@ipvinner wrote:

Hello, I’d like to start Freepbx 15 on nginx and php-fpm
php 7.1 and latest version nginx installed
nginx.conf
server {
…
root /var/www/html/;
index index.php index.html index.htm;

    location ~* \.(jpg|jpeg|gif|png|ico|css|bmp|swf|js|html|txt)$ {
        root /var/www/html;
    }

	location / {
		try_files $uri $uri/ /index.php?q=$uri&$args;
	}

        location ~ \.php$ {
fastcgi_pass unix:/var/run/php-fpm/php-fpm.sock;
fastcgi_index index.php;

fastcgi_param DOCUMENT_ROOT /var/www/html/;
fastcgi_param SCRIPT_FILENAME /var/www/html$fastcgi_script_name;
fastcgi_param PATH_TRANSLATED /var/www/html/$fastcgi_script_name;

include fastcgi_params;
fastcgi_param QUERY_STRING $query_string;
fastcgi_param REQUEST_METHOD $request_method;
fastcgi_param CONTENT_TYPE $content_type;
fastcgi_param CONTENT_LENGTH $content_length;
fastcgi_intercept_errors on;
fastcgi_ignore_client_abort off;
fastcgi_connect_timeout 60;
fastcgi_send_timeout 180;
fastcgi_read_timeout 180;
fastcgi_buffer_size 128k;
fastcgi_buffers 4 256k;
fastcgi_busy_buffers_size 256k;
fastcgi_temp_file_write_size 256k;
}

and have 2 issues:
$.removeCookie is not a function
and
https://url/admin/ajax.php?module=core&command=getExtensionGrid&type=custom&search=&order=asc&_=1581405862113 401
{“error”:“Not Authenticated”}

althought redirected to dashboard page after login. Possible somebody has already installed freepbx with nginx, please could you you help how to configure that?

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@MrMaxFR wrote:

Hello,

I wanted to know if the Mitel 6867i is fully compatible with FreePBX, since I wanted to buy it I would like to make sure.

Sincerely
Maxime / MrMaxFR

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@Philipp971 wrote:

Hi there!

So - i’am looking for a good and simple Wallboard to show some details and infos from the PBX within the browser. Something like: https://www.fop2.com/index.php

So - there is no official installation guide for the FOP2 (or i dont found it yet) and i want to know what Info/Wallboards you are using? Did you got any experience with FOP2? Any recommended alternative?

Thanks for any help!
Philipp

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@AmidouFlorian92 wrote:

Hi
I’m trying to send SMSs using my freepbx/asterisk server.
On asterisk website I saw this solution : https://wiki.asterisk.org/wiki/display/AST/SMS
So I tried It but I got this error :

[2020-02-11 13:11:48] WARNING[2712]: pbx_spool.c:292 parse_line: Unknown keyword ‘action’ at line 1 of /var/spool/asterisk/outgoing/hello-world.call
[2020-02-11 13:11:48] WARNING[2712]: pbx_spool.c:292 parse_line: Unknown keyword ‘exten’ at line 3 of /var/spool/asterisk/outgoing/hello-world.call
[2020-02-11 13:11:48] WARNING[2712]: pbx_spool.c:329 apply_outgoing: At least on e of app or extension must be specified, along with tech and dest in file /var/s pool/asterisk/outgoing/hello-world.call
[2020-02-11 13:11:48] WARNING[2712]: pbx_spool.c:514 scan_service: Invalid file contents in /var/spool/asterisk/outgoing/hello-world.call, deleting

This is my original helloword.call file :
action: originate
callerid: Salut tout le monde
exten: to
channel: SIP/1900/53584523
context: smsdial
priority: 1

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@MrXirtam wrote:

FreePBX 15.0.16.42
Asterisk Version: 16.6.2

I have a VPS hosted FreePBX server, and I am testing out a different SIP trunk provider. Doing so has forced me to change the SIP port to the standard 5060. I started seeing call attempts from non-existent extensions to international numbers. The calls weren’t able to complete, but they are filling my logs and CDR’s with junk. I have disabled my responsive firewall for my PJSIP traffic. I have added my local static IP to the networks tab as trusted for my phones. They register and work just fine. However, watching my SIP logs, I constantly keep seeing invites trying to place calls from extensions I do not have on my PBX. I added the source IP’s to the blacklist in the firewall settings and I have restarted asterisk, and it seems like it hasn’t made a difference. Those same IP’s I have blocked still persist with attempts. Under services, I have changed my SIP protocol to Local only as well. Under SIP settings, I have disabled both Allow Anonymous Inbound SIP Calls and Allow SIP Guests. I’m not sure if I am missing something in the firewall to prevent these attempts.I have my intrusion detection [fail2ban] running and I increased those restrictions too.

Can someone help me identify what I am missing to help tighten security?

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@fbedia wrote:

I can`t work with FREEPBX GUI…

In general everything goes slow, but certain actions are impossible to perform… for example, if I press the button to add recording, the page keeps loading continuously…

In fwconsole debug, I see some warnings when I the button are clicked

2020-02-11 17_34_42-AsteriskPBX - VMware Workstation.png815×146 1.38 KB

It is impossible to work or make modifications …

Only I have this problem?

Does anyone know how to fix it?

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@Adalis wrote:

Calendar module 14.0.3.9. Customer uses receptionist in two different office locations. So at HQ setup Calendar to have receptionist answer on Mon-Wed-Fri and IVR answer on Tues-Thurs while she is offsite. Works great! The challenge comes when management decides to give some time off and close the office early.

I have the calendars setup as recurring, but if I need to alter a day everything beyond that date disappears and I have to recreate the calendar item on the next day. The next time they change the schedule ,which is completely random, then I do this all over again. Setting up remote calendars for all our clients would be a pain.

Can I create a separate calendar item for the date in question and will it override the recurring item on that same day?

I’m spoiled as in Outlook it will give you the option of changing one or all items in the series.

Appreciate anyone’s experience that will make these calendar changes easier.

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@RobS wrote:

I have a bunch of remaining 90xxx extensions even after removing Zulu, WebRTC, etc (even reinstalling and making sure it was disabled for all users, then removing)… They will not go away.

How do I finally get rid of them and keep them from polluting my logs?

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@edwinr wrote:

Got problem with my audio won’t play specially the internal message like not inservice. I can see from the logs that it attempt to play but no audio was played then busy tone after. I verify the audio file and it exist and work just fine. File permission is fine also. Hope anyone can help me. ty!
FreePBX 12
asterisk: 11
It work before but suddenly this week this thing happen.

logs:

– Executing [s-INVALIDNMBR@macro-dialout-trunk:1] NoOp(“SIP/101-00004696”, “Dial failed due to trunk reporting Address Incomplete - giving up”) in new stack
– Executing [s-INVALIDNMBR@macro-dialout-trunk:2] Progress(“SIP/101-00004696”, “”) in new stack
– Executing [s-INVALIDNMBR@macro-dialout-trunk:3] Playback(“SIP/101-00004696”, “ss-noservice,noanswer”) in new stack
– <SIP/101-00004696> Playing ‘ss-noservice.gsm’ (language ‘en’)
– Executing [s-INVALIDNMBR@macro-dialout-trunk:4] Busy(“SIP/101-00004696”, “20”) in new stack
== Spawn extension (macro-dialout-trunk, s-INVALIDNMBR, 4) exited non-zero on ‘SIP/101-00004696’ in macro ‘dialout-trunk’

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@Philipp971 wrote:

Hi there!

I’m a bit confused. For first, i used regular ring groups which are processed by an IVR and it’s digits.
Its working fine - but its actually not a real queue for sure. The calls are hanging up when the number of seconds is reached but the caller needs to be in a queue for a long ttime (when a lot of customers are calling e.g).

So - but here’s the “problem”. The queues are (and i absolutely dont know why) using differend ring strategys as the ring groups. I need the memoryhunt-strategy to use it in the queue.

There’s an linaer-strategy, but within this strategy the call ends in a extension and goes right to the next one. I dont know what the “rrmemory” does because setting this one up it’s actually the same like linaer, i dont figure out the difference for now.

Is there any way to configure the memory-hunt feature within a queue?

Thanks a lot!
Philipp

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@plasky wrote:

Is there any way to add prefix to a forwarded calls?

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@AmidouFlorian92 wrote:

Hi
I’m trying to send SMS trough mu freepbx server using this tutorial http://forum.thewebmachine.net/enable-sip-sms-support-this-is-not-cellular-sms-t42.html
But It seems not to be working and I don’t receive any information on asterisk console so I don’t know from what come the problem.I proceed like this :
Image 1 ==> asterisk advanced settings
In /etc/asterisk/extensions_custom.conf :
[astsms]
exten => _.,1,NoOp(SMS receiving dialplan invoked)
exten => _.,n,NoOp(To ${MESSAGE(to)})
exten => _.,n,NoOp(From ${MESSAGE(from)})
exten => _.,n,NoOp(Body ${MESSAGE(body)})
exten => _.,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)})
exten => _.,n,MessageSend(${ACTUALTO},${MESSAGE(from)})
exten => _.,n,NoOp(Send status is ${MESSAGE_SEND_STATUS})
exten => _.,n,GotoIf($["${MESSAGE_SEND_STATUS}" != “SUCCESS”]?sendfailedmsg)
exten => _.,n,Hangup()
;
; Handle failed messaging
exten => _.,n(sendfailedmsg),Set(MESSAGE(body)="[${STRFTIME(${EPOCH},%d%m%Y-%H:%M:%S)}] Your message to ${EXTEN} has failed. Retry later.")
exten => _.,n,Set(ME_1=${CUT(MESSAGE(from),<,2)})
exten => _.,n,Set(ACTUALFROM=${CUT(ME_1,@,1)})
exten => _.,n,MessageSend(${ACTUALFROM},ServiceCenter)
exten => _.,n,Hangup()
exten => _.,n,Hangup()
I tried to send SMS trough the sip line using linphone softphone but It doesn’t work
Need help to solve the problem
Thanks

1.JPG723×161 9.31 KB

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@ghurty wrote:

I am running an old version of asterisk on rentpbx. I dont remember the exact version but was deployed in 2016. The server stopped working, and I rebooted it, now it asterisk doesnt want to start. When I run amportal start I get the following error. I googled but I done see any restults with that line 46 error.

Please wait…
/usr/local/sbin/amportal: line 46: [FATAL]: command not found

/var/lib/asterisk/bin/freepbx_engine: line 98: [FATAL]: command not found
**** WARNING: ERROR IN CONFIGURATION ****
astrundir in ‘/etc/asterisk’ is set to but the directory
does not exists. Attempting to create it with: 'mkdir -p ’

mkdir: missing operand
Try `mkdir --help’ for more information.
**** ERROR: COULD NOT CREATE ****
Attempt to execute 'mkdir -p ’ failed with an exit code of 1
You must create this directory and the try again.

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@Hawkeye wrote:

Hi, getting complaints about dropped calls. When given the time and extension looked in /var/log/asterisk/full log and see the following entry which was one of the reported dropped calls:

[2020-02-12 11:28:36] NOTICE[5602] chan_sip.c: Disconnecting call ‘SIP/227-00004dbe’ for lack of RTP activity in 31 seconds

The person complaining about the dropped call had said there was conversation going on and then it dropped.

Is there something we can do to fix this ?
Thanks,.

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@Answerphone wrote:

We had someone call in over the weekend and somehow gain access to the phone system, was able to forward all our extensions to 301-4XX-4XXX using what appears to be zulu and feature codes.

I think I figured it out and semi-patched the hole. In my findings, a hacker could simply press * in an IVR and access the *2 and ## feature codes and transfer around, or forward lines, so I disabled those codes. We also we had trunk mode T in advanced options. I took that out. Lastly, I was able to go into the asterisk CLI and remove the forwarding of the extensions via “database del CF (EXT#)”.

I have a feeling that back door is still open. Suggestions going forward? Appreciate any feedback.

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@jcadman wrote:

I have a few Mitel 5330 IP phones running SIP firmware as well as Grandstream IP Phones and when making an external call, the screen shows CID: (My external Number / CID) which then blocks the number I have dialled whilst on the phone. As A few times I have been on the phone I have needed to know the number I have been on to and not been able to due to it showing my caller ID. Is this possible to change to only display the number I have dialled instead of showing me my external number?

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@fanian wrote:

FreePBX 15, newcomer.
I make a backup, but had errors.

[root@freepbx ~]# fwconsole bu --backup cbc69ebd-ef07-4d42-8dc9-972f46abeb2a
Transaction ID is: ae2f91e9-e1b5-4c44-9417-b679ac0a4cc6
Running Backup ID: cbc69ebd-ef07-4d42-8dc9-972f46abeb2a
Transaction: ae2f91e9-e1b5-4c44-9417-b679ac0a4cc6
Starting backup backupLocal
This backup will be stored locally and is subject to maintenance settings
Backup File Name: 20200213-111457-1581581697-15.0.16.42-851846936.tar.gz
Working with amd module
Exporting KVStore from Amd
Adding module manifest for amd
Working with announcement module
Adding module manifest for announcement
Working with areminder module
Adding module manifest for areminder
Adding module ttsengines to queue because areminder depends on it
Working with arimanager module
Adding module manifest for arimanager
Working with backup module
Exporting KVStore from Backup
Adding module manifest for backup
Working with blacklist module
Exporting Feature Codes from blacklist
Adding module manifest for blacklist
Working with broadcast module
Exporting Databases from broadcast
Adding module manifest for broadcast
Working with calendar module
Adding module manifest for calendar
Working with callback module
The module callback returned no data, No backup created
Working with callerid module
The module callerid returned no data, No backup created
Working with callforward module
Exporting Feature Codes from callforward
Adding module manifest for callforward
Working with calllimit module
The module calllimit returned no data, No backup created
Working with callrecording module
Exporting Advanced settings from callrecording
Adding module manifest for callrecording
Working with callwaiting module
Exporting Feature Codes from callwaiting
Adding module manifest for callwaiting
Working with cdr module
/usr/bin/mysqldump --host localhost --user freepbxuser -pbSQinaeZyLDH asteriskcdrdb --opt --compact --table cdr --skip-lock-tables --skip-triggers --no-create-info --result-file=/tmp/dbdump/cdr.sql
Adding directory to tar: /tmp/dbdump
Adding file to tar: files/tmp/dbdump/cdr.sql
Adding module manifest for cdr
Working with cel module
Exporting Advanced settings from cel
Adding directory to tar: /tmp/dbdump
Adding file to tar: files/tmp/dbdump/cel.sql
Adding module manifest for cel
Working with certman module
Adding directory to tar: /etc/asterisk/keys
Adding directory to tar: /etc/asterisk/keys/integration
Adding file to tar: files/etc/asterisk/keys/.rnd
Adding file to tar: files/etc/asterisk/keys/default.crt
Adding file to tar: files/etc/asterisk/keys/default.csr
Adding file to tar: files/etc/asterisk/keys/default.key
Adding file to tar: files/etc/asterisk/keys/default.pem
Adding file to tar: files/etc/asterisk/keys/integration/certificate.pem
Adding file to tar: files/etc/asterisk/keys/integration/webserver.crt
Adding file to tar: files/etc/asterisk/keys/integration/webserver.key
Adding file to tar: files/etc/asterisk/keys/ca.cfg
Adding file to tar: files/etc/asterisk/keys/ca.key
Adding file to tar: files/etc/asterisk/keys/ca.crt
Adding module manifest for certman
Working with cidlookup module
Adding module manifest for cidlookup
Working with conferences module
Exporting KVStore from Conferences
Exporting Feature Codes from conferences
Exporting Advanced settings from conferences
Adding module manifest for conferences
Working with conferencespro module
The module conferencespro returned no data, No backup created
Working with contactmanager module
Exporting KVStore from Contactmanager
Exporting Feature Codes from contactmanager
Exporting Advanced settings from contactmanager
Adding module manifest for contactmanager
Working with core module
Exporting Feature Codes from core
Exporting Advanced settings from core
Exporting KVStore from Core
Adding module manifest for core
Working with cos module
Exporting KVStore from Cos
Exporting Advanced settings from cos
Adding module manifest for cos
Skpping cos which depends on framework because framework is a system requirement. Framework should be removed as a dependency
Working with customappsreg module
Exporting KVStore from Customappsreg
Adding module manifest for customappsreg
Working with dahdiconfig module
Exporting Advanced settings from dahdiconfig
Adding module manifest for dahdiconfig
Working with dashboard module
Exporting Advanced settings from dashboard
Adding module manifest for dashboard
Working with daynight module
Exporting Databases from daynight
Exporting Feature Codes from daynight
Exporting Advanced settings from daynight
Adding module manifest for daynight
Working with dictate module
Exporting Feature Codes from dictate
Adding module manifest for dictate
Working with directory module
Adding module manifest for directory
Working with disa module
Adding module manifest for disa
Working with donotdisturb module
Exporting Feature Codes from donotdisturb
Adding module manifest for donotdisturb
Working with endpoint module
Exporting Advanced settings from endpoint
Exporting Feature Codes from endpoint
Exporting Databases from endpoint
Exporting KVStore from Endpoint

In Backup.php line 6:

RecursiveDirectoryIterator::__construct(/tftpboot
/images): failed to open dir: Нет такого файла ил
и каталога (No such file or directory in English)
What I am doing wrong?

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@amontefiore wrote:

In first call leg sip Intercom correctly sets m=video, a=sendonly
In second call leg sip videophone incorrectly receives m=video, a=sendrecv

Does anyone know what Asterisk/FreePBX would be doing this?

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@TonyX wrote:

Hey guys,

Using a call file without parameters like WaitTime, MaxRetries oder RetryTime all is working perfectly fine. The waittime is like the default configurated and i dont get any other calls if i dont answer the first one because default retries is 0.

The problem is happening when i set e.g. MaxRetries to 3 and RetryTime to 30 or sth. Instead of 4 calls im getting multiple calls imcoming with less retrytime than 15 seconds. Sometimes its like getting simultaneous calls.

My phone rings for about 3 seconds and i can see 5 missed calls.
Waiting for about 10-20 seconds and i can see 20+ missed calls. Checking CDR i can see that a few of the calls are completed as noanswer and some of them ending with answer.
Different WaitTime, RetryTime or anything is making it even worse.

I really have no idea what causes the problem.

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@shokoienia wrote:

Hi All, I’m newbie in FreePBX and have a question about GUI in chrome browser. Today morning as I want to load GUI, the browser can not load …/admin/config.php page and still waiting for server. There aren’t any uapdate or upgrade in server for days. After a search in forum, found an issue that recomends to use another browser and I tried with IE. It was usefull and GUI has been loaded, but “Live Network Usage” window is empty with a message “Loading eth0 Interface …”. The PBX works fine and the calls exists as usual but I want to know, what is the problem with eth0 or probably another parts of system. What should I check to be ensure that everythings is OK?
Thanks in advanced

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