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How to test Automatic Page Group without schedule?

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@BGM wrote:

My polycom 601’s won’t auto-answer pages, so I’m playing with their config settings, but I have to constantly change the time on the scheduler and wait for it.

Is there a way to test the page-group without having to set the scheduler? If I just dial the page extension, it connects the calling extension and I don’t want to do that. I want the system to send the announcement to the destination.

I am trying to make bells play over the phone. Once I get it working then I can set the schedule.

It used to be working before (I have a thread here), but now it isn’t and I can’t figure out why. The phone always rings instead of answering.

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Port Conflict error

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@krdodia wrote:

On my dashboard of FreePBX there is a following error : An unknown port conflict has been detected in PJSIP. Please check and validate your PJSIP Ports to ensure they’re not overlapping.

what does that error mean? and how could I fix it?

My SIP ports are as follows:
pjsip
Port to Listen On: 5060

Chan Sip
Bind Port: 5160
TLS Bind Port: 5161
how they are overlapping?

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Problem with Distro Conversion tool

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@SBaalman wrote:

I’ve been playing with setting up FreePBX on a few VM’s in preparation to port from our old one to new one. I have a VM with clean install of FreePBX up and running. I started the Distro Conversion tool on it, located here on this wiki https://wiki.freepbx.org/display/PPS/Elastix+and+PBXinaFlash+to+FreePBX+Distro+Conversion+Tool

I received my code, thought I’d be smart and shut down the other test VM that I had made with FreePBX running on it. Started the tool on the donor machine, went back the new machine and realized I had accidentally shut down the wrong one. I started it back up, ran the tool command

curl -s https://convert.freepbx.org | bash

which did something for a bit and then came back to a normal prompt instead of outputting the normal stuff that should have looked like this;

[root@freepbx ~] # curl -s https://convert.freepbx.org | bash

Checking that 'curl' exists ... OK!

Validating sha256 integrity ... OK!

Trying to download converter to /tmp/tmp .AKctIWjXSd ... Complete

Validating download ... OK!

Testing connectivity to Conversion server...Success!

FreePBX Conversion Wizard

-------------------------

The FreePBX Conversion Wizard needs to be run on two machines, the NEW

machine, which must be an ACTIVATED FreePBX Distro machine, and then it must

be run on the DONOR machine.

The DONOR machine is the machine that is currently processing calls, and is

the machine that will be migrated to the NEW machine. No changes will be made

to the DONOR machine, and this script will not stop or restart any services

that may cause an outage.

If this is the NEW machine, just push 'Enter' to prepare this machine

Enter ID (blank if this is NEW):

Testing FreePBX functionality ... Success!

Getting Deployment ... 12345678

Getting all Module versions ( for conversion) ... OK

Reserving a conversion slot ... Reserved!

The Conversion process is now ready. Please run the script on the

DONOR node now, and when asked for a slot identifier, please enter

the following ID:

12345789-abcd-4ef0-1234-56789abcdef0

[-] Waiting for Donor...

I need that ID to put in the donor machine, but have no idea what my new machine is doing much less the code it had before.

Any help would be appreciated if anyone knows more about this tool and if it is still running or not

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Where are conference bridge configs stored?

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@bksales wrote:

Hoping this is an easy question to answer. I went digging through the asterisk database tables and cant find it. Also don’t see it in /etc/asterisk (confbridge.conf and associated files) and digging through the conferences.class.php file for hints but I guess I’m not much of a programme. functions.php mentions some meetme files in /etc/asterisk that don’t have anything in them either.

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Call hung up, cause:busy

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@krdodia wrote:

I am trying to web call using freepbx and asterisk, but outgoing call always hung up by saying it’s busy. There are no errors in asterisk log. It happens after updating the SSL certificate, before that the call was working. Please let me know what can be the possible cause? and also, if you need any additional information. I have attached the screenshot of call log :

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How do I change my password for these forums?

Permission Denied

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@v70ff wrote:

Any “.conf” file that i try to access I get the permissions denied message.

I am trying to access it as root user “[root@freepbx ~]#”

Does anyone know why this is happening?

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Follow Me Ring Time doesn't work for cell phones

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@cfapress wrote:

I’m attempting to understand the Follow Me - Ring Time

Depending upon the Ring Strategy I understand the numbers in the Follow Me list will be called in various orders. They should be called for the time defined in Ring Time.

However … when the Follow Me List includes a cell phone that phone will ring and ring and ring - completely ignoring the Ring Time setting.

It’s as if Asterisk ignores the Ring Time setting for any external number - why?

My goal is to have an incoming call ring an internal extension for 8-seconds.
If nobody picks up the ring the cell phone for 8-seconds.
If nobody answers then fall back to the Asterisk voice mailbox.

I haven’t been able to find the correct mix of Initial Ring Time, Ring Strategy, Ring Time, and List values.

Help!

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FreePBX 14 GUI very slow

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@SBaalman wrote:

The GUI is super slooow logging in, pulling up the dashboard and going to other functions within the GUI.

I have added the IP and hostname to the DNS server which helped a bit for some things, the first entry in DNS setting on the unit is 127.0.0.1

Here’s what I pulled from running fwconsole debug

[root@FreePBX ~]# fwconsole debug
±----------------------+
| FreePBX Notifications |
±----------------------+
OUT > ==> /var/log/asterisk/freepbx_dbug <==

==> /var/log/httpd/error_log <==
140466771507088:error:0200100D:system library:fopen:Permission denied:bss_file.c:402:fopen(’/etc/pki/tls/certs/localhost.crt’,‘r’)
140466771507088:error:20074002:BIO routines:FILE_CTRL:system lib:bss_file.c:404:
unable to load certificate
[Thu Aug 29 18:38:55.443293 2019] [mpm_prefork:notice] [pid 1230] AH00170: caught SIGWINCH, shutting down gracefully
[Thu Aug 29 18:38:56.443693 2019] [core:notice] [pid 1230] AH00052: child pid 12117 exit signal Segmentation fault (11)
[Thu Aug 29 19:42:34.704777 2019] [suexec:notice] [pid 1324] AH01232: suEXEC mechanism enabled (wrapper: /usr/sbin/suexec)
[Thu Aug 29 19:42:34.772183 2019] [auth_digest:notice] [pid 1324] AH01757: generating secret for digest authentication …
[Thu Aug 29 19:42:34.772849 2019] [lbmethod_heartbeat:notice] [pid 1324] AH02282: No slotmem from mod_heartmonitor
[Thu Aug 29 19:42:35.735719 2019] [mpm_prefork:notice] [pid 1324] AH00163: Apache/2.4.6 (Sangoma) OpenSSL/1.0.2k-fips PHP/5.6.40 configured – resuming normal operations
[Thu Aug 29 19:42:35.735750 2019] [core:notice] [pid 1324] AH00094: Command line: ‘/usr/sbin/httpd -D FOREGROUND’

==> /var/log/asterisk/freepbx_security.log <==
[2019-08-29 15:08:32] Authentication failure for vsradmin from 192.168.2.11
[2019-08-29 15:08:32] Possible proxy detected, forwarded headers forvsradmin set to

==> /var/log/asterisk/freepbx.log <==
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module queuestats, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module qxact_reports, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module recording_report, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module restapps, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module sangomacrm, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module sipstation, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module soundlang, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module userman, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module vqplus, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module zulu, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
OUT > [2019-Aug-29 21:16:52] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 67
OUT > [2019-Aug-29 21:16:52] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 76
OUT > [2019-Aug-29 21:16:53] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 67
[2019-Aug-29 21:16:53] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 76
OUT > [2019-Aug-29 21:18:13] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 67
OUT > [2019-Aug-29 21:18:13] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 76

Here’s what is in functions.inc.php between lines 67 and 76 since those two are referenced

 67         $um = module_getinfo('userman', MODULE_STATUS_ENABLED);
 68         if(file_exists($amp_conf['AMPWEBROOT'].'/admin/modules/userman/functions.inc.php') && (isset($um['userman']['status']) && $um['userman']['status'] === MODULE_STATUS_ENABLED))         {
 69                 include_once($amp_conf['AMPWEBROOT'].'/admin/modules/userman/functions.inc.php');
 70         }
 71 }
 72
 73 //Ensure that the manager module has loaded. If not, load it.
 74 if(!function_exists('manager_add')){
 75         global $amp_conf;
 76         $um = module_getinfo('manager', MODULE_STATUS_ENABLED);

I noticed in debug it throws those bits about lines 67 and 76 and suddenly the GUI responds

Any ideas?

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SQLSTATE[HY000]: General error: 130 Incorrect file format 'featurecodes'

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@bksales wrote:

I don’t know whats going on here but I’ve had a dozen servers take a dump in recent weeks. There are different symptoms but the results are the same, namely that asterisk continues to run but the GUI is unusable. Once the server is rebooted for any reason it wont start back up again. Restoring snapshots doesn’t work either. Most have some error or other about the database. If I see a message about tampered files in the dashboard that seems to be the precursor to this.

In this particular case I can get to most of the modules in the GUI but if I try to edit an extension I get the error in the thread title. I’ve seen half a dozen different errors starting with SQLSTATE.

How can I go about diagnosing this? And are these machines salvageable at all?

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DID and CID?

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@jpark1205 wrote:

Hello

I am confused on the difference between DID and CID. I know that DID stands for Direct in Dial and CID stands for Caller Id. I did google it on line but the examples that they put is kind of confusing… Can anyone please provide one good example??

Thank You

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No assisted transfers are possible, Asterisk 13.22.0, FreePBX Utility 14.0.5.5

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@ELindemann wrote:

Hello @all,

i did not found a solution to this, my problem, therefore i am asking here.

I cant do none assisted calls, the system will loose functionality to calls from outside.

*2 < ext > # does not work. (sorry brackets are gone, try it with space, textform: star two bracket ext bracket hash, with no space)

# < ext > # does work. (textform: hash hash bracket ext bracket hash, with no space)

What is going wrong?

Example, what goes wrong. :frowning:
Outside call comes to a process/queue and so on, all that stuff works as designed. Also pressing 1/2/5 in IVR works fine. If nothing happens -nobody will answer the call, after X min. the system plays an announcement, and hangs up. That all is very fine.
The call flow ist fine.

If the queue members ring (211 or 212) -somebody get che call, we can do ## < ext > # many times. As playing football. The call will not get lost.

But no *2 < ext > # will work.

  • call comes to 211/212, somebody answers
  • speaking … then
  • *2 < ext > # will ring the other ext. (at *2 MOH is online to callee)
    now, the two exts can speak to each other.
    Until now, it works as wanted.

But …
After the first ext hangs up, the callee will not be connected to the new ext, instead, the callee will hear the music/MOH until he dies. :wink:

But after this procedure, a new call will not ring on the queue members, on ext 211/212. They are dead for the outside call.

This is possible:
216 can call 211/212
211 -> 212 OK
212 -> 212 OK

But, if somebody calls from outside, none of the queue-members 211/212 will ring. There is NO timeout, that it will works after this timeout. Nope. Only a

/var/lib/asterisk/bin/amportal restart

will reset the system.

asterisk -rx “core show channels” shows:

Channel Location State Application(Data)
PJSIP/211-00000011 s@macro-dial-one:1 Up AppDial((Outgoing Line))
PJSIP/212-00000013 s@macro-dial-one:1 Up AppDial((Outgoing Line))
2 active channels
0 active calls
43 calls processed
Asterisk ending (0).

Assisted call transfer kills (somehow) the whole functionality (for the outside callee)

Other calls to ext 214/215 etc will ring. Also

Call to 214 (direct form outside) rings, ##<211># is OK.
Call to 214 (direct form outside) rings, *2# is also OK, as pressed *2 MOH will plays, the other rings, get the call, speak to 214, as above after 214 hangs up, MOH will play to callee, the call is not asssited transferd to other .

At this status, no intercom possible. Only
amportal restart
will clean/reset the system, brings (for a short period) the whole functionality, untel the first assisted transfers (which also will not works). :frowning: :frowning: :frowning:

asterisk -rx “pjsip show endpoints” | grep -e ‘in|Unava’

Endpoint: 11/11 Unavailable 0 of inf - OK
Endpoint: 12/12 Unavailable 0 of inf - OK
Endpoint: 13/13 Unavailable 0 of inf - OK

Endpoint: 211/211 In use 2 of inf - why? QMember
Endpoint: 212/212 In use 1 of inf - why? QMember

Endpoint: 213/213 In use 1 of inf - why?
Endpoint: 214/214 On Hold 3 of inf - ?
Endpoint: 215/215 Not in use 0 of inf OK
Endpoint: 216/216 In use 2 of inf - why?
Endpoint: 217/217 Unavailable 0 of inf - OK/NC
Endpoint: 218/218 Not in use 0 of inf OK
Endpoint: 219/219 Not in use 0 of inf OK
Endpoint: 220/220 Not in use 0 of inf OK
Endpoint: 221/221 Not in use 0 of inf OK
Endpoint: 99/99 Not in use 0 of inf OK

After a time period:

asterisk -rx “core show channels”
Channel Location State Application(Data)
PJSIP/216-0000004d (None) Up AppDial((Outgoing Line))
PJSIP/216-00000053 (None) Up AppDial((Outgoing Line))
PJSIP/211-00000011 s@macro-dial-one:1 Up AppDial((Outgoing Line))
PJSIP/214-0000004c s@macro-dial-one:54 Up Dial(PJSIP/216/sip:216@IP-Addr
PJSIP/214-00000052 s@macro-dial-one:54 Up Dial(PJSIP/216/sip:216@IP-Addr
PJSIP/214-00000039 (None) Up AppDial((Outgoing Line))
Local/211@from-inter 211@from-internal-xf Up (None)
Local/211@from-inter s@macro-dial-one:54 Up Dial(PJSIP/211/sip:211@IP-Addr
PJSIP/211-0000003a (None) Up AppDial((Outgoing Line))
Local/213@from-inter s@macro-dial-one:54 Up Dial(PJSIP/213/sip:213@IP-Addr
Local/213@from-inter 213@from-internal-xf Up (None)
PJSIP/213-00000054 (None) Up AppDial((Outgoing Line))
PJSIP/212-00000013 s@macro-dial-one:1 Up AppDial((Outgoing Line))
13 active channels
4 active calls
132 calls processed

The HW is, YL 48S (211/217) /46S (212) /60p (213/4/5/6/8) with five 56H.
Tested also temp. with Grandstream 2170 as 211, same.

Any, any help appreciated. Thanks in advance.

ELindemann

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Zulu "There is an error connecting to the server" desktop client

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@dan_ce wrote:

One of my desktop PC’s Zulu desktop client doesn’t login to the Zulu server, when I open the desktop client window it says “there is an error” but as soon as I click login, it works.

I wonder
a) why it’s having this error (when there doesn’t seem to even be a login error at all)
and
b) if there is a way for it to auto retry say every five mins, hour, 2 hours etc?

Alternatively I could write a batch script to nuke the desktop client and reload it say 2 minutes after boot up? But seems odd to have to do that.

Thanks

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How to use existing 70V speakers for IP paging with separate ze

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@ghurty wrote:

Can anyone recommend a product that would let me use existing 70V speakers as an IP paging system? We would want each speaker to be individually addressed.

Thank you

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Outgoing call ends with cause busy

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@krdodia wrote:

I am sorry that I am posting this question again. But I am stuck and I don’t know why it happens. I am using freepbx, and my outgoing call always ends up with the cause busy. No errors in asterisk log. I have attached the screenshot of my asterisk log.

I have also attached my pjsip configuration.
pjsip.conf :w3
pjsip.endpoint.conf :w4
pjsip.transport.conf : w5

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Inbound Routes takes 2 minutes to load

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@jmarsh wrote:

We are using FreePBX 14.0.13.4 (migrated from 13), and it takes a full two minutes to load the list of inbound routes. Any idea what would cause this? We currently have 30 inbound routes. Other pages such as extensions and user listings load in seconds.

I looked at performance logging, and found this chunk in the logs while loading the page:

OUT > 
==> /var/log/asterisk/freepbx.log <==
[2019-Aug-30 11:54:11] [WARNING] (core/functions.inc.php:6175) - Depreciated Function core_users_list detected in /var/www/html/admin/modules/core/functions.inc.php on line 860
OUT > [2019-Aug-30 11:54:11] [WARNING] (core/functions.inc.php:6175) - Depreciated Function core_did_list detected in /var/www/html/admin/modules/core/functions.inc.php on line 907
OUT > 
==> /var/log/asterisk/freepbx_dbug <==
2019-Aug-30 11:54:11	/var/www/html/admin/libraries/BMO/Performance.class.php:118


Array
(
    [type] => PERF
    [str] => processHooks-Contactmanager::usermanAddContactInfo_start
    [now] => 1567191251.625433
    [timediff] => 128.163869
    [mem] => 43.99688 MB
    [memdiff] => 35.612595 MB
    [file] => /var/www/html/admin/modules/userman/Userman.class.php:1210
)

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System Time and Time Conditions bug

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@dan_ce wrote:

Pretty sure I’ve just found a bug. I don’t know for sure, but I’m also tentatively linking it to the fact that when I look at my log files, the time is always an hour behind the actual time (due to DST?).
HOWEVER, The Freepbx “server time” sometimes shown in the web GUI was showing the correct time, in this case about 20:09

Anyway I have a time condition which is set to only be active between 1800 and 1930. A caller just alerted me to the fact they couldn’t get through at 2005 (due to the time condition). I went in and checked and was initially confused as to why things weren’t working as they should.

I then noticed that within “time conditions” there is an additional timezone dropdown box you can optionally fill in. I changed this from “system time” (should have worked fine) to Europe/London and lo and behold, the caller was able to get through!

Weird.

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How to disable "Potential Security Breach" page?

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@BGM wrote:

I like to keep a bunch of FreePBX pages open in my browser; sometimes when I return to one of those pages, the page is replaced with:

Potential Security Breach
You are attempting to modify settings from a URL that does not appear to have come from a FreePBX page link or button. This can occur if you manually typed in the URL below. This action has been blocked because the HTTP_REFERER does not match your current SERVER. If you require this access, you can set Check Server Referrer=false in Advanced Settings to disable this security check

The suspect URL is listed below. If this action is intended, you can click this link and your action will be processed. Do not proceed with this if you did not intended to execute this command as it may result in changes to your configuration.

Is there a way to disable this? Refreshing the page doesn’t work and I have to navigate back to where I was.

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Question on CDR related to ring groups

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@cfapress wrote:

We have a single ring group, 0, which then rings two phones 1200 and 1201.

I need to report on how many calls each extension took. Even if that extension then forwarded the call to a different final destination.

I’ve been unable to figure out how to perform this with CDR. Even dumping the data into a CSV file wasn’t helpful.

For example, an incoming call is answered by x1200 and then forwarded to x1230. This needs to count as a call to x1200 in my desired report.

Thoughts?

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HTTPS Provisioning Not Working Through Firewall

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@drummerjoe wrote:

When the system firewall is turned on, phones external to our network are unable to autoprovision via HTTPS. As soon as we disable the system firewall, they are able to autoprovision. We’ve tried setting the HTTPS autoprovisioning service in the firewall to Local, Other, and Internet, and it doesn’t work on any of those settings.

What is the proper setting to allow HTTPS provisioning through the firewall?

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