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Calls coming in on 2nd line on soft phone

June 13, 2018, 9:25 pm
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@fabbobabbo wrote:

Hi,

We have several soft phones at our callcenter for our agents. When we have a lot of traffic, sometimes the 2nd line rings. The 2nd line should only be used for transferring calls (attended transfers) because our agent software gets really confused when 2 lines are ringing simultaneously. Is there a general feature to disable incoming calls on more than one line on (soft) phones? Either to restrict incoming calls to only one line, or to disable incoming calls on 2nd line.

Thanks,
Janne

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Outbound call changes to hold music and drops

June 14, 2018, 8:39 am
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@SirDudeofWI wrote:

One out of 500 numbers we call will have the same problem every time. The call is fine until the receiving party tries to switch extensions. At the moment the call is transferred, we hear our hold music for half a second, then the call drops. This happens every time we call the same number.

So, we call the number, we hear their “stay on the line for service” message, then we hear our hold music and the call is dropped.

Help please!!!

This is at the end of the “full” log:

[2018-06-14 10:23:16] SECURITY[2001] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2018-06-14T10:23:16.008-0500”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7f7ae5255158”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/51363”,UsingPassword=“0”,SessionTV=“2018-06-14T10:23:16.008-0500”
[2018-06-14 10:23:16] SECURITY[2001] res_security_log.c: SecurityEvent=“RequestBadFormat”,EventTV=“2018-06-14T10:23:16.033-0500”,Severity=“Error”,Service=“AMI”,EventVersion=“1”,SessionID=“0x7f7ae5255158”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/51363”,RequestType=“Action: DPMALicenseStatus”,SessionTV=“2018-06-14T10:23:16.008-0500”,AccountID=“admin”

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SIP Trunk Inbound getting "The number you have call is out of Service"

June 14, 2018, 1:02 pm
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@VoipKiller wrote:

Hello new here I have Inbound Route/ and SIP Trunk that receives several DID and they are routing to our main IVR of the company but some call are getting “The number you have call is out of Service” ( Intermittent ) below is the configuration of the Inbound route and SIP Trunk

Inbound Route:
Description: Main Corp IVR
DID Number: 1XXXXXXXXXX
CallerID Number:
CID Priority Route:

Alert Info:
CID name prefix:
Music On Hold: PM-BILAT
Signal RINGING:
Pause Before Answer:

Privacy Manager: No
Source: None

Language: None
Set Destination
IVR : Main I VR

SIP Trunk
Trunk Name: NAME
Outbound CallerID: blank
CID Options: Allow my CID
Maximum Channels: Blank
Disable Trunk: Uncheck
Monitor Trunk Failures: Blank

Dialed Number Manipulation Digits BLANK
Trunk Name NAME

Peer-Details
type=peer
nat=no
insecure=very
host=X.X.X.X
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
canredirect=no
allow=ulaw

Any help will be appreciated

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Root Space 100% - Critical

June 14, 2018, 5:05 pm
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@sistemasgesinco wrote:

Hi, everybody.

We (the “Gesinco” IT team) are in a rather conflictive situation.

It seems the /dev/mapper/s7_freepbx-root partition on our Sangoma FreePBX distro (version 12.7.5-1805-2.sng7) is 100% full. We’ve been having alerts at 95%, 96% but it always came back to 90% or so.

Now, it has reached 100% and we’ve been having trouble using the web GUI for administration. Thankfully we are still operational, but it’s something we’ve got to solve.

The question is: How can we expand the size of our root partition in our current filesystem configuration ? Or at least, How can we free some space on the root partition.

As you can see, we have a separate device for call recordings. Then, the tmpfs partitions and the boot partition:

Filesystem Size Used Avail Use% Mounted_on
/dev/mapper/s7_freepbx-root 34G 34G 214M 100% /
devtmpfs 5.8G 0 5.8G 0% /dev
tmpfs 5.8G 4.0K 5.8G 1% /dev/shm
tmpfs 5.8G 481M 5.4G 9% /run
tmpfs 5.8G 0 5.8G 0% /sys/fs/cgroup
/dev/sdb1 1.8T 102G 1.7T 6% /var/spool/asterisk/monitor
/dev/sda1 497M 220M 278M 45% /boot
tmpfs 1.2G 0 1.2G 0% /run/user/2
tmpfs 1.2G 0 1.2G 0% /run/user/0

We were surprised to notice how exponentially the used space on root has been growing.

Maybe there is something we’re missing, maybe something has been filling a great deal of space on the partition and we can’t tell what it is.

Any help or advice will be welcome.

Thanks in advance.

—

The Gesinco IT Team

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System Recording File Formats

June 14, 2018, 9:52 pm
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@adtopkek wrote:

I was looking at system recordings and wondering if I am causing unnecessary processor usage by having to convert audio files on the fly or storing completely useless audio files.

There are the options of: alaw g722 gsm sln sln16 sln48 ulaw wav

Right now we are using wav, gsm, sln, sln16, and ulaw normally. Does anything even load the sln file types? I know wav is used on the UCP and as a raw. Ulaw is the default audio format in the USA. GSM is for compatability. Should recordings have a G722 version when inbound calls seem to universally be ULAW?

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System Admin and Endpoint manager cant be installed

June 14, 2018, 11:52 pm
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@icekid_mpj wrote:

Hi i just installed raspbx on a Raspberry PI 3 Model B with version 14.0.3.6 of freepbx.

Im currently trying to set this up for my home.

im getting the following error when i try to install system admin, which is required to install Endpoint manager:

System Admin cannot be installed:

PHP Component Zend Guard Loader is required but missing from you PHP installation.
The File “/usr/sbin/incrond” must exist.

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Asterisk not load when exists more than 2200 extensions

June 15, 2018, 6:56 am
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@jgilson wrote:

Hello!

I have a server using FreePBX … When I use the system with more than 2200 extensions, the system works fine, but when I reboot the server, a message is printed on the console: Asterisk not currently running. The system is re-established using the fwconsole restart command. Has anyone had this problem or know how to solve it?

Thank you!

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Conference Bridge Issues

June 15, 2018, 8:24 am
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@assyrian47 wrote:

I have a simple configuration that consists of three extensions and three conference bridges.  This is to allow smartphones to join a conference while in the field.  There are no trunks. The idea is to allow 4-5 users conference. I have the extensions pointing to the conference numbers.  

Extension 1001 >> Conference 5060
Extension 1002 >> Conference 5061
Extension 1003 >> Conference 5062

1001@pbx.domain.com

The logs indicate the connection is made to the pbx but the connection is dropped.  When I look at the conference settings. I get this error

No such command 'confbridge list' (type 'core show help confbridge list' for other possible commands)

What am I doing wrong?

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Cant connect to local web interface in Raspbx

June 16, 2018, 9:09 am
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@icekid_mpj wrote:

Hi im working on my RasPBX server used for my home server and i was setting up a soundpoint ip 450 in the process, then, the interface got reset.

Raspberry PI 3 Model B

Exception (2002)
SQLSTATE[HY000] [2002] Can’t connect to local MySQL server through socket ‘/var/run/mysqld/mysqld.sock’ (2)::SQLSTATE[HY000] [2002] Can’t connect to local MySQL server through socket ‘/var/run/mysqld/mysqld.sock’ (2)
and when i go to the directory, theres nothing in there. and mysql is not running when i type mysql and basically gives me the same error.

Stack frames (8)
7
Exception
/var/www/html/admin/libraries/utility.functions.php204
6
die_freepbx
/var/www/html/admin/libraries/BMO/Database.class.php142
5
PDOException
/var/www/html/admin/libraries/BMO/Database.class.php137
4
PDO __construct
/var/www/html/admin/libraries/BMO/Database.class.php137
3
FreePBX\Database __construct
/var/www/html/admin/libraries/BMO/FreePBX.class.php71
2
FreePBX __construct
/var/www/html/admin/bootstrap.php153
1
require_once
/etc/freepbx.conf9
0
include_once
/var/www/html/admin/config.php100

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Configure pri trunk from tatateleservices

June 17, 2018, 3:59 am
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@shivaramkar wrote:

Hi,
I am trying to configure PRI line from Tatateleservices from india.
I would like to know what are the settings i need to use for PRI line. The hardware interface is Digium 2 span 1TE235BF model. the device is listed as Two (2) span digital T1/E1/J1/PRI PCI-Express x1 card and hardware echo cancellation (VPM064) in freepbx.
Help needed on this.

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Remove outbound route prefix from CDR

June 18, 2018, 11:58 am
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@jgilson wrote:

Hello,

How can I remove the prefix used on outbound route from CDR. The rule is: prefix 0 and match pattern ..
When the dialed number 088884444 is used, asterisk send to trunk Dial(“SIP/1002-00000086”, “SIP/9999/88884444,300,T”) in new stack, the CDR ticket is generated with 088884444 and the number 088884444 is received on the trunk.
The help says: Prefix to remove on a successful match. The dialed number is compared to this and the subsequent columns for a match. Upon a match, this prefix is removed from the dialed number before sending it to the trunks.
Is it possible to remove this 0 from the ticket?

Thanks!

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Contact Manager error after recient updates

June 18, 2018, 12:20 pm
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@westcana wrote:

Recently (june 11/2018) updated freepbx 13.18.0 to current versions of all modules, using the Freepbx Module Admin. All modules are on Stable track.
I didn’t make any changes to the configuration of the system, other that applying the updates and updating the phone firmware. (All phones are Sangoma S500’s)
After the updates, users complained that when they try to transfer to voicemail, the list of extensions/users no longer displays anything.
After digging around for awhile, I discovered that if I go into the contact manager in the administrator web interface, it throws an error.
SQLSTATE[42S22]: Column not found: 1054 Unknown column ‘n.countrycode’ in ‘field list’
Various other menu’s related to the contact list in other modules will throw the same error.

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Delete Inbound Route from Command Line

June 19, 2018, 12:28 am
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@blaskurdo wrote:

Hi, I’m looking for a Bash code that allow delete some ‘Inbound Route’ from command line. I found this php code of ‘lgaetz’ that do the same, but for extensions. There is something similar but for ‘inbound routes’?

Thanks.

#!/usr/bin/env php
<?php
if (!isset($argv[1])){
        echo "
 ***** ***** ***** ***** ***** ***** ***** ***** ***** ***** ***** ***** ***** ***** *****
 *
 * Script: extension-del.php
 *
 * Latest version: https://gist.github.com/lgaetz/f2d3d717520f8adf2018643473d1f748
 *
 * Usage: Run at bash prompt of FreePBX system running FreePBX 13+ with single argument
 *        of extension to be deleted. Follow with fwconsole reload for changes to be applied
 *
 *        # extension-del.php 3002
 *
 * License: GNU/GPL3+
 *
 * History:
 *         2017-09-24   First commit by lgaetz
 *         2017-09-25   Update for virtual extensions
 *         2017-09-26   General polish
 ***** ***** ***** ***** ***** ***** ***** ***** ***** ***** ***** ***** ***** ***** *****
";
exit;
}

include '/etc/freepbx.conf';
$FreePBX = FreePBX::Create();
$device=$FreePBX->Core->getDevice($argv[1]);
$user=$FreePBX->Core->getUser($argv[1]);
if($device["user"]){
        // normal extensions tested with sip, pjsip and dahdi
        echo "Found device ".$argv[1].", deleting...\n";
        echo "Found user ".$device["user"].", deleting...\n";
        $foo=$FreePBX->Core->delDevice($argv[1]);
        $foo=$FreePBX->Core->delUser($device["user"]);
} elseif ($user) {
        // for extensions with users but no device i.e. virtual
        echo "Found user ".$user['extension'].", deleting...\n";
        $foo=$FreePBX->Core->delUser($argv[1]);
} else {
        echo "Neither user nor device ".$argv[1]." exists, exiting...\n";
}

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Cell phone call as notification about email

June 19, 2018, 4:59 am
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@akononov85 wrote:

Hello.
Could you help me.
I have the task of organizing notifications about important email messages by calling a mobile phone. Could you tell me, is it possible to organize this without using custom scripts?

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FreePBX install corrupts microSD card when wifi busy

June 19, 2018, 9:28 am
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@ddobbins wrote:

We’re using FreePBX on rPi server and 60 speakers for general PA announcing and bells in our school. The speakers are all wifi. System works great until the wifi gets crowded with traffic (1:1 Chromebooks for 200 users), then we hear a corrupted bell signal which tells me the microSD card needs replacement with another good copy of the install, and system works once again. This happens 1-2 times per week when wifi activity levels are all high, but since school is out for a month, the system has been up 24x7 no interruptions. The manuf of system is Innovation Wireless and they have not been able to resolve this; suggested I try here. FreePBX 12.0.76.2 VoIP Server.

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DADHI CONFIGS Help keep the settings after upgrade or modify

June 20, 2018, 2:54 am
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@fazmalj wrote:

Ok new to freepbx but all was working well.

I have now installed a sangoma A200 with ech cancel and all working well. I did initially have a problem of delayed rings etc but trolling the forums post figured that as i use UK BT i need to include the following

in chan_dadhi_conf

cidsignalling=v23
cidstart=polarity

Now all working well but every time i make changes to system or update apply settings i need to redo the chan_dadhi conf with the above lines.

Now i am trying to understand which conf file i can adjust or edit to keep changes perminant.

If it can be done throug GUI all the better.

Just need the guidence otherwise all is running well.

Thanks to all in advance

Freepbx version 13.0.194.10
Asterisk 11.25.1
Endpoint manager 13.0.118.17 Stable

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Create linux user in FreePBX's OS

June 20, 2018, 8:04 am
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@pasi wrote:

I am running FreePBX 13.0.194.10 on the SHMZ release 6.6 (Final) operating system.
I want to create a login user account with the permission to do the following:
1- Copy a __.call file to /tmp directory
2- Move the __.call file from /tmp directory to /var/spool/asterisk/outgoing/
The __.call file is owned by the asterisk user.
What is the best way to archive this?

p/s: an trying to automate an auto dial out phone call for alerting.

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Central Management and monitoring of FreePBX systems

June 20, 2018, 11:00 am
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@dragonparoxysm wrote:

Hey everyone!

Has anyone heard of a solution (whether made by Sangoma or not) to manage multiple FreePBX or PBXact systems?
It would be ideal for notifications, bulk updates, provisioning servers from templates etc…

I have done some research on it and found that Sangoma released their RMS solution a while back, has anyone had experience with this?

I have also run into “M3 Multi Machine Management Server” but it says on their website that support has ended in 2015.

Any current info on planned management systems would be greatly appreciated! Thanks.

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Failed to authenticate on INVITE to with Nexmo

June 20, 2018, 6:25 pm
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@unclesamsfw wrote:

Migrating VoIP providers to Nexmo, have incoming calls working great. Outbound calls fail with:

chan_sip.c: Failed to authenticate on INVITE to '<sip:ourNumber@OurExternalIP>;tag=as7130c686'

Outgoing SIP Settings (From Nexmo):

host=sip nexmo com
type=friend
insecure=port,invite
qualify=yes
allow=ulaw,alaw,g729
dtmfmode=rfc2833
username=xxx
fromuser=xxOurNumber
secret=xxx

Removed the periods from the host line, the forums don’t like new users posting URLs.

Have ports 10,000-20,000 forwarded, 5060 forwarded… I’ve Googled around for answers but haven’t had any luck. Not sure what other information is useful here, any help or suggestions are most welcome.

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Backup and restore

June 21, 2018, 12:08 am
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@klrgirish198117 wrote:

hi sir i am using FreePBX 13.0.192.19 is licensed under the GPL
Copyright© 2007-20.

i am looking for backup and restore the extension in ssh command line.

thanks & Regards

girish

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