@alenguav wrote:

I currently have an Elastix PBX running on a Digital Ocean VPS
that would like to replace with a fresh installation of FreePBX 14
however I can’t boot from the ISO located here
so what alternatives do I have appart of the installation procedures published on the wiki?

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@eeverson wrote:

Hi,

I am using a vanilla build of Asterisk along with some Sangoma S500 phones. All I need to get working is a phone directory function on the phones so that users can find and dial the correct contact numbers.

I have used the LDAP directory and this is just far too slow for use.

The XML Directory although exists as a menu item within the phone configuration is not supported and there is no documentation as to the correct XML format to even try it.

Asterisk 'voice' directory where you call the number and search for the correct extension works but again is a bit too slow for everyday use.

The only option seems to be using the App that is part of EPM... So here is the question(s)

1 - Can I install EPM without installing FreePBX?
2- Is there another way to get a phone directory on the Sangoma S500

Cheers
Euge

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@sorvani wrote:

I upgraded my FreePBX 13 instance to FreePBX 14 on Friday night. I have a weird issue with TLS registration not working well, but otherwise everything seemed to be working.

Today I finally setup my UCP because I will be working from the back porch for a while and I use the UCP to start calls (I wear a DECT Headset).

Well the UCP works and I setup the widgets I want, but the data is all from prior to the upgrade.

This is a small system and I can easily blow it up and install clean, but I wanted to test the upgrade process.

image.png1101x774 45.6 KB

I also no longer see a way to dial from the UCP. So I will have to use FOP2 for that.

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@expelliamus wrote:

I have a time group, used elsewhere.
My single outgoing route marked "permanent" and was working fine till yesterday. My users told me "no outgoing calls".
My outgoing route suddenly stoped working!!!!
After some time spent on investigation I found that some of the module autoupdates (probably) broken my PBX.
Here what I have now:
1. my outgoing route marked "permanent" this is IMPORTANT and it is not working
2. open Applications -> Time conditions
3. click Linked Item "Time Group"
4. GOT AN EMPTY TIMEGROUP IN USE BY MY OUTGOING ROUTE!!!

LOL, SANGOMA I had a huge frustration, thinking I was "hacked" or something like this

some additional info:
SNG7 OS, with all updates (autoupdates was turned on till yesterday)
that single time group used in time condition in INBOUND routing and working fine, it is 08:00-18:00 for incoming call routing
workaround make new time group 00:00-23:59 and assign it to my OUTGOING ROUTE, because permanent route setting pulling my 08:00-18:00 time group

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@johnpotter7 wrote:

Hi,

Using FreePBX 13.0.192.8 for several months now without any issues, it's great. I have a new User requirement which I'm struggling to configure so if anyone could point me in the right direction it would be greatly appreciated.

I have an incoming number routed to a ring group. The ring group is configured to always ring the first extension in the group unless it is busy, in which case it will ring the next non busy extension in the group, i.e firstavailable ring strategy. This works fine.

The new requirement I'm trying to configure is that out of hours calls to the ring group can be immediately forwarded to a cell phone. Users must be able to change the cell phone destination remotely when they swap who is on call. I am trying to achieve this via UCP and the Call Forwarding or Follow me feature.

I've configured UCP and tried setting up Unconditional Call Forwarding of the first extension in the ring group but incoming calls ring the next available extension in the ring group. I've also tried setting up Follow Me on the first extension in the ring group with 'Ring Extension First' set to 0 seconds but this seems to make no difference and incoming calls still ring the first extension in the ring group. I suspect the behaviour described above is as designed.

Can anyone please point me in the direction to achieve what I'm trying to do ?

Any help greatly appreciated.

Thanks....John.

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@a5t1 wrote:

Scenario:

We have doctors who want to be able to dial an extension, say 5555. That extension should ring the receptionist with a caller ID of "Black File". Code word for something bad is happening. Easy, right?

The problem is we have one PBX and multiple remote locations. Phones are tied via VPN back to the PBX. If Doctor at location "A" calls 5555, then receptionist at location "A" should be notified.

If they call call 5555 at location "B" then receptionist at "B" should be notified.

I've changed outbound routes and only allow certain extensions access to that route. It works by setting the destination on congestion but I cannot get the caller ID to change. Any thoughts?

We want to keep the same "extension" (really a ring group) for consistency.

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@alenguav wrote:

Hello,
I am currently testing the VULTR feature for booting from a custom ISO to install version 14 of FreePBX
however although the process of mounting ISO and booting is working perfectly, it seems like the installation
process get stuck in step 571 of 649 as you can see in the attached picture

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@alenguav wrote:

I am setting up a new PBX but after forwarding the incoming SIP calls to my PBX IP
I am getting these error lines in Asterisk log

log_failed_request: Request 'INVITE' from '' failed for '200.126.52.37:5060' (callid: 252483460_13691946@200.126.52.37) - No matching endpoint found
log_failed_request: Request 'INVITE' from '' failed for '204.10.205.149:5060' (callid: 1493478672_65823245@204.10.205.149) - No matching endpoint found
log_failed_request: Request 'INVITE' from '"101" ' failed for '158.69.251.14:5070' (callid: a6923d0b94488185c4426e998b7433b5) -

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@alisaspinelle wrote:

I'm hoping someone can help me with an issue that we're having. We just switched over to using FreePBX about a month ago, and generally, everything is going well, but I have one user who says that she's randomly getting complaints from callers that they can't hear her, that she's muffled, or cutting in and out, that kind of thing. She's the only user out of 16 that is having this issue and I can't figure out how to track it down.

We're on FreePBX version 13, Asterisk version 13, and it's hosted on Vultr. I've got our voice and data completely separated, so we have a cable modem, router, and switch specifically dedicated to voice traffic only. I've swapped out her phone with no improvement. I've tried plugging her phone into 3 other jacks in her office. I've cable tested both the cables used to connect the phone to the jack, as well as the connection from the jack to the patch panel, and found no connection issues. The router is a Ubiquiti EdgeRouter Lite and the switch is a Ubiquiti 24 port POE EdgeSwitch. I don't have a VLAN set up since all traffic is voice, so it didn't seem to be necessary. Our phones are all Sangoma S500s. Anyone got any recommendations? I'm at a loss!

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@antont17 wrote:

Hi All,
ive been trying to figure this issue out for a few hours now and im almost ready to give up. im trying to get set up a cloud pbx on googles compute platform with no luck.
im using the latest freepbx distro. and voip.ms is my trunk.
Im able to get my phone connected to the pbx and can dial out, but I have no audio both ways. out of desperation i created an any any rule in the firewall for testing and still no luck.
here is my log:

[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [6475395971@from-internal:1] Macro("SIP/100-00000004", "user-callerid,LIMIT,EXTERNAL,") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:1] Set("SIP/100-00000004", "TOUCH_MONITOR=1503699693.4") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:2] Set("SIP/100-00000004", "AMPUSER=100") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:3] GotoIf("SIP/100-00000004", "0?report") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:4] ExecIf("SIP/100-00000004", "1?Set(__REALCALLERIDNUM=100)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:5] Set("SIP/100-00000004", "AMPUSER=100") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:6] GotoIf("SIP/100-00000004", "0?limit") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:7] Set("SIP/100-00000004", "AMPUSERCIDNAME=Anton") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:8] GotoIf("SIP/100-00000004", "0?report") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:9] Set("SIP/100-00000004", "AMPUSERCID=100") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:10] Set("SIP/100-00000004", "_DIALOPTIONS=Ttr") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:11] Set("SIP/100-00000004", "CALLERID(all)="Anton" <100>") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:12] GotoIf("SIP/100-00000004", "0?limit") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:13] ExecIf("SIP/100-00000004", "1?Set(GROUP(concurrency_limit)=100)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:14] ExecIf("SIP/100-00000004", "0?Set(CHANNEL(language)=)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:15] GotoIf("SIP/100-00000004", "1?continue") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx_builtins.c: Goto (macro-user-callerid,s,29)
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:29] Set("SIP/100-00000004", "CALLERID(number)=100") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:30] Set("SIP/100-00000004", "CALLERID(name)=Anton") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:31] GotoIf("SIP/100-00000004", "0?cnum") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:32] Set("SIP/100-00000004", "CDR(cnam)=Anton") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:33] Set("SIP/100-00000004", "CDR(cnum)=100") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-user-callerid:34] Set("SIP/100-00000004", "CHANNEL(language)=en") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [6475395971@from-internal:2] Gosub("SIP/100-00000004", "sub-record-check,s,1(out,6475395971,dontcare)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@sub-record-check:1] GotoIf("SIP/100-00000004", "0?initialized") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@sub-record-check:2] Set("SIP/100-00000004", "_RECSTATUS=INITIALIZED") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@sub-record-check:3] Set("SIP/100-00000004", "NOW=1503699693") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@sub-record-check:4] Set("SIP/100-00000004", "__DAY=25") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@sub-record-check:5] Set("SIP/100-00000004", "__MONTH=08") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@sub-record-check:6] Set("SIP/100-00000004", "__YEAR=2017") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@sub-record-check:7] Set("SIP/100-00000004", "__TIMESTR=20170825-182133") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@sub-record-check:8] Set("SIP/100-00000004", "__FROMEXTEN=100") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@sub-record-check:9] Set("SIP/100-00000004", "_MONFMT=wav") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@sub-record-check:10] NoOp("SIP/100-00000004", "Recordings initialized") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@sub-record-check:11] ExecIf("SIP/100-00000004", "0?Set(ARG3=dontcare)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@sub-record-check:12] Set("SIP/100-00000004", "REC_POLICY_MODE_SAVE=") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@sub-record-check:13] ExecIf("SIP/100-00000004", "0?Set(REC_STATUS=NO)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@sub-record-check:14] GotoIf("SIP/100-00000004", "3?checkaction") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx_builtins.c: Goto (sub-record-check,s,17)
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@sub-record-check:17] GotoIf("SIP/100-00000004", "1?sub-record-check,out,1") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx_builtins.c: Goto (sub-record-check,out,1)
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [out@sub-record-check:1] NoOp("SIP/100-00000004", "Outbound Recording Check from 100 to 6475395971") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [out@sub-record-check:2] Set("SIP/100-00000004", "RECMODE=dontcare") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [out@sub-record-check:3] ExecIf("SIP/100-00000004", "1?Goto(routewins)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx_builtins.c: Goto (sub-record-check,out,7)
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [out@sub-record-check:7] Gosub("SIP/100-00000004", "recordcheck,1(dontcare,out,6475395971)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp("SIP/100-00000004", "Starting recording check against dontcare") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [recordcheck@sub-record-check:2] Goto("SIP/100-00000004", "dontcare") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx_builtins.c: Goto (sub-record-check,recordcheck,3)
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [recordcheck@sub-record-check:3] Return("SIP/100-00000004", "") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [out@sub-record-check:8] Return("SIP/100-00000004", "") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [6475395971@from-internal:3] ExecIf("SIP/100-00000004", "0 ?Set(CDR(accountcode)=)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [6475395971@from-internal:4] Set("SIP/100-00000004", "MOHCLASS=default") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [6475395971@from-internal:5] Set("SIP/100-00000004", "_NODEST=") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [6475395971@from-internal:6] Macro("SIP/100-00000004", "dialout-trunk,1,6475395971,,off") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:1] Set("SIP/100-00000004", "DIAL_TRUNK=1") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:2] GosubIf("SIP/100-00000004", "0?sub-pincheck,s,1()") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:3] GotoIf("SIP/100-00000004", "0?disabletrunk,1") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:4] Set("SIP/100-00000004", "DIAL_NUMBER=6475395971") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:5] Set("SIP/100-00000004", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:6] Set("SIP/100-00000004", "OUTBOUND_GROUP=OUT_1") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:7] GotoIf("SIP/100-00000004", "1?nomax") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx_builtins.c: Goto (macro-dialout-trunk,s,9)
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:9] GotoIf("SIP/100-00000004", "0?skipoutcid") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:10] Set("SIP/100-00000004", "DIAL_TRUNK_OPTIONS=T") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:11] Macro("SIP/100-00000004", "outbound-callerid,1") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:1] ExecIf("SIP/100-00000004", "0?Set(CALLERPRES(name-pres)=)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:2] ExecIf("SIP/100-00000004", "0?Set(CALLERPRES(num-pres)=)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:3] ExecIf("SIP/100-00000004", "0?Set(REALCALLERIDNUM=100)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:4] GotoIf("SIP/100-00000004", "1?normcid") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx_builtins.c: Goto (macro-outbound-callerid,s,7)
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:7] Set("SIP/100-00000004", "USEROUTCID=") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:8] Set("SIP/100-00000004", "EMERGENCYCID=") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:9] Set("SIP/100-00000004", "TRUNKOUTCID="Anton Treister" <6475395971>") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:10] GotoIf("SIP/100-00000004", "1?trunkcid") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx_builtins.c: Goto (macro-outbound-callerid,s,15)
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:15] ExecIf("SIP/100-00000004", "1?Set(CALLERID(all)="Anton Treister" <6475395971>)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:16] ExecIf("SIP/100-00000004", "0?Set(CALLERID(all)=)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:17] ExecIf("SIP/100-00000004", "0?Set(CALLERID(all)=)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:18] ExecIf("SIP/100-00000004", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:19] ExecIf("SIP/100-00000004", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:20] Set("SIP/100-00000004", "CDR(outbound_cnum)=6475395971") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-outbound-callerid:21] Set("SIP/100-00000004", "CDR(outbound_cnam)=Anton Treister") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:12] GosubIf("SIP/100-00000004", "1?sub-flp-1,s,1()") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@sub-flp-1:1] ExecIf("SIP/100-00000004", "0?Return()") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@sub-flp-1:2] ExecIf("SIP/100-00000004", "1?Return()") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:13] Set("SIP/100-00000004", "OUTNUM=6475395971") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:14] Set("SIP/100-00000004", "custom=SIP/VoipMs") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:15] ExecIf("SIP/100-00000004", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:16] ExecIf("SIP/100-00000004", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:17] Macro("SIP/100-00000004", "dialout-trunk-predial-hook,") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/100-00000004", "") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:18] GotoIf("SIP/100-00000004", "0?skipcrm") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:19] Set("SIP/100-00000004", "_CRMDIRECTION=OUTBOUND") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:20] Set("SIP/100-00000004", "_CRMDESTINATION=6475395971") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:21] Set("SIP/100-00000004", "_CRMSOURCE=100") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:22] AGI("SIP/100-00000004", "sangomacrm.agi") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] res_agi.c: AGI Script sangomacrm.agi completed, returning 0
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:23] Set("SIP/100-00000004", "CHANNEL(hangup_handler_push)=crm-hangup,s,1") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:24] NoOp("SIP/100-00000004", "CRM Finished") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:25] GotoIf("SIP/100-00000004", "0?bypass,1") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:26] ExecIf("SIP/100-00000004", "1?Set(CONNECTEDLINE(num,i)=6475395971)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:27] ExecIf("SIP/100-00000004", "1?Set(CONNECTEDLINE(name,i)=CID:6475395971)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:28] ExecIf("SIP/100-00000004", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)6475395971)") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:29] GotoIf("SIP/100-00000004", "0?customtrunk") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-dialout-trunk:30] Dial("SIP/100-00000004", "SIP/VoipMs/6475395971,300,T") in new stack
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] netsock2.c: Using SIP RTP TOS bits 184
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] netsock2.c: Using SIP RTP CoS mark 5
[2017-08-25 18:21:33] VERBOSE[13151][C-00000002] app_dial.c: Called SIP/VoipMs/6475395971
[2017-08-25 18:21:37] VERBOSE[13151][C-00000002] app_dial.c: SIP/VoipMs-00000005 is making progress passing it to SIP/100-00000004
[2017-08-25 18:21:42] VERBOSE[13151][C-00000002] app_dial.c: SIP/VoipMs-00000005 answered SIP/100-00000004
[2017-08-25 18:21:42] VERBOSE[13166][C-00000002] bridge_channel.c: Channel SIP/VoipMs-00000005 joined 'simple_bridge' basic-bridge <0a7bde78-fb0f-4ef0-b722-5d55431a3d97>
[2017-08-25 18:21:42] VERBOSE[13151][C-00000002] bridge_channel.c: Channel SIP/100-00000004 joined 'simple_bridge' basic-bridge <0a7bde78-fb0f-4ef0-b722-5d55431a3d97>
[2017-08-25 18:21:48] WARNING[3021] chan_sip.c: Retransmission timeout reached on transmission 1400274457-5060-6@BJC.BGI.A.BBB for seqno 31 (Critical Response) -- See
Packet timed out after 6401ms with no response
[2017-08-25 18:21:48] WARNING[3021] chan_sip.c: Hanging up call 1400274457-5060-6@BJC.BGI.A.BBB - no reply to our critical packet (see ).
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] bridge_channel.c: Channel SIP/100-00000004 left 'simple_bridge' basic-bridge <0a7bde78-fb0f-4ef0-b722-5d55431a3d97>
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] app_macro.c: Spawn extension (macro-dialout-trunk, s, 30) exited non-zero on 'SIP/100-00000004' in macro 'dialout-trunk'
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] pbx.c: Spawn extension (from-internal, 6475395971, 6) exited non-zero on 'SIP/100-00000004'
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] pbx.c: Executing [h@from-internal:1] Macro("SIP/100-00000004", "hangupcall") in new stack
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("SIP/100-00000004", "1?theend") in new stack
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2017-08-25 18:21:48] VERBOSE[13166][C-00000002] bridge_channel.c: Channel SIP/VoipMs-00000005 left 'simple_bridge' basic-bridge <0a7bde78-fb0f-4ef0-b722-5d55431a3d97>
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("SIP/100-00000004", "0?Set(CDR(recordingfile)=)") in new stack
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] pbx.c: Executing [s@macro-hangupcall:4] Hangup("SIP/100-00000004", "") in new stack
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/100-00000004' in macro 'hangupcall'
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-00000004'
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] app_stack.c: SIP/100-00000004 Internal Gosub(crm-hangup,s,1) start
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] pbx.c: Executing [s@crm-hangup:1] NoOp("SIP/100-00000004", "Sending Hangup to CRM") in new stack
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] pbx.c: Executing [s@crm-hangup:2] NoOp("SIP/100-00000004", "HANGUP CAUSE: 18") in new stack
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] pbx.c: Executing [s@crm-hangup:3] ExecIf("SIP/100-00000004", "0?Set(_CRMVOICEMAIL=)") in new stack
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] pbx.c: Executing [s@crm-hangup:4] NoOp("SIP/100-00000004", "MASTER CHANNEL: 1503699693.4 = 1503699693.4") in new stack
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] pbx.c: Executing [s@crm-hangup:5] GotoIf("SIP/100-00000004", "0?return") in new stack
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] pbx.c: Executing [s@crm-hangup:6] Set("SIP/100-00000004", "_CRMHANGUP=1") in new stack
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] pbx.c: Executing [s@crm-hangup:7] AGI("SIP/100-00000004", "sangomacrm.agi") in new stack
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] res_agi.c: AGI Script sangomacrm.agi completed, returning 0
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] pbx.c: Executing [s@crm-hangup:8] Return("SIP/100-00000004", "") in new stack
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-00000004'
[2017-08-25 18:21:48] VERBOSE[13151][C-00000002] app_stack.c: SIP/100-00000004 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

any help would be greatly appreciated.
thanks

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@datazero wrote:

hi everyone,
I have installed latest stable freepbx distro, Sangoma A200 (2xFXS, 2xFXO) and Sangoma S500 phones.
I have not inbound caller identification through the FXO ports. When a new inbound call arrives, only the number of the caller is displayed.
Through "Contact Manager" i have created an external group with some contacts.
I have also create a CIDLookup source with the external group.
In the dahdi trunks i allow any CID and the "Hide CallerID" is OFF.
Any help?
Rgrds

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@Wallyrt wrote:

Hi,

I try To install the actual FREEPBX Distro on an APU2C Board via a serial Null Modem Cable. I have saved the ISO to an USB Stick and when I try to install and after I select the USB Stick to boot the screen will be stay blank and Nothing happens anymore.

Is there somebody around who can help me to solve this problem or is it possible anyway ?

Greetings
Wallyrt

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@bnguyen654 wrote:

Hello all,

When running the ./install -n command, this is what I get:

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...Yes. Determined Asterisk version to be: 14.6.0
Preliminary checks done. Starting FreePBX Installation
Checking if this is a new install...Yes (No /etc/freepbx.conf file detected)
Database installation checking credentials and permissions..Connected!
Empty asterisk_cdrdb Database going to populate it
Initializing FreePBX Settings


  [Exception]
  fatal error reading freepbx_settings::


install [--dbengine DBENGINE] [--dbname DBNAME] [--cdrdbname CDRDBNAME] [--dbuser DBUSER] [--dbpass DBPASS] [--user USER] [--group GROUP] [--dev-links] [--skip-install] [--webroot WEBROOT] [--astetcdir ASTETCDIR] [--astmoddir ASTMODDIR] [--astvarlibdir ASTVARLIBDIR] [--astagidir ASTAGIDIR] [--astspooldir ASTSPOOLDIR] [--astrundir ASTRUNDIR] [--astlogdir ASTLOGDIR] [--ampbin AMPBIN] [--ampsbin AMPSBIN] [--ampcgibin AMPCGIBIN] [--ampplayback AMPPLAYBACK] [-r|--rootdb] [-f|--force]

I have looked around with not much luck.

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@alenguav wrote:

Hello,
I am receiving this error message when forwarding a call to my cellphone. Any ideas?
(I already have an outbound route)

-- Executing [in@sub-record-check:1] NoOp("PJSIP/Convergia-00000010", "Inbound Recording Check to 16401230") in new stack
-- Executing [in@sub-record-check:2] Set("PJSIP/Convergia-00000010", "FROMEXTEN=unknown") in new stack
-- Executing [in@sub-record-check:3] ExecIf("PJSIP/Convergia-00000010", "8?Set(FROMEXTEN=13480230)") in new stack
-- Executing [in@sub-record-check:4] Gosub("PJSIP/Convergia-00000010", "recordcheck,1(dontcare,in,16401230)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/Convergia-00000010", "Starting recording check against dontcare") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("PJSIP/Convergia-00000010", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [recordcheck@sub-record-check:3] Return("PJSIP/Convergia-00000010", "") in new stack
-- Executing [in@sub-record-check:5] Return("PJSIP/Convergia-00000010", "") in new stack
-- Executing [16401230@from-pstn:3] Gosub("PJSIP/Convergia-00000010", "app-blacklist-check,s,1()") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("PJSIP/Convergia-00000010", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("PJSIP/Convergia-00000010", "CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("PJSIP/Convergia-00000010", "") in new stack
-- Executing [16401230@from-pstn:4] Set("PJSIP/Convergia-00000010", "__FROM_DID=16401230") in new stack
-- Executing [16401230@from-pstn:5] Set("PJSIP/Convergia-00000010", "CDR(did)=16401230") in new stack
-- Executing [16401230@from-pstn:6] ExecIf("PJSIP/Convergia-00000010", "1 ?Set(CALLERID(name)=13480230)") in new stack
-- Executing [16401230@from-pstn:7] Set("PJSIP/Convergia-00000010", "__MOHCLASS=") in new stack
-- Executing [16401230@from-pstn:8] Set("PJSIP/Convergia-00000010", "__REVERSAL_REJECT=FALSE") in new stack
-- Executing [16401230@from-pstn:9] GotoIf("PJSIP/Convergia-00000010", "1?post-reverse-charge") in new stack
-- Goto (from-pstn,16401230,11)
-- Executing [16401230@from-pstn:11] NoOp("PJSIP/Convergia-00000010", "") in new stack
-- Executing [16401230@from-pstn:12] Set("PJSIP/Convergia-00000010", "__CALLINGNAMEPRES_SV=allowed_not_screened") in new stack
-- Executing [16401230@from-pstn:13] Set("PJSIP/Convergia-00000010", "__CALLINGNUMPRES_SV=allowed_not_screened") in new stack
-- Executing [16401230@from-pstn:14] Set("PJSIP/Convergia-00000010", "CALLERID(name-pres)=allowed_not_screened") in new stack
-- Executing [16401230@from-pstn:15] Set("PJSIP/Convergia-00000010", "CALLERID(num-pres)=allowed_not_screened") in new stack
-- Executing [16401230@from-pstn:16] NoOp("PJSIP/Convergia-00000010", "CallerID Entry Point") in new stack
-- Executing [16401230@from-pstn:17] Set("PJSIP/Convergia-00000010", "__CRM_DIRECTION=INBOUND") in new stack
-- Executing [16401230@from-pstn:18] Set("PJSIP/Convergia-00000010", "__CRM_SOURCE=13480230") in new stack
-- Executing [16401230@from-pstn:19] Set("PJSIP/Convergia-00000010", "__CRM_LINKEDID=1503830202.22") in new stack
-- Executing [16401230@from-pstn:20] ExecIf("PJSIP/Convergia-00000010", "1?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
-- Executing [16401230@from-pstn:21] Goto("PJSIP/Convergia-00000010", "ext-miscdests,1,1") in new stack
-- Goto (ext-miscdests,1,1)
-- Executing [1@ext-miscdests:1] NoOp("PJSIP/Convergia-00000010", "MiscDest: Celular Ale") in new stack
-- Executing [1@ext-miscdests:2] Goto("PJSIP/Convergia-00000010", "from-internal,993055623,1") in new stack
-- Goto (from-internal,993055623,1)
-- Executing [993055623@from-internal:1] ResetCDR("PJSIP/Convergia-00000010", "") in new stack
-- Executing [993055623@from-internal:2] NoCDR("PJSIP/Convergia-00000010", "") in new stack
-- Executing [993055623@from-internal:3] Progress("PJSIP/Convergia-00000010", "") in new stack
-- Executing [993055623@from-internal:4] Wait("PJSIP/Convergia-00000010", "1") in new stack
   > 0x7f3ff40b2820 -- Probation passed - setting RTP source address to 200.126.52.36:16976
-- Executing [993055623@from-internal:5] Playback("PJSIP/Convergia-00000010", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- <PJSIP/Convergia-00000010> Playing 'silence/1.alaw' (language 'en')
-- <PJSIP/Convergia-00000010> Playing 'cannot-complete-as-dialed.alaw' (language 'en')
-- <PJSIP/Convergia-00000010> Playing 'check-number-dial-again.alaw' (language 'en')
-- Executing [993055623@from-internal:6] Wait("PJSIP/Convergia-00000010", "1") in new stack
-- Executing [993055623@from-internal:7] Congestion("PJSIP/Convergia-00000010", "20") in new stack
  == Spawn extension (from-internal, 993055623, 7) exited non-zero on 'PJSIP/Convergia-00000010'
-- Executing [h@from-internal:1] Macro("PJSIP/Convergia-00000010", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/Convergia-00000010", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/Convergia-00000010", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("PJSIP/Convergia-00000010", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'PJSIP/Convergia-00000010' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/Convergia-00000010'

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@StephanK wrote:

Hi,

We are having trouble with SMS messages through the UCP. Often we have partial messages and loose the rest of the text. Debugging showed that the messages arrive correctly on our Freepbx system, just do not end up in the UCP.

Is there anybody else using this? Are you having problems?

I thought the UCP SMS SipStation was not public, but when I contacted the Sagoma support, they could not help me.

Is there and other sip-carrier hat has a good SMS/MMS client?

Thanks,

Stephan
PS: We are still on Version 13.

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@Bindu wrote:

I am new to FreePBX, I was able to login into FreePBX server using c#. Can anyone please provide me the information to get the caller details like caller phone number, call Id using c#. I checked with the below code snippet but I get a null value when there is an ongoing call. I have used Aster.NET library.
Please required the suggestion for the above implementation.

NewExtenEvent extnEvents = new NewExtenEvent(manager);
NewChannelEvent channelEvents = new NewChannelEvent(manager);
string callerNumber=channelEvents.CallerIdNum;
NewCallerIdEvent callEvents = new NewCallerIdEvent(manager);

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@Cheetor wrote:

Hi There,

We have a problem with our server, coming from a PIAF/ Freepbx 12 server , we are upgrading to a new server Freepbx 14 Asterisk 13, we loaded all our trunks,extensions,inbound routes.

we tested and can register a chansip account and place inbound and outbound calls,problem is we have about 1200 Clients. but we switched over to the new server and registered the trunks and clients. but it seems that asterisk keeps throwing clients unreachable then reachable again. and it seems it wont register more than 512 clients at once, so that is 600-700 clients offline , the server then seems to go unstable and freeze the webmin, then you have to stop asterisk to log in again.

it cant be the server specs as it is:
Intel Wildcat rack mount server with a intel server board s1200btl
2x xeon E5-2640V4
161 GB Samsung DDR4 Ram
2x 960GB SSD HDD runing in hardware mirror raid

If we switch the server back to our old server with the exact same public ip . all the clients register again. obviously both server are not plugged in at once.

Please can someone help. we do have logs.

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@alenguav wrote:

I was wondering if there is any special step for installing the g729 codec and license in a fresh installation of FREEPBX 14

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@wa4zlw wrote:

what are the vairables used by Superfecta so I can tailor the emailed message to have other information? Where can this be found?

Thanks leon

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@jsilva wrote:

I am provisioning AudioCodes 310HD and after configured the equipment have the following faults. When you redial any historical call it displays the following message on the "Calling 000" screen, I notice that configured manual does not happen.

Thanks
Janduy

Below a picture of what happens!

Screen Shot 2017-08-28 at 3.55.03 PM.png684x533 222 KB

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