@josephchrz wrote:

Hello I’m running a old freepbx setup from years ago. Someone help me to setup the tftp server and oss end point manger. But I’m in the proccess of settuping up the latest freebpx. I saw how to enable the tftp from SkyKingOH he put

tftp server:

/etc/xinet/d/tftp - Modify as below:

[root@maieast xinetd.d]# cat tftp
default: off
description: The tftp server serves files using the trivial file transfer
protocol. The tftp protocol is often used to boot diskless
workstations, download configuration files to network-aware printers,
and to start the installation process for some operating systems.
service tftp
{
socket_type = dgram
protocol = udp
wait = yes
user = root
server = /usr/sbin/in.tftpd
server_args = -v -s /tftpboot
disable = no
per_source = 11
cps = 100 2
flags = IPv4
}
[root@maieast xinetd.d]#

Run the command ‘service xinetd restart’
You can tail the messages file to see the tftp requests tail -f /var/log/messages
I’m guessing to enable it. But i was wondering there is anything else i would i would need to get tftp server to work? One thing i remember something about DHCP 150? I don’t remember what is that for?

Joseph

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@xwebnetwork wrote:

Been doing some research and just want to get some input regarding a FreePBX setup for a 16 room hotel. Right now, we have a Comdial system and over the years, before purchasing the place, wires have been run, and re-run, and left behind so in an effort to clean up all of the crawl spaces and what not, the phone system immediately comes to mind. Ideally, I’d like to scrap the whole Comdial system in favor of a wireless solution so I have been looking into FreePBX as an option. I’ve got solid WIFI setup throughout the place and have plenty of remote server resources that I can use to deploy a VM with the PBX software. My question to those of you whom are well versed, do you you think this is a viable solution? From what I have read, seems like this would be perfect. Anyways, just to recap:

• Scrap old wired, digital Comdial system
• Deploy remote VM with FreePBX server
• Purchase some inexpensive WIFI SIP phones and provision 1 per room with room # as extension (ie, 101, 102…)
• The room phones will only be able to dial internal extensions, primarily 0 just to get to the front office
• The primary office phone will be the ONLY phone to have multiple lines, internal and external, as it will also connect to the primary public facing numbers

Anyways, thank you in advance!

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@bksales wrote:

I feel like this is going to be something stupid but I cant find it and I’m sure its in the PBX. I ran the conversion tool on an old PBX and after that point I started seeing messages like this in the maillog

Nov 20 14:52:15 atlas postfix/smtp[19704]: B24ED2169B35: to=<email@url.com >, relay=mailstore1.secureserver.net[72.167.238.32]:25, delay=0.61, delays=0.15/ 0.02/0.45/0, dsn=4.0.0, status=deferred (host mailstore1.secureserver.net[72.167 .238.32] refused to talk to me: 421 p3plibsmtp01-15.prod.phx3.secureserver.net C MGW Temporarily rejected. Reverse DNS for failed. IB108 <http:// x.co/srbounce>)

IP of PBX and my email address are removed in this post but are correct in the log entry. Confirmed that they’re in fact using Go Daddy (72.167.238.32). I also confirmed that I have the reverse DNS PTR record created. Using the same DNS server as before in ifcfg-eth0 (8.8.8.8).This one appears to be set up identically to another server that is working, at least everywhere that I’ve looked so far.

Any suggestions are welcomed.
FreePBX 14.0.13.12

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@VoIPTek wrote:

Hello All,

I have an issue that happened once I upgraded to FreePBX 15.
I have a running IAX from FreePBX 13 to what was 14 for a long time, now after the upgrade 15 can call 13 no problem, but 13 can not call 15.

Are there any changes to take into consideration on 15?
I did notice under 15 there is a new IAX Settings menu and from what I can tell, there wasn’t anything I needed to change, leaving all blank per say.

My standard outgoing IAX looks like this:
username=IAX-2-PBX123
type=friend
trunk=yes
secret=THEPASSWORD
requirecalltoken=no
host=10.1.0.1
disallow=all
context=default
allow=ulaw

incoming:
username=PBX123-IAX
secret=THEPASSWORD
host=10.2.0.1
type=peer
trunk=yes
context=from-internal
requirecalltoken=no
disallow=all
allow=ulaw

Any insight appreciated.

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@daansk44 wrote:

For some reason, the port 5060 seems to be closed and I do not know exactly why.
When I run an nmap command I get this back

Starting Nmap 6.40 ( http://nmap.org ) at 2019-11-07 18:30 UTC
Nmap scan report for localhost (127.0.0.1)
Host is up (0.0000080s latency).
Other addresses for localhost (not scanned): 127.0.0.1
Not shown: 983 closed ports
PORT     STATE SERVICE
22/tcp   open  ssh
25/tcp   open  smtp
53/tcp   open  domain
80/tcp   open  http
81/tcp   open  hosts2-ns
82/tcp   open  xfer
83/tcp   open  mit-ml-dev
84/tcp   open  ctf
111/tcp  open  rpcbind
443/tcp  open  https
3306/tcp open  mysql
4000/tcp open  remoteanything
5000/tcp open  upnp
5222/tcp open  xmpp-client
8001/tcp open  vcom-tunnel
8088/tcp open  radan-http
8089/tcp open  unknown

You can see there is no port 5060

image.png701×936 34.6 KB

When I look at the specefic port it is saying that it is closed

[root@freepbx ~]# nmap -v -sV localhost -p 5060

Starting Nmap 6.40 ( http://nmap.org ) at 2019-11-07 18:46 UTC
NSE: Loaded 23 scripts for scanning.
Initiating SYN Stealth Scan at 18:46
Scanning localhost (127.0.0.1) [1 port]
Completed SYN Stealth Scan at 18:46, 0.00s elapsed (1 total ports)
Initiating Service scan at 18:46
NSE: Script scanning 127.0.0.1.
Nmap scan report for localhost (127.0.0.1)
Host is up (0.00012s latency).
Other addresses for localhost (not scanned): 127.0.0.1
PORT     STATE  SERVICE VERSION
5060/tcp closed sip

Read data files from: /usr/bin/../share/nmap
Service detection performed. Please report any incorrect results at http://nmap.org/submit/ .
Nmap done: 1 IP address (1 host up) scanned in 0.16 seconds
           Raw packets sent: 1 (44B) | Rcvd: 2 (84B)

With a netstatus I get the next

[root@freepbx ~]# netstat -na |grep 5060
udp        0      0 0.0.0.0:5060            0.0.0.0:*
[root@freepbx ~]#

Where is it going wrong?

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@tgross wrote:

Is there a command that allows you to set a trunk as disabled but wait until it is idle?

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@hook wrote:

FreePBX 14
Framework v 14.0.13.12
Ubuntu Server

Followed directions on the Wiki for installation on Ubuntu. Loaded a backup that I had from a Raspberry PI PBX we were using.

Getting an error in the Dashboard notification area:

Module: “FreePBX Framework”, File: “/var/lib/asterisk/agi-bin/phpagi-asmanager.php missing”
Module: “FreePBX Framework”, File: “/var/lib/asterisk/agi-bin/phpagi.php missing”

I was also getting an error related to retrieve_conf and had to manually add /var/lib/asterisk/agi-bin folder and those errors went away.

I have run:
fwconsole ma upgradeall
fwconsole chown

I have also run:
fwconsole ma refreshsignatures

Let me know what you think and if there is any way you can help I would appreciate it!

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@daansk44 wrote:

For some reason, the port 5060 seems to be closed and I do not know exactly why.
When I run an nmap command I get this back

Starting Nmap 6.40 ( http://nmap.org ) at 2019-11-07 18:30 UTC
Nmap scan report for localhost (127.0.0.1)
Host is up (0.0000080s latency).
Other addresses for localhost (not scanned): 127.0.0.1
Not shown: 983 closed ports
PORT     STATE SERVICE
22/tcp   open  ssh
25/tcp   open  smtp
53/tcp   open  domain
80/tcp   open  http
81/tcp   open  hosts2-ns
82/tcp   open  xfer
83/tcp   open  mit-ml-dev
84/tcp   open  ctf
111/tcp  open  rpcbind
443/tcp  open  https
3306/tcp open  mysql
4000/tcp open  remoteanything
5000/tcp open  upnp
5222/tcp open  xmpp-client
8001/tcp open  vcom-tunnel
8088/tcp open  radan-http
8089/tcp open  unknown

You can see there is no port 5060

image.png701×936 34.6 KB

When I look at the specefic port it is saying that it is closed

[root@freepbx ~]# nmap -v -sV localhost -p 5060

Starting Nmap 6.40 ( http://nmap.org ) at 2019-11-07 18:46 UTC
NSE: Loaded 23 scripts for scanning.
Initiating SYN Stealth Scan at 18:46
Scanning localhost (127.0.0.1) [1 port]
Completed SYN Stealth Scan at 18:46, 0.00s elapsed (1 total ports)
Initiating Service scan at 18:46
NSE: Script scanning 127.0.0.1.
Nmap scan report for localhost (127.0.0.1)
Host is up (0.00012s latency).
Other addresses for localhost (not scanned): 127.0.0.1
PORT     STATE  SERVICE VERSION
5060/tcp closed sip

Read data files from: /usr/bin/../share/nmap
Service detection performed. Please report any incorrect results at http://nmap.org/submit/ .
Nmap done: 1 IP address (1 host up) scanned in 0.16 seconds
           Raw packets sent: 1 (44B) | Rcvd: 2 (84B)

With a netstatus I get the next

[root@freepbx ~]# netstat -na |grep 5060
udp        0      0 0.0.0.0:5060            0.0.0.0:*
[root@freepbx ~]#

Where is it going wrong?

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@tbagalini wrote:

I am running freepbx Centos on a GCE VM. Syncing stops when FreePBX/Asterisk tries to write and create a folder in the mounted folder and FreePBX is unable to write / create folder in the folder. Everything works again when the system is rebooted because gcsfuse is no longer mounted.

Gcsfuse works when: .

-create a folder /var/spool/asterisk/cloud.
-mount it using gcsfuse --implicit-dirs siphqrecordings /var/spool/asterisk/cloud.
-create a sample text file nano test.txt.
-it syncs up to my bucket just fine

It completely stops working when:

-FreePBX tries to place a file and folder in the directory (a recorded call), the folder and file never gets written. I have unmount/reboot the server and then freepbx is able to use the folder and write files and directories to it. Even if I try to create a manual file using nano it never sync up.

FreePBX is unable to write to folder mounted with gcsfuse and the sync stops even for manually created files.

Thank you!

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@bksales wrote:

This is a new one for me. They want to do their annual meeting virtually and want to use the PBX to have everyone be able to call in and listen while muted. Not sure what the practical and theoretical limits on the PBX are or what the best solution would be. I know we used the broadcast module last week to do 90 concurrent calls and that seemed to work fine, but I’m nervous about trying 1000 concurrent calls let alone 1000 members of a conference bridge.

Any ideas would be appreciated.

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@waseemly wrote:

Hi

For some strange reason i have started to get the below error. It happens when i try and do apply config in freepbx frontend

I did delete some files in tmp the other day but to try and fix it i did create a new cron.error in tmp and set to ownership and grp to asterisk, still no joy

   Unable to continue. touch(): Unable to create file /tmp/cron.error because No such file or directory in /var/www/html/admin/libraries/BMO/Cron.class.php on line 279
    #0 [internal function]: Whoops\Run->handleError(2, 'touch(): Unable...', '/var/www/html/a...', 279, Array)
    #1 /var/www/html/admin/libraries/BMO/Cron.class.php(279): touch('/tmp/cron.error')
    #2 /var/www/html/admin/libraries/BMO/Cron.class.php(164): FreePBX\Cron->installCrontab(Array)
    #3 /var/www/html/admin/libraries/Builtin/UpdateManager.php(231): FreePBX\Cron->remove('11 4 * * * [ -e...')
    #4 /var/lib/asterisk/bin/retrieve_conf(1012): FreePBX\Builtin\UpdateManager->updateCrontab()
    #5 {main}

Also in asterisk CLI i get the following when doing apply config in freepbx

- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected

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@ITconsultant wrote:

in freepbx GUI when I click on reports and then I click on asterisk info and then on the right I click on peers I can see my sip trunk IP address on 5060 but how can I see what IP address the media is going to

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@shakawkaw wrote:

Hi

Sometimes a module app_queue.so crashes with an error. And I can not find a reason for this behavior. Possible, someone will be able to indicate what I’m doing wrong.

# dmesg -T | grep -P ‘segfault’
[Wed Nov 13 10:33:52 2019] asterisk[18466]: segfault at 10 ip 00007f21859aad9c sp 00007f21707aeb40 error 4 in app_queue.so[7f2185992000+35000]
[Wed Nov 20 15:13:52 2019] asterisk[310]: segfault at 10 ip 00007feff3540d9c sp 00007fefd19b7b40 error 4 in app_queue.so[7feff3528000+35000]
[Thu Nov 21 10:06:08 2019] asterisk[10940]: segfault at 10 ip 00007f9888f67d9c sp 00007f982f9a5b40 error 4 in app_queue.so[7f9888f4f000+35000]

next i’m get backtrace prom core dump

# gdb /usr/sbin/asterisk /tmp/core.AsterPBX.localdomain-2019-11-21T10:06:12+0300 > ~/core.AsterPBX.localdomain-2019-11-21T10-06-12.new
# gdb /usr/sbin/asterisk /tmp/core.AsterPBX.localdomain-2019-11-20T15:13:56+0300 > ~/core.AsterPBX.localdomain-2019-11-21T15-13-56.new
# gdb /usr/sbin/asterisk /tmp/core.AsterPBX.localdomain-2019-11-13T10:33:56+0300 > ~/core.AsterPBX.localdomain-2019-11-13T10-33-56.new

and get error function from backtrace files

# cat ~/core.AsterPBX.localdomain-2019-11-*new | grep -A 3 Program
Program terminated with signal 11, Segmentation fault.
#0 0x00007f21859aad9c in handle_hangup (userdata=userdata@entry=0x7f21400c5a60, sub=sub@entry=0x7f21401e1460, msg=msg@entry=0x7f224407b910) at app_queue.c:6235
6235 ast_queue_log(queue_data->queue->name, caller_snapshot->uniqueid, queue_data->member->membername,
(gdb) quit

Program terminated with signal 11, Segmentation fault.
#0 0x00007feff3540d9c in handle_hangup (userdata=userdata@entry=0x7ff07005dce0, sub=sub@entry=0x7ff0700d36b0, msg=msg@entry=0x7ff040154140) at app_queue.c:6235
6235 ast_queue_log(queue_data->queue->name, caller_snapshot->uniqueid, queue_data->member->membername,
(gdb) quit

Program terminated with signal 11, Segmentation fault.
#0 0x00007f9888f67d9c in handle_hangup (userdata=userdata@entry=0x7f98e00c00f0, sub=sub@entry=0x7f98e0018ec0, msg=msg@entry=0x7f98f805d340) at app_queue.c:6235
6235 ast_queue_log(queue_data->queue->name, caller_snapshot->uniqueid, queue_data->member->membername,
(gdb) quit

I understand that the problem is in the function “handle_hangup”, but what to do next I can not understand.

asterisk 13.19.1
CentOS Linux release 7.5.1804 (Core) 3.10.0-862.2.3.el7.x86_64
FreePBX framework 14.0.13.12

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@majbom wrote:

Hi

I’m trying to write my own little API so that I can create extensions and users from my external web application.

I have tried this:
$settings = [
‘extension’ => $extension,
‘devinfo_secret’ => $password,
‘name’ => $name,
‘tech’ => ‘pjsip’,
‘devinfo_sipdriver’ => ‘chan_pjsip’,
‘userman_directory’ => 1,
‘userman_assign’ => ‘add’,
‘userman_password’ => $password,
‘userman_group’ => [1]
];
$FreePBX = FreePBX::Create();
$FreePBX->Core->addUser($settings[‘extension’], $settings);
$FreePBX->Core->addDevice($settings[‘extension’], $settings[‘tech’], $sip);
$FreePBX->Core->addDevice(‘99’ . $settings[‘extension’], $settings[‘tech’], $webrtc);

The variables $sip and $webrtc are arrays filled with keys/values, that looks just like those FreePBX uses when creating through the webgui.

But neither the “normal” extension or the “webrtc” extension can register.

I have looked deep in the FreePBX code, but I think it’s pretty messy with both procedural and OO mixed together and some functions returns/echos
<script>javascript:alert([...]

I really hope that someone can give some input here.

Thanks

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@geeklessnerd wrote:

old system PBX iaf (Fpbx13 )… recordings were .wav and .mp3 and located at /var/lib/asterisk/sounds/custom/

i tar’d and moved to the same location in the new box (FreePBX distro 13/16)…

now normally (on PBXiaf) they would simply show up as available for use in Announcements after being dropped in the correct folder… not so now…
anything I need to do? or do they need to go to a different folder?

tx!

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@GSnover wrote:

[2019-11-25 13:54:02] WARNING[96942][C-000002e0]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected ‘=’, expecting $end; Input:
= 1 & 0 = 0
^
[2019-11-25 13:54:02] WARNING[96942][C-000002e0]: ast_expr2.fl:474 ast_yyerror: If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
– Executing [s@macro-outbound-callerid:24] ExecIf(“SIP/239-000005ea”, “?Set(CALLERID(all)=239)”) in new stack
[2019-11-25 13:54:02] WARNING[96942][C-000002e0]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected ‘=’, expecting $end; Input:
= 1 & 0 = 0
^
[2019-11-25 13:54:02] WARNING[96942][C-000002e0]: ast_expr2.fl:474 ast_yyerror: If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables

FreePBX14 box with all modules and updates current - Core 14.28.19. Asterisk 13

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@GSnover wrote:

[2019-11-25 15:36:02] WARNING[15402][C-0000000d]: chan_sip.c:16755 __set_address_from_contact: Invalid contact uri (missing sip: or sips:), attempting to use anyway
[2019-11-25 15:36:02] WARNING[15402][C-0000000d]: chan_sip.c:16768 __set_address_from_contact: Invalid URI: parse_uri failed to acquire hostport

Is this coming from me (FreePBX) or is it coming from the Trunking Provider? My suspicion is from the trunking provider.

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@mniess wrote:

Hi,

on my machine all cores run php /var/lib/asterisk/bin/schedtc.php with full CPU usage. This seems to be a recurring problem. Running it by hand (as user asterisk) resulted in a PHP stacktrace pointing to a SysV semaphore exception.

Shutting down and rebooting freepbx helped, but I’d still like to investigate this further, so any pointers are greatly appreciated.

Thanks,
Matthias

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@asyscom1 wrote:

Hello,
this is my situation

PBX1 with 15 lines PRI and IVR
PBX2 with 2 lines BRI and IVR

The two systems are independent but being 2 partner companies can work together and the extensions are called to each other.

The PBX2 having only 2 lines available asked me for this possibility.

Create on the pbx1 two new numbers to give to the customers, when the numbers are called the call must go on the trunk iax that is between the 2 PBXs and go to the queue present in the pbx2 but passing first from the time control.

Until now I managed to transfer the call via trunk iax and get the queue to answer but I can’t send it to the time group first

This in detail:
PBX1

/etc/asterisk/extensions_custom.conf
[From-pstn-custom]
exten => _XXXXXX900,1, Dial (IAX2 / 192.168.2.160 / $ {EXTEN: 1}, tT)
exten => _XXXXXX900, n, HangUp
exten => _XXXXXX901,1, Dial (IAX2 / 192.168.2.160 / $ {EXTEN: 1}, tT)

PBX2

/etc/asterisk/extensions_custom.conf
exten => _XXXXXX900,1, Dial (SIP / 900) (where 900 is the queue number)
exten => _XXXXXX900, n, HangUp
exten => _XXXXXX901,1, Dial (SIP / 901) (where 901 is the queue number)
exten => _XXXXXX901, n, HangUp

Is it possible to send the call to the queue time group instead of directly to the queue?

P.S
the XXXX are only to mask the public number
Thanks in advance

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@SITAU wrote:

Is there a way to stop this from occuring? Read on…

I have an issue where inbound queued ringing calls get connected to a parked call if the user tries to pickup a previously parked call while his handset is ringing as part of said queue. Happens on all handsets. End result, one inbound caller ends up talking with another inbound caller whilst there is no evidence that either call exist on handset.

User parks a call by pressing the corresponding button in queue. Pick up (same call) from queue by pressing same park button.

Handsets are Aastra 6737i.
Parking is default setup on server. Parking Lot 70 with 9 slots 71-79.
Parking buttons on phones are setup as BLF 71 to BLF 79 using softkeys.
Free PBX version 14.0.13.6
Asterisk version 16.4.1
Using PJSip for extensions and trunks.

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