@sk_paruthi wrote:

Hello,
I want to set up an IVR in FreePBX and then export the data records of keys pressed by the users for each call in a day-wise file.
How can I set this up functionality…
Please help.

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@PoDuck wrote:

I have multiple phones in a ring group, but when I try to add a cell phone to an extension’s follow me, it won’t call it. I have read in other posts about putting pound signs after extensions, putting the cell phone first in the follow me list, etc., but none of that is working.

If I put the cell phone number in the ring group, it will dial it. If I put the cell phone in the follow me on an extension, it will ring it, but if I put the cell phone in the follow me on an extension, and place that extension in a ring group, it won’t follow.

As an example:

Ring group:
2101#

Extension 2101 follow me:
5555551212#
2101

Anyway, what do I need to do to get this to work correctly?

Thanks

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@keh192 wrote:

Hi everyone. I need to hide incoming CallerID’s for all my extensions.

When I calls on my FreePBX phone I see incoming number on my softfone, but I need hiding wthis number an type: unknown or everyone else for all extensoins in my FreePBX

How can I do this? Please help.

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@faisalkhan wrote:

Hi All,

need help.

I want to implement a solution where a specified list of callers when call comes from these known callers it will be routed to a specific queue and the rest goes to general queue.

I want to implement this for Inbound route.

Any suggestions will be highly appreciated.

Thanks in advance.

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@moodinsk wrote:

Greetings,

I’ve been battling an issue that I cannot quite put my thumb on. On my most recent install for a customer, they started complaining of their phones “locking up”. As I have looked into it, i have found that the PBX is sending a ridiculous amount of SIP notify packets to the endpoints, which causes their CPU’s to spike to 100% and become pretty much unresponsive. I have 50MB wireshark dumps rotating on eth0, and when it starts ramping up, it will fill a 50MB capture file about once a minute… approximately 70k SIP NOTIFY packets.

I have ruled out loop onsite, replaced all of their switches, phones, and firewall. I have a phone registered to it remotely over a site-to-site VPN and that phone also experiences this, so it doesnt appear to be caused by anything on the customer site. I have even setup another brand new image and reconfigured everything from scratch for them and the same thing happens. I may even try using an old distro ISO to see if the issue follows me there…

The PBX exists in our “cloud”, which is a hyperV failover cluster. None of the other images have this issue that i know of. I am currently firing up another image with the latest CentOS7.6 GA that was just released to see if it happens there as well… i know more about that later.

Doing a fwconsole restart, will temporarily stop it, but then it starts back up again soon after.

I am using Polycom 410’s, I have tried several different firmwares and it happens on all of them, so I dont believe it’s the phones that are causing it. I have other semi-recent installs for other customers that do not experience this issue, so I have to think that it is something to do with a recent module update and/or asterisk 13.22… but I have no specific proof… just a hunch.

I have updated to the edge track for Core and Framework to see if that makes a difference, so far no help there.

Here is my environment:

FPBX Distro
12.7.6-1904-1.sng7
FreePBX 14.0.12.10
Asterisk: 13.22

[root@localhost~]# fwconsole ma list
No repos specified, using: [standard,extended,commercial] from last GUI settings

+----------------------+------------+---------+------------+
| Module               | Version    | Status  | License    |
+----------------------+------------+---------+------------+
| accountcodepreserve  | 13.0.2.2   | Enabled | GPLv2      |
| announcement         | 13.0.7.7   | Enabled | GPLv3+     |
| arimanager           | 13.0.5.2   | Enabled | GPLv3+     |
| asterisk-cli         | 14.0.1     | Enabled | GPLv3+     |
| asteriskinfo         | 13.0.7.1   | Enabled | GPLv3+     |
| backup               | 14.0.10.3  | Enabled | GPLv3+     |
| blacklist            | 14.0.2     | Enabled | GPLv3+     |
| builtin              |            | Enabled |            |
| bulkhandler          | 13.0.14.8  | Enabled | GPLv3+     |
| calendar             | 14.0.2.16  | Enabled | GPLv3+     |
| callback             | 13.0.5.4   | Enabled | GPLv3+     |
| callforward          | 14.0.1.3   | Enabled | AGPLv3+    |
| callrecording        | 14.0.14    | Enabled | AGPLv3+    |
| callwaiting          | 14.0.1.1   | Enabled | GPLv3+     |
| cdr                  | 14.0.5.19  | Enabled | GPLv3+     |
| cel                  | 14.0.2.12  | Enabled | GPLv3+     |
| certman              | 14.0.3.2   | Enabled | AGPLv3+    |
| cidlookup            | 14.0.1.8   | Enabled | GPLv3+     |
| conferences          | 13.0.23.15 | Enabled | GPLv3+     |
| configedit           | 13.0.7.1   | Enabled | AGPLv3+    |
| contactmanager       | 14.0.5.4   | Enabled | GPLv3+     |
| core                 | 14.0.28.9  | Enabled | GPLv3+     |
| customappsreg        | 13.0.5.7   | Enabled | GPLv3+     |
| dahdiconfig          | 14.0.1.4   | Enabled | GPLv3+     |
| dashboard            | 14.0.6.2   | Enabled | AGPLv3+    |
| daynight             | 14.0.1     | Enabled | GPLv3+     |
| dictate              | 13.0.5     | Enabled | GPLv3+     |
| digiumaddoninstaller | 13.0.1.1   | Enabled | GPLv2      |
| directory            | 13.0.19.12 | Enabled | GPLv3+     |
| disa                 | 13.0.6.12  | Enabled | AGPLv3+    |
| donotdisturb         | 14.0.1.1   | Enabled | GPLv3+     |
| dundicheck           | 2.11.0.3   | Enabled | GPLv3+     |
| endpoint             | 14.0.2.188 | Enabled | Commercial |
| extensionroutes      | 13.0.10.7  | Enabled | Commercial |
| extensionsettings    | 13.0.4     | Enabled | GPLv3+     |
| fax                  | 14.0.2.7   | Enabled | GPLv3+     |
| featurecodeadmin     | 13.0.6.4   | Enabled | GPLv3+     |
| findmefollow         | 14.0.1.23  | Enabled | GPLv3+     |
| firewall             | 13.0.57.1  | Enabled | AGPLv3+    |
| framework            | 14.0.12.10 | Enabled | GPLv2+     |
| fw_langpacks         | 14.0.1     | Enabled | GPLv3+     |
| hotelwakeup          | 14.0.1.6   | Enabled | GPLv2      |
| iaxsettings          | 14.0.1.4   | Enabled | AGPLv3     |
| infoservices         | 13.0.1.4   | Enabled | GPLv2+     |
| ivr                  | 14.0.4     | Enabled | GPLv3+     |
| languages            | 14.0.1.4   | Enabled | GPLv3+     |
| logfiles             | 13.0.10.5  | Enabled | GPLv3+     |
| manager              | 13.0.2.5   | Enabled | GPLv2+     |
| miscapps             | 13.0.3.1   | Enabled | GPLv3+     |
| miscdests            | 13.0.7     | Enabled | GPLv3+     |
| music                | 13.0.22.7  | Enabled | GPLv3+     |
| outroutemsg          | 14.0.1     | Enabled | GPLv3+     |
| paging               | 14.0.12    | Enabled | GPLv3+     |
| parking              | 13.0.19.11 | Enabled | GPLv3+     |
| pbdirectory          | 2.11.0.6   | Enabled | GPLv3+     |
| phonebook            | 13.0.6.4   | Enabled | GPLv3+     |
| phpinfo              | 13.0.2     | Enabled | GPLv2+     |
| pinsets              | 13.0.13    | Enabled | GPLv3+     |
| pm2                  | 13.0.7.1   | Enabled | AGPLv3+    |
| presencestate        | 14.0.1.7   | Enabled | GPLv3+     |
| printextensions      | 13.0.3.2   | Enabled | GPLv3+     |
| queuemetrics         | 2.11.0.3   | Enabled | GPLv3+     |
| queueprio            | 13.0.6     | Enabled | GPLv3+     |
| queues               | 14.0.2.25  | Enabled | GPLv2+     |
| recordings           | 13.0.30.13 | Enabled | GPLv3+     |
| restapi              | 13.0.21.2  | Enabled | AGPLv3     |
| ringgroups           | 14.0.1.8   | Enabled | GPLv3+     |
| setcid               | 13.0.6.3   | Enabled | GPLv3+     |
| sipsettings          | 14.0.27.12 | Enabled | AGPLv3+    |
| soundlang            | 14.0.7     | Enabled | GPLv3+     |
| speeddial            | 2.11.0.4   | Enabled | GPLv3+     |
| superfecta           | 14.0.18    | Enabled | GPLv2+     |
| sysadmin             | 14.0.33    | Enabled | Commercial |
| timeconditions       | 14.0.2.17  | Enabled | GPLv3+     |
| ucp                  | 14.0.3.3   | Enabled | AGPLv3+    |
| userman              | 14.0.3.49  | Enabled | AGPLv3+    |
| vmblast              | 13.0.11    | Enabled | GPLv3+     |
| voicemail            | 14.0.6.5   | Enabled | GPLv3+     |
| weakpasswords        | 13.0.2     | Enabled | GPLv3+     |
+----------------------+------------+---------+------------+

The phones have a pretty basic template. One Line Key for one SIP account, 2 Parking BLFs, and 4 BLF-XFER BLF’s for the 4 other extensions on the PBX (5 total PJSIP endpoints). Nothing crazy at all.

Here is a screenshot of a packet capture, I can provide the full dumpfile if needed. The capture filled with these NOTIFY packets, to each of the 5 registered endpoints.

301.png1083×1017 88.6 KB

All of the SIP Notify packets have the same Message Body, which leaves me to believe it has something to do with the voicemail module or MWI in asterisk…

Messages-Waiting: no\r\n
Voice-Message: 0/0 (0/0)\r\n

I would appreciate any additional insight as to why this is happening, I am running out of ideas here.

thanks in advance!

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@herbw wrote:

I can’t seem to find any way to get rid of this warning that has been present ever since I installed freepbx (version 14.0.11). Checking online for Module Updates shows nothing to update – only commercial modules to buy.

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@dodgly wrote:

I’ve used the Missed Call Notification third-party module written by @TSM for quite a few years. Back in September 2014 he mentioned there was a .6 version in the wings but I’ve not seen anything since. Read more on that thread at Missed Call Notification Module?.

So, I’ve started in on some improvements to the v0.5 codebase to implement some features I’ve wanted. Sending missed call notifications to email is cool and useful, but I think sending an SMS and push alert services like Pushover, PushBullet, etc. is even better!

While I’m diving in on this, are there any other interesting Missed Call Notification features that would be of interest you?

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@decibel83 wrote:

Hi,
I am using a SIP trunk from a ISDN BRI gateway and DTMF are not working in both incoming and outgoing calls.

Everything seems to be OK on the ISDN BRI gateway (unfortunately it is not good documented but I compared the configuration with another working installation) and I cannot understand if something is wrong on my FreePBX or on the BRI gateway.

dtmfmode is “rfc2833” on every trunks.

Could you help me please?
Thanks!

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@sse450 wrote:

I have installed the latest FreePBX distro with FreePBX 14.0.11 and Asterisk 13.22.0. It is working nicely.

I signed up with a trunk provider and obtained the credentials. Entered these data in Connectivity | Trunks. Also entered register string in the incoming. I made chan_sip port as 5060, pfsip as 5160. My FreePBX connects to the SIP trunk provider as seen in the Reports | Asterisk Info | Registries:
12.123.123.24:5060 N 902124000212 285 Registered Tue, 14 May 2019 18:41:10
Created some Outbound patterns in Connectivity | Outbound Route. To me, so far, so good.

My problem starts here: I cannot register any phone to my FreePBX. I have a Yealink T29G. IP of my FreePBX is 10.10.1.20. I try to register the phone using 10.10.1.20:5060. No way! Web page of the phone says “Register Failed”.

Chan_Sip Peers:
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
200 (Unspecified) D No No A 0 UNKNOWN
201 (Unspecified) D No No A 0 UNKNOWN
202 (Unspecified) D No No A 0 UNKNOWN
xyzTelecom/902124000 12.123.123.24 Yes Yes 5060 OK (4 ms)
4 sip peers [Monitored: 1 online, 3 offline Unmonitored: 0 online, 0 offline]

Furthermore, I noticed something on Dashboard. System overview box says Default bind port for CHAN_PJSIP is: 5060, CHAN_SIP is: 5160 although the bind port is 5060 on Settings | Asterisk SIP settings | ChanSIP settings.

What gives? What am I doing wrong? It is quite possible that I misunderstand the setup process.

I would appreciate any help.

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@mvogel4949 wrote:

When a caller enters a queue they can hear a variety of announcements. What number in line they are and more. Does the service level timer start prior to these announcements or after?

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@shuijsen wrote:

Hi there,

I’m trying to make a complete installation of FreePBX 14 - Asterisk 16 with PHP 7.2 at Ubuntu 18.04. I’ve made a couple of changes in some PHP files. Like changing some each lines to foreach lines. That made things work great.

Now I want to apply the config in FreePBX, and it’s giving the error:
Reload failed because retrieve_conf encountered an error: 1

When I execute the command: /var/lib/asterisk/bin/retrieve_conf it gives the following output:

image.png1889×204 14 KB

Looks like a PHP method from PHP 5 that isn’t working in PHP 7 anymore. I’m not new to PHP, but I like to hear things from you guys.

How to solve this PHP problem, and does anyone ever succeed a working installation with PHP 7?

Thanks for the answers.

Kind regards,
Serge

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@PitzKey wrote:

Hello,

In a recent update, if you have a virtual extension in a queue with follow me and confirm calls enabled, once you press 1 to accept the call, you get the “too late announcement”. So to reproduce this, I went to a server that the modules weren’t updated yet:

core 14.0.18.49
findmefollow 14.0.1.21
framework 14.0.5.25
queues 14.0.2.23
Call trace: https://pastebin.freepbx.org/view/a512ae85
macro-confirm in extensions_additional.conf: https://pastebin.freepbx.org/view/fd2244f4

Everything works great.

I went ahead and updated the modules to:
core 14.0.25.4
findmefollow 14.0.1.23
framework 14.0.11
queues 14.0.2.25

Now it’s not working…
Call trace: https://pastebin.freepbx.org/view/f871d296
Looking at the call trace you see:

[2019-05-15 08:13:24] VERBOSE[5550][C-0000908d] file.c: <Local/12128885500@from-internal-000064ab;1> Playing 'incoming-call-1-accept-2-decline.slin' (language 'en')
[2019-05-15 08:13:24] VERBOSE[5580][C-0000908d] bridge_channel.c: Channel SIP/AdTran-Primary-000109c2 joined 'simple_bridge' basic-bridge <79b06009-1719-4706-8d1e-d6bc1a0733e7>
[2019-05-15 08:13:24] VERBOSE[5559][C-0000908d] bridge_channel.c: Channel Local/12128885500@from-internal-000064ab;2 joined 'simple_bridge' basic-bridge <79b06009-1719-4706-8d1e-d6bc1a0733e7>
[2019-05-15 08:13:28] VERBOSE[5550][C-0000908d] pbx.c: Executing [1@macro-confirm:1] GotoIf("Local/12128885500@from-internal-000064ab;1", "1?toolate,1") in new stack
[2019-05-15 08:13:28] VERBOSE[5550][C-0000908d] pbx_builtins.c: Goto (macro-confirm,toolate,1)
[2019-05-15 08:13:28] VERBOSE[5550][C-0000908d] pbx.c: Executing [toolate@macro-confirm:1] Set("Local/12128885500@from-internal-000064ab;1", "MSG2="incoming-call-no-longer-avail"") in new stack
[2019-05-15 08:13:28] VERBOSE[5550][C-0000908d] pbx.c: Executing [toolate@macro-confirm:2] Playback("Local/12128885500@from-internal-000064ab;1", ""incoming-call-no-longer-avail"") in new stack
[2019-05-15 08:13:28] VERBOSE[5550][C-0000908d] file.c: <Local/12128885500@from-internal-000064ab;1> Playing 'incoming-call-no-longer-avail.slin' (language 'en'

macro-confirm in extensions_additional.conf: https://pastebin.freepbx.org/view/f354765e

So I compared both macros and the only difference is on line 12.
Old:

exten => 1,1,GotoIf($["${DB_EXISTS(RG/${ARG3}/${UNIQCHAN})}"="0" & "${SHARED(ANSWER_STATUS,${FORCE_CONFIRM})}"=""]?toolate,1)

New:

exten => 1,1,GotoIf($[$[("${DB_EXISTS(RG/${ARG3}/${UNIQCHAN})}"="0") | ("${SHARED(BLKVM,${UNIQCHAN})}"="")] & "${SHARED(ANSWER_STATUS,${FORCE_CONFIRM})}"=""]?toolate,1)

–
So in a attempt to fix things on my own, I went ahead a replaced the “|” with a “&”, like this:

exten => 1,1,GotoIf($[$[("${DB_EXISTS(RG/${ARG3}/${UNIQCHAN})}"="0") & ("${SHARED(BLKVM,${UNIQCHAN})}"="")] & "${SHARED(ANSWER_STATUS,${FORCE_CONFIRM})}"=""]?toolate,1)

Ran core reload, tested a call, and now I am able to take calls again.
Is this a bug?

Thanks

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@bajramia wrote:

Hi All,
I have Vega Gateway 3050 i have mapped all the extension form PBXact Vega Management Module
I’m having issue with some extension are getting unreachable then they get back registered by them self.

Added contact ‘sip:6222@192.168.126.29:5160’ to AOR ‘6222’ with expiration of 900 seconds
– Removed contact ‘sip:6222@192.168.25.30:5060’ from AOR ‘6222’ due to remove existing
== Contact 6222/sip:6222@192.168.25.30:5060 has been deleted
== Endpoint 6222 is now Unreachable

after few min this extension will be register

== Endpoint 6222 is now Reachable
– Contact 6222/sip:6222@192.168.25.30:5060 is now Reachable. RTT: 6.137 msec
and the msec is very high but when i ping the gateway from the server

PING 192.168.25.30 (192.168.25.30) 56(84) bytes of data.
64 bytes from 192.168.25.30: icmp_seq=1 ttl=64 time=0.435 ms
64 bytes from 192.168.25.30: icmp_seq=2 ttl=64 time=0.396 ms
64 bytes from 192.168.25.30: icmp_seq=3 ttl=64 time=0.343 ms
64 bytes from 192.168.25.30: icmp_seq=4 ttl=64 time=0.342 ms
64 bytes from 192.168.25.30: icmp_seq=5 ttl=64 time=0.371 ms
64 bytes from 192.168.25.30: icmp_seq=6 ttl=64 time=0.398 ms
64 bytes from 192.168.25.30: icmp_seq=7 ttl=64 time=0.374 ms

thank you,

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@mvogel4949 wrote:

I have an extension that’s won’t come off Away/DND. Setting it to available in the UCP within 5s it will revert back to Away. Same with the extension when viewed through the Zulu desktop or mobile app. The extension within freepbx has the dnd box checked. I’ve tried to restart the restapps server but still it goes back to away after I change it to available. I also tried to dial *76 from the softphone to toggle it but no luck

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@volarmc wrote:

Hello everyone, this is my first post and I’m fairly new to Freepbx so please forgive me if I don’t initially include enough info. I’ll be happy to provide any logs needed to help diagnose the issue.

My issue is that my Cyberdata intercom works perfectly if I have it dial a direct extension but when I have it dials a ring group instead the intercom speaker will not work. The extension that picks up can hear the intercom but can’t speak back. I tested with a page group as well and two way audio worked correctly but in my use case a page group won’t work.

I should note that the intercom and the extensions are connected to the freepbx server over a site-to-site vpn. Calls and all other options are working fine. Thanks in advance.

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@PitzKey wrote:

Hello,

After the tool finished importing all extensions, this is what happens next

Adding user 3291 (Fax ATA Upstairs) ... User created (Note: No email address!) with password store6Toast6remit
Adding user 3300 (Lunch Room) ... User created (Note: No email address!) with password heard9grip3Sprint
Exception: SQLSTATE[08004] [1040] Too many connections::SQLSTATE[08004] [1040] Too many connections in file /var/www/html/admin/libraries/utility.functions.php on line 204
Stack trace:
  1. Exception->() /var/www/html/admin/libraries/utility.functions.php:204
  2. die_freepbx() /var/www/html/admin/libraries/BMO/Database.class.php:142
  3. PDOException->() /var/www/html/admin/libraries/BMO/Database.class.php:137
  4. PDO->__construct() /var/www/html/admin/libraries/BMO/Database.class.php:137
  5. FreePBX\Database->__construct() /var/www/html/admin/libraries/BMO/Self_Helper.class.php:124
  6. FreePBX\Self_Helper->autoLoad() /var/www/html/admin/libraries/BMO/Self_Helper.class.php:53
  7. FreePBX\Self_Helper->__call() /var/www/html/admin/modules/sysadmin/Sysadmin.class.php:1231
  8. FreePBX\modules\Sysadmin->Database() /var/www/html/admin/modules/sysadmin/Sysadmin.class.php:1231
  9. FreePBX\modules\Sysadmin->getDDNSSettings() /var/www/html/admin/modules/sysadmin/VPN.class.php:104
 10. FreePBX\modules\Sysadmin\VPN->putClient() /var/www/html/admin/modules/sysadmin/VPN.class.php:508
 11. FreePBX\modules\Sysadmin\VPN->linkClients() /var/www/html/admin/modules/sysadmin/VPN.class.php:436
 12. FreePBX\modules\Sysadmin\VPN->usermanUpdateGroup() /var/www/html/admin/modules/sysadmin/Sysadmin.class.php:1793
 13. FreePBX\modules\Sysadmin->usermanUpdateGroup() /var/www/html/admin/libraries/BMO/Hooks.class.php:189
 14. call_user_func_array() /var/www/html/admin/libraries/BMO/Hooks.class.php:189
 15. FreePBX\Hooks->processHooksByClassMethod() /var/www/html/admin/modules/userman/functions.inc/auth/Auth.php:49
 16. FreePBX\modules\Userman\Auth\Auth->updateGroupHook() /var/www/html/admin/modules/userman/functions.inc/auth/modules/Freepbx.php:218
 17. FreePBX\modules\Userman\Auth\Freepbx->updateGroup() /var/www/html/admin/modules/userman/Userman.class.php:1923
 18. FreePBX\modules\Userman->updateGroup() /tmp/tmp.YyLjMkLf1y/freepbx.import.php:172
 19. include() /tmp/tmp.YyLjMkLf1y/newmachine.php:231
 20. include() /tmp/tmp.YyLjMkLf1y/convert.php:36
Cleaning up...Done!
[root@freepbx ~]#

I am not sure what is missing, what else it’s supposed to be importing after importing the extensions.

What happens if I run the tool again? and what can I do to fix this error?

Thanks

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@benrooke wrote:

I want to set up a ring group that sends all calls to external numbers on a ring all strategy, however what i am finding is that when the call hits the ring group if there is a number that has VM it answers the call. I cannot set it for users to dial 1 as they are using a sfb client

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@dan_ce wrote:

Does anyone have any advice on how to send the incoming call number to MQTT from within the GUI? Thanks!

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@astiles94 wrote:

Okay, so I’ve posted here a few time (I’m a slow responder most generally) and usually gotten my answer or been pointed down the correct path. I’m super new to the PBX thing so don’t be too hard on me.

I have my phone system up and running (40+ phones) everything seemed fine however when someone calls in and they need to press 4 to be directed somewhere it says Invalid response same with outgoing calls as well.

I can provide most info needed if I’m laking any. I have tried searching the interwebs but my searches have come up with problems not similar to mine maybe in searching for the wrong thing I’m not to familiar with all the terms just yet.

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@4ra wrote:

Hello I am running into an issue when I try to look at Intrusion Detection

3. Exception
   /­var/­www/­html/­admin/­modules/­sysadmin/­Sysadmin.class.php1737

4. FreePBX\modules\Sysadmin getFail2BanList
   /­var/­www/­html/­admin/­modules/­sysadmin/­functions.inc/­intrusion.php58

5. sysadmin_get_banned
   /­var/­www/­html/­admin/­modules/­sysadmin/­page.sysadmin.php399

6. include
   /­var/­www/­html/­admin/­config.php559

I tried the suggested fix to run yum upgrade -y in the similar post I found listed below.

I’ve tried to re-install the module and also have run fwconsole chown and fwconsole restart and no results.

Any advice would be appreciated as I’m fairly new to working with FreePBX.

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