000a3c1", "__NODEST=") in new stack
-- Executing [1106@ivr-3:3] Goto("SIP/Claro-0000a3c1", "from-did-direct,1106,1") in new stack
-- Goto (from-did-direct,1106,1)
-- Executing [1106@from-did-direct:1] Set("SIP/Claro-0000a3c1", "__RINGTIMER=15") in new stack
-- Executing [1106@from-did-direct:2] Macro("SIP/Claro-0000a3c1", "exten-vm,1106,1106,0,0,0") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/Claro-0000a3c1", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/Claro-0000a3c1", "TOUCH_MONITOR=1503931872.225705") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/Claro-0000a3c1", "AMPUSER=17435737") in new stack
2409- -- Executing [s@macro-user-callerid:3] GotoIf("SIP/Claro-0000a3c1", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/Claro-0000a3c1", "1?Set(REALCALLERIDNUM=17435737)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/Claro-0000a3c1", "AMPUSER=") in new stack
2873- -- Executing [s@macro-user-callerid:6] GotoIf("SIP/Claro-0000a3c1", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/Claro-0000a3c1", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/Claro-0000a3c1", "1?report") in new stack
--
-- Executing [s@macro-exten-vm:3] Set("SIP/Claro-0000a3c1", "__EXTTOCALL=1106") in new stack
4922: -- Executing [s@macro-exten-vm:4] Set("SIP/Claro-0000a3c1", "__PICKUPMARK=1106") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/Claro-0000a3c1", "RT=15") in new stack
-- Executing [s@macro-exten-vm:6] Gosub("SIP/Claro-0000a3c1", "sub-record-check,s,1(exten,1106,dontcare)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("SIP/Claro-0000a3c1", "8?initialized") in new stack
-- Goto (sub-record-check,s,10)
-- Executing [s@sub-record-check:10] NoOp("SIP/Claro-0000a3c1", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("SIP/Claro-0000a3c1", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("SIP/Claro-0000a3c1", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("SIP/Claro-0000a3c1", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("SIP/Claro-0000a3c1", "5?checkaction") in new stack
6431- -- Goto (sub-record-check,s,17)
6467- -- Executing [s@sub-record-check:17] GotoIf("SIP/Claro-0000a3c1", "1?sub-record-check,exten,1") in new stack
--
6652: -- Executing [exten@sub-record-check:1] NoOp("SIP/Claro-0000a3c1", "Exten Recording Check between 17435737 and 1106") in new stack
6820- -- Executing [exten@sub-record-check:2] Set("SIP/Claro-0000a3c1", "CALLTYPE=external") in new stack
6957- -- Executing [exten@sub-record-check:3] ExecIf("SIP/Claro-0000a3c1", "0?Set(CALLTYPE=)") in new stack
-- Executing [exten@sub-record-check:4] Set("SIP/Claro-0000a3c1", "CALLEE=yes") in new stack
-- Executing [exten@sub-record-check:5] ExecIf("SIP/Claro-0000a3c1", "0?Set(CALLEE=dontcare)") in new stack
-- Executing [exten@sub-record-check:6] GotoIf("SIP/Claro-0000a3c1", "1?callee") in new stack
-- Goto (sub-record-check,exten,11)
-- Executing [exten@sub-record-check:11] Gosub("SIP/Claro-0000a3c1", "recordcheck,1(yes,external,1106)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("SIP/Claro-0000a3c1", "Starting recording check against yes") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("SIP/Claro-0000a3c1", "yes") in new stack
-- Goto (sub-record-check,recordcheck,9)
-- Executing [recordcheck@sub-record-check:9] ExecIf("SIP/Claro-0000a3c1", "0?Return()") in new stack
-- Executing [recordcheck@sub-record-check:10] Set("SIP/Claro-0000a3c1", "_RECPOLICY_MODE=YES") in new stack
-- Executing [recordcheck@sub-record-check:11] Goto("SIP/Claro-0000a3c1", "startrec") in new stack
-- Goto (sub-record-check,recordcheck,16)
-- Executing [recordcheck@sub-record-check:16] NoOp("SIP/Claro-0000a3c1", "Starting recording: external, 1106") in new stack
-- Executing [recordcheck@sub-record-check:17] Set("SIP/Claro-0000a3c1", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [recordcheck@sub-record-check:18] Set("SIP/Claro-0000a3c1", "__CALLFILENAME=external-1106-17435737-20170828-095112-1503931872.225705") in new stack
-- Executing [recordcheck@sub-record-check:19] MixMonitor("SIP/Claro-0000a3c1", "2017/08/28/external-1106-17435737-20170828-095112-1503931872.225705.wav,ai(LOCAL_MIXMON_ID),") in new stack
== Begin MixMonitor Recording SIP/Claro-0000a3c1
-- Executing [recordcheck@sub-record-check:20] Set("SIP/Claro-0000a3c1", "_MIXMONID=0x7f4c88408620") in new stack
-- Executing [recordcheck@sub-record-check:21] Set("SIP/Claro-0000a3c1", "_RECORDID=SIP/Claro-0000a3c1") in new stack
-- Executing [recordcheck@sub-record-check:22] Set("SIP/Claro-0000a3c1", "_RECSTATUS=RECORDING") in new stack
-- Executing [recordcheck@sub-record-check:23] Set("SIP/Claro-0000a3c1", "CDR(recordingfile)=external-1106-17435737-20170828-095112-1503931872.225705.wav") in new stack
-- Executing [recordcheck@sub-record-check:24] Return("SIP/Claro-0000a3c1", "") in new stack
-- Executing [exten@sub-record-check:12] Return("SIP/Claro-0000a3c1", "") in new stack
-- Executing [s@macro-exten-vm:7] GotoIf("SIP/Claro-0000a3c1", "1?macrodial") in new stack
-- Goto (macro-exten-vm,s,13)
-- Executing [s@macro-exten-vm:13] GosubIf("SIP/Claro-0000a3c1", "0?clrheader,1()") in new stack
-- Executing [s@macro-exten-vm:14] Macro("SIP/Claro-0000a3c1", "dial-one,15,Ttr,1106") in new stack
-- Executing [s@macro-dial-one:1] Set("SIP/Claro-0000a3c1", "DEXTEN=1106") in new stack
-- Executing [s@macro-dial-one:2] Set("SIP/Claro-0000a3c1", "DIALSTATUS_CW=") in new stack
-- Executing [s@macro-dial-one:3] GosubIf("SIP/Claro-0000a3c1", "0?screen,1()") in new stack
-- Executing [s@macro-dial-one:4] GosubIf("SIP/Claro-0000a3c1", "0?cf,1()") in new stack
-- Executing [s@macro-dial-one:5] GotoIf("SIP/Claro-0000a3c1", "1?skip1") in new stack
-- Goto (macro-dial-one,s,8)
-- Executing [s@macro-dial-one:8] GotoIf("SIP/Claro-0000a3c1", "0?nodial") in new stack
-- Executing [s@macro-dial-one:9] GotoIf("SIP/Claro-0000a3c1", "0?continue") in new stack
-- Executing [s@macro-dial-one:10] Set("SIP/Claro-0000a3c1", "EXTHASCW=ENABLED") in new stack
-- Executing [s@macro-dial-one:11] GotoIf("SIP/Claro-0000a3c1", "0?next1:cwinusebusy") in new stack
--
-- Executing [dstring@macro-dial-one:2] Set("SIP/Claro-0000a3c1", "DEVICES=1106") in new stack
-- Executing [dstring@macro-dial-one:3] ExecIf("SIP/Claro-0000a3c1", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:4] ExecIf("SIP/Claro-0000a3c1", "0?Set(DEVICES=106)") in new stack
-- Executing [dstring@macro-dial-one:5] Set("SIP/Claro-0000a3c1", "LOOPCNT=1") in new stack
-- Executing [dstring@macro-dial-one:6] Set("SIP/Claro-0000a3c1", "ITER=1") in new stack
-- Executing [dstring@macro-dial-one:7] Set("SIP/Claro-0000a3c1", "THISDIAL=SIP/1106") in new stack
-- Executing [dstring@macro-dial-one:8] GosubIf("SIP/Claro-0000a3c1", "1?zap2dahdi,1()") in new stack
-- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/Claro-0000a3c1", "0?Return()") in new stack
-- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/Claro-0000a3c1", "NEWDIAL=") in new stack
-- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/Claro-0000a3c1", "LOOPCNT2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/Claro-0000a3c1", "ITER2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/Claro-0000a3c1", "THISPART2=SIP/1106") in new stack
-- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/Claro-0000a3c1", "0?Set(THISPART2=DAHDI/1106)") in new stack
-- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/Claro-0000a3c1", "NEWDIAL=SIP/1106&") in new stack
-- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/Claro-0000a3c1", "ITER2=2") in new stack
-- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/Claro-0000a3c1", "0?begin2") in new stack
-- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/Claro-0000a3c1", "THISDIAL=SIP/1106") in new stack
-- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/Claro-0000a3c1", "") in new stack
-- Executing [dstring@macro-dial-one:9] GotoIf("SIP/Claro-0000a3c1", "1?doset") in new stack
-- Goto (macro-dial-one,dstring,13)
-- Executing [dstring@macro-dial-one:13] Set("SIP/Claro-0000a3c1", "DSTRING=SIP/1106&") in new stack
-- Executing [dstring@macro-dial-one:14] Set("SIP/Claro-0000a3c1", "ITER=2") in new stack
-- Executing [dstring@macro-dial-one:15] GotoIf("SIP/Claro-0000a3c1", "0?begin") in new stack
-- Executing [dstring@macro-dial-one:16] ExecIf("SIP/Claro-0000a3c1", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:17] Set("SIP/Claro-0000a3c1", "DSTRING=SIP/1106") in new stack
-- Executing [dstring@macro-dial-one:18] Return("SIP/Claro-0000a3c1", "") in new stack
-- Executing [s@macro-dial-one:27] GotoIf("SIP/Claro-0000a3c1", "0?nodial") in new stack
-- Executing [s@macro-dial-one:28] GotoIf("SIP/Claro-0000a3c1", "0?skiptrace") in new stack
-- Executing [s@macro-dial-one:29] GosubIf("SIP/Claro-0000a3c1", "1?ctset,1():ctclear,1()") in new stack
-- Executing [ctset@macro-dial-one:1] Set("SIP/Claro-0000a3c1", "DB(CALLTRACE/1106)=17435737") in new stack
-- Executing [ctset@macro-dial-one:2] Return("SIP/Claro-0000a3c1", "") in new stack
-- Executing [s@macro-dial-one:30] Set("SIP/Claro-0000a3c1", "D_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dial-one:31] ExecIf("SIP/Claro-0000a3c1", "0?SIPAddHeader(Alert-Info: )") in new stack
-- Executing [s@macro-dial-one:32] ExecIf("SIP/Claro-0000a3c1", "0?SIPAddHeader()") in new stack
-- Executing [s@macro-dial-one:33] ExecIf("SIP/Claro-0000a3c1", "1?Set(CHANNEL(musicclass)=default)") in new stack
-- Executing [s@macro-dial-one:34] GosubIf("SIP/Claro-0000a3c1", "0?qwait,1()") in new stack
-- Executing [s@macro-dial-one:35] Set("SIP/Claro-0000a3c1", "__CWIGNORE=") in new stack
-- Executing [s@macro-dial-one:36] Set("SIP/Claro-0000a3c1", "__KEEPCID=TRUE") in new stack
-- Executing [s@macro-dial-one:37] GotoIf("SIP/Claro-0000a3c1", "0?usegoto,1") in new stack
--
-- Executing [s@macro-dial-one:45] Dial("SIP/Claro-0000a3c1", "SIP/1106,15,trI") in new stack
== Using SIP RTP TOS bits 184
20870- == Using SIP RTP CoS mark 5
-- Called SIP/1106
-- Connected line update to SIP/Claro-0000a3c1 prevented.
-- Got SIP response 480 "Temporarily not available" back from 181.48.43.34:62900
21151: -- SIP/1106-0000a3c2 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dial-one:46] ExecIf("SIP/Claro-0000a3c1", "0?MacroExit()") in new stack
-- Executing [s@macro-dial-one:47] ExecIf("SIP/Claro-0000a3c1", "0?Set(DIALSTATUS=)") in new stack
-- Executing [s@macro-dial-one:48] GosubIf("SIP/Claro-0000a3c1", "0?s-CONGESTION,1()") in new stack
-- Executing [s@macro-dial-one:49] MacroExit("SIP/Claro-0000a3c1", "") in new stack
-- Executing [s@macro-exten-vm:15] Set("SIP/Claro-0000a3c1", "SV_DIALSTATUS=CONGESTION") in new stack
-- Executing [s@macro-exten-vm:16] GosubIf("SIP/Claro-0000a3c1", "0?docfu,1()") in new stack
-- Executing [s@macro-exten-vm:17] GosubIf("SIP/Claro-0000a3c1", "0?docfb,1()") in new stack
-- Executing [s@macro-exten-vm:18] Set("SIP/Claro-0000a3c1", "DIALSTATUS=CONGESTION") in new stack
--
-- Executing [s@macro-exten-vm:21] Macro("SIP/Claro-0000a3c1", "vm,1106,CONGESTION,") in new stack
-- Executing [s@macro-vm:1] Macro("SIP/Claro-0000a3c1", "user-callerid,SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/Claro-0000a3c1", "TOUCH_MONITOR=1503931872.225705") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/Claro-0000a3c1", "AMPUSER=17435737") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/Claro-0000a3c1", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/Claro-0000a3c1", "0?Set(REALCALLERIDNUM=17435737)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/Claro-0000a3c1", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/Claro-0000a3c1", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/Claro-0000a3c1", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/Claro-0000a3c1", "1?report") in new stack
--
-- Executing [vmx@macro-vm:1] Set("SIP/Claro-0000a3c1", "MEXTEN=1106") in new stack
-- Executing [vmx@macro-vm:2] Set("SIP/Claro-0000a3c1", "MMODE=CONGESTION") in new stack
-- Executing [vmx@macro-vm:3] Set("SIP/Claro-0000a3c1", "RETVM=") in new stack
-- Executing [vmx@macro-vm:4] Set("SIP/Claro-0000a3c1", "MODE=unavail") in new stack
-- Executing [vmx@macro-vm:5] Macro("SIP/Claro-0000a3c1", "get-vmcontext,1106") in new stack
-- Executing [s@macro-get-vmcontext:1] Set("SIP/Claro-0000a3c1", "VMCONTEXT=default") in new stack
-- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/Claro-0000a3c1", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s@macro-get-vmcontext:300] NoOp("SIP/Claro-0000a3c1", "") in new stack
-- Executing [vmx@macro-vm:6] Set("SIP/Claro-0000a3c1", "MODE=unavail") in new stack
-- Executing [vmx@macro-vm:7] NoOp("SIP/Claro-0000a3c1", "MODE IS: unavail") in new stack
-- Executing [vmx@macro-vm:8] GotoIf("SIP/Claro-0000a3c1", "1?chknomsg") in new stack
-- Goto (macro-vm,vmx,10)
-- Executing [vmx@macro-vm:10] GotoIf("SIP/Claro-0000a3c1", "0?s-CONGESTION,1") in new stack
--
-- Executing [vmx@macro-vm:13] NoOp("SIP/Claro-0000a3c1", "Checking if ext 1106 is enabled: ") in new stack
-- Executing [vmx@macro-vm:14] GotoIf("SIP/Claro-0000a3c1", "1?s-CONGESTION,1") in new stack
-- Goto (macro-vm,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-vm:1] Macro("SIP/Claro-0000a3c1", "get-vmcontext,1106") in new stack
-- Executing [s@macro-get-vmcontext:1] Set("SIP/Claro-0000a3c1", "VMCONTEXT=default") in new stack
-- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/Claro-0000a3c1", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s@macro-get-vmcontext:300] NoOp("SIP/Claro-0000a3c1", "") in new stack
-- Executing [s-CONGESTION@macro-vm:2] VoiceMail("SIP/Claro-0000a3c1", "1106@default,u") in new stack
-- Playing 'vm-theperson.gsm' (language 'es')
-- Playing 'digits/1.gsm' (language 'es')
-- Playing 'digits/1.gsm' (language 'es')
-- Playing 'digits/0.gsm' (language 'es')
-- Playing 'digits/6.gsm' (language 'es')
== Spawn extension (macro-vm, s-CONGESTION, 2) exited non-zero on 'SIP/Claro-0000a3c1' in macro 'vm'
30625- == Spawn extension (macro-exten-vm, s, 21) exited non-zero on 'SIP/Claro-0000a3c1' in macro 'exten-vm'
== Spawn extension (from-did-direct, 1106, 2) exited non-zero on 'SIP/Claro-0000a3c1'
-- Executing [h@from-did-direct:1] Macro("SIP/Claro-0000a3c1", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] ExecIf("SIP/Claro-0000a3c1", "0?Set(CDR(recordingfile)=external-1106-17435737-20170828-095112-1503931872.225705.wav)") in new stack
-- Executing [s@macro-hangupcall:2] GotoIf("SIP/Claro-0000a3c1", "1?theend") in new stack
-- Goto (macro-hangupcall,s,4)
31448- -- Executing [s@macro-hangupcall:4] Hangup("SIP/Claro-0000a3c1", "") in new stack
31567- == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/Claro-0000a3c1' in macro 'hangupcall'
== Spawn extension (from-did-direct, h, 1) exited non-zero on 'SIP/Claro-0000a3c1'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/Claro-0000a3c1
-- Registered SIP '4504' at 181.129.27.90:43007

Thanks

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@stevensedory wrote:

Hi all,

We need to monitor an extension, from the endpoint perspective. We just need to know when it can no longer communicate with the server.

Right now, when that happens, the extension on our phone goes red (on grandstream) and has an x through it (polycom). However, I want to get notified as soon as that happens. I was hoping to use our monitoring software and an SNMP item, but I noticed most endpoints don't have SNMP.

Is there a way to do this? My first thought is to use a software phone that has SNMP enabled, but thought I'd reach out to the community beforehand to see if that's the best solution.

Reason is, we are having some serious issues as explained in the link below. Often times, asterisk doesn't crash for a long time, but there are issues with extensions going offline (even though the server shows said extension as "OK"). Here's the link to our issue: https://community.freepbx.org/t/consistent-asterisk-freepbx-crash-issue/43682/5

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@V4mpire wrote:

Hi,

I am having issues setting up freepbx, I have used the official listed guide for Centos 6.x, however I have got to the following issue:

freepbx]# ./start_asterisk start

STARTING ASTERISK
Asterisk Started
freepbx]# ./install -n
Assuming you are Database Root
Checking if SELinux is enabled...Its not (good)!
Reading /etc/asterisk/asterisk.conf...Done
Checking if Asterisk is running and we can talk to it as the 'asterisk' user...Error!
Error communicating with Asterisk. Ensure that Asterisk is properly installed and running as the asterisk user
Asterisk appears to be running as asterisk
Try starting Asterisk with the './start_asterisk start' command in this directory

I have checked it's under correct user:

freepbx]# ps aux |grep asterisk
asterisk 1547 0.0 1.3 293440 6612 ? S 11:22 0:00 /usr/sbin/httpd
asterisk 1548 0.0 1.4 293440 7132 ? S 11:22 0:00 /usr/sbin/httpd
asterisk 1549 0.0 1.3 293440 6612 ? S 11:22 0:00 /usr/sbin/httpd
asterisk 1550 0.0 1.4 293440 7052 ? S 11:22 0:00 /usr/sbin/httpd
asterisk 1551 0.0 1.2 293440 6448 ? S 11:22 0:00 /usr/sbin/httpd
asterisk 1552 0.0 1.2 293440 6448 ? S 11:22 0:00 /usr/sbin/httpd
asterisk 1553 0.0 1.2 293440 6448 ? S 11:22 0:00 /usr/sbin/httpd
asterisk 1554 0.0 1.2 293440 6448 ? S 11:22 0:00 /usr/sbin/httpd
root 3111 0.0 0.1 106080 764 pts/0 S 11:29 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk
asterisk 3114 0.0 5.8 632728 29344 pts/0 Sl 11:29 0:00 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c

However I did find this:

freepbx]# ./start_asterisk stop

STOPPING ASTERISK
/usr/sbin/asterisk: error while loading shared libraries: libjansson.so.4: cannot open shared object file: No such file or directory
Asterisk Stopped

I wouldn't assume this is related, correct me if i'm wrong considering it runs.

Hope someone can shed some light on this as I have looked around and seems the answers point to incorrect user etc which I have already confirmed

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@gwntc wrote:

Hi, We have a PBXact system that is still using older Aastra 6755i telephones. The voicemail app used to work correctly but has stopped working since the last updates several months ago. We would like to get it working again.

I've looked at the provisioning template in EPM and no longer have the option for XML-API with a drop down menu that allows me to select "REST-Voicemail". Is there a manual way of setting it via XML with a URI value?

For now, I will set it to speed dial 97.

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@pt1xoom wrote:

Hello,

I have just installed FreePBX 14. I am not able to upload greetings to a users voicemail box.

The error message is "There was an error. See the console logs for more details". Where are the console logs? I was watch the full log while upload and received the error, but no log entry was made on that log.

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@alenguav wrote:

I installed recently FREEPBX 14 and everything seems to be working except for this problem:
I can't hear caller’s voice and the caller also can't hear my voice

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@ivoxs wrote:

hi, guys thank for time and help, i need to get the Queque name on a custom context, thanks

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@Dan2731 wrote:

Hello,

I have an issue with remote backups. I'm trying to setup a "Warm Spare" configuration as outlined in these instructions.

Most of it is pretty straight forward. The issue I'm running into is the "Exclude NAT settings" option when backing up production server with a warm spare (henceforth referred to as spare server). When this option is selected, the IP Address of the spare server doesn't change (which is what it is supposed to do). However, the External IP Address references in Settings>Asterisk SIP Settings> Chan SIP Settings is changed to the primary IP Address.

This issue has come up in the forums before, but there appears to be no resolution.

Any help would be greatly appreciated. There may be a simple work around by excluding the settings from the backup, or running a hook, but I don't know which values those are.

P.S. Both the Production and Spare server are running on FreePBX 13.0.192.9

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@gwntc wrote:

Is there a way to disable call forwarding for an extension without actually going to that telephone? I am trying to disable CF for a user that has enabled it.

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@stevensedory wrote:

Brand spankin new FreePBX 13.0.192.16 with Asterisk 13.17.0

Doing our routine checks on in/out calls, two way audio, voicemail to email, etc., when I noticed I wasn't getting any voicemail emails. I then noticed voicemails aren't being stored on the phones (server I know) either.

This is a brand new, fully updated to 66-21 as explained here: https://wiki.freepbx.org/display/PPS/FreePBX-Distro-10.13.66

I saw a post by Andrew Nagy that said to do a voicemail reload in asterisk cli. Here's what I get:

[root@v12 ~]# asterisk -rvvvvvvv
Asterisk 13.17.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
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.
=========================================================================
Connected to Asterisk 13.17.0 currently running on v12 (pid = 5079)
v12*CLI> voicemail reload
Reloading voicemail configuration...
== Parsing '/etc/asterisk/voicemail.conf': Found
v12*CLI> exit
Asterisk cleanly ending (0).
Executing last minute cleanups
[root@v12 ~]#

Looks a bit shorter that what Andrew mentioned herer: https://community.freepbx.org/t/voicemail-not-working/31180/9

Any ideas?

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@bksales wrote:

We have several queues and several employees that work remotely sometimes and for one reason or another are forwarding/following out to their cell phones rather than using a softphone I want to create some kind of interactive menu that will prompt them for their extension number and then log them in or out of all the queues for which they are dynamic agents.

If they log into the queues via isymphony and their phone in the office becomes unreachable follow me and forwarding don't get respected for calls that go through the queue, but logging in with *45 does. I thought about just making a miscellaneous destination and pointing at them with IVR entries but the employee extensions change often enough I'd rather not have to update the IVR and miscellaneous destinations constantly.

Any tips? Feel free to just link to some article where this is already done. Thanks.

FreePBX 13.0.192.8

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@eikaf wrote:

Hi guys,
I'm new to this forum. I know this is Freepbx Forum and i don't know if this is right subforum.

1) Some months ago i bought three Fanvil X4.
Yesterday i upgraded the firmware to latest available version 2.2.0.3685.
Reading the release notes, I saw they updated and improved LDAP search. Unfortunately, even if LDAP phonebook is working great, phones do not manage to lookup CID directly from LDAP anymore. Before upgrading, all was working like a charm.

Do you have some advice to fix the problem?

Thanks.

I wrote to Fanvil support e-mail, hoping they would answer my question.

2) By now i solved with CID Superfecta looking up through LDAP Directory for inbound calls.
Does someone know if there is the chance to implement the lookup for outbound calls (dialed numbers).
I mean, should it be possible to lookup the dialed number and make it display on the phone? Something like reverse CID lookup.

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@oliverjkb wrote:

Hi everyone,

I am currently evaluating freepbx as a replacement for our "askozia" pbx system, since the company was just recently bought by 3CX.
I am really happy about the sheer amount of possibilites that freepbx offers and am coming along very well so far.

I am not completely new to asterisk, but till now I never really had to dig deeper than the web interface or the asterisk cli.
Right now I am trying to connect our odoo crm to freepbx. This is a core functionality that we use with our current pbx.
Our crm uses the akretion telephony plugin (which I am not allowed to post a Link to )

I can identify incoming calls already and see customer Information as soon as I take the call.
However, I can not use the click2dial feature hence not originate calls.
The ami returns the error message "extension doesn't exist".

From what I understand, this has to do with the context that I try to use.
The former context was in the style of "CFE-XXXXXX-frominternalandexternal".
I figured out, that I can view the dialplan with "dialplan show" from the asterisk cli and found my own extension in the context "ext-local".
Here is a screenshot for reference:

asterisk_settings_odoo.png877x486 24.8 KB

Now, when I use "ext-local" als the context to be used by the connector, the error message remains the same.

Is there a good way to narrow down the reason for this error message?
I am trying to avoid digging into textbooks etc. at this early stage, but understand that I surely have to, once I decide to go with freepbx.

Thanks a lot!

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@dcitelecom wrote:

If I understand correctly, the first lookup source should be the superfecta cache and anything not in the cache is looked up and then stored in the cache but superfecta stores all results in the cache even if it does not make sense like "ODN" or "WSHNGTNZN1, DC" and then next time the same number calls, superfecta displays the chache result instead of doing a lookup.

Is there any way to cleanup the cache data or set it to only save actual names instead of city or location info?

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@fboyd wrote:

Hello! Odd issue here, randomly when calling cell phones that have a ringback tone instead of a normal ring the PBX plays its own hold music until the call is answered. This doesn't happen on every call to the number, but randomly happens. What is very odd is that it only happens when calling a number which has a ringback and does not have a normal ring. Does anyone know what could be causing this? I seem to be getting the 180/183 messages back during the initial call setup, so I'm not sure why the system would be playing the hold music other than no media is actually being delivered, however that is very difficult to tell with the hold music playing. At first I thought it may be because of early media, or reinvites, so I tried the early media settings (prematuremedia=no progressinband=yes) and still was able to reproduce the problem regularly. Any setting change I could make to prevent this?

I've attached a log of a call in which this happens, I do see during the call it actually looks like the call is put on hold until the party answers:
--- (12 headers 11 lines) ---
-- Started music on hold, class 'default', on channel 'SIP/999-000134c0'
sip_route_dump: route/path hop:
-- SIP/sips-000134c1 is making progress passing it to SIP/999-000134c0

EDIT:
I had to post the call log in a reply below

EDIT 2:
Talked to someone at verizon, they say that this is probably happening because when you call one of their customers that has a ringback there can sometimes be a longer than normal "pause", the pause is probably what is causing the system to kick in the hold music is what I am guessing, but I have no idea how to tell the system to wait longer.

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@SmithErick wrote:

Updated some modules and upon returning to the dashboard the system overview widget wasn't displaying and I recieved the following error in a red popup:
Whoops\Exception\ErrorException Call to undefined method Error::singleton() File:/var/www/html/admin/modules/dashboard/classes/phpsysinfo/includes/output/class.Output.inc.php:40

Any insight?

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