@StephanK wrote:

Hi, besides my Sipstaion and Twillio trunks I have a dedicated fax line from an old fashion phone company connected through a hardware card.

I also have an extension for our fax machine on the same card.

Incoming calls get routed to the fax directly.

What is the easiest way to set up that this extension only uses this line for outgoing calls?

Thanks!

(FreePBX 13, Asterisk 13)

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@gwntc wrote:

I've got an issues with a FreePBX 50 system with built in FXO card. The client is telling me that when they call out a lot of the time the phone just ring and rings and rings but nobody ever answers. They then call with their cell phone and the person picks up and tells them they didn't get a call with the previous attempt.

Anyone ever see this?

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@florianahm wrote:

After i screw it trying things in freepbx, i did a fresh install and then i restore from a fullbackup did a week later.

i installed FreePBX-6.12.65-30 and did 2 upgrades that was available now is FreePBX-6.12.65-32, the same version it was running, then i did the restore and now im having error:

retrieve_conf failed, config not applied,
Reload failed because retrieve_conf encountered an error: 1

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@Eres wrote:

Hi,

I recently installed a Cisco SPA112 and registered a extension with no problems. The only problem is with Call Transfers. It is not possible to put the call on hold with, in this case, the R button (Siemens Gigaset). The same issue occurs when using a other type of Analog Phone. Also holding the # doesn't work. You can hear a "click" when you try to put a call on hold (the one who is putted to hold).

I've searched the internet but could not find a solution to this.

If any logs are needed just tell me which one and ill provide.

Provider -> Elastix -> SIP -> SPA112 -> Gigaset C590 (Analog)

Asterisk 11.21.0
elastix-4.0.0-1.noarch
elastix-system-4.0.0-9.noarch
Cisco SPA122 Version 1.4.1

Hope you can help me with this.

btw: im not a expert in this

F* me.. on the dect handset i needed to set the flashtime higher (400) than default (100)

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@byrnejb wrote:

FreePBX distro 10.13.66-17
FreePBX 13.0.190.7
Asterisk 11.24.1
Jitsi 2.8.5426

When attempting to connect a softphone I am seeing these errors in the Asterisk logs:

SIP Peer ACL: Rejecting 'a.b.c.d' due to a failure to pass ACL '(BASELINE)'
Registration from '"X" <sip:312@example.com>' failed for 'a.b.c.d:56534' - Device does not match ACL

I can find no explanation as to what this means *(ACL==Access Control List? Which one? Where is it defined?) and so I have no clue as to why this has occurred. This software and sip account have been used together from the same laptop to our Asterisk server before without this difficulty. Other than updating the system to the current channel I have made no configuration changes since the last successful connection, which admittedly was some time ago.

SOLUTION:

Advanced Device settings

Deny = 0.0.0.0/0.0.0.0
Permit = x.y.z.0/255.255.255.0&a.b.c.d/255.255.255.255

I had disabled access to this device from anything other than x.y.z0/255.255.255.0 so I added a.b.c.d/255.255.255.255 and :bob's your uncle - we're in".

Too many types of ACLs for my poor head to recall.

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@andy_woolford wrote:

I have set up some rules in iptables for port forwarding and masquerading. These have been saved in the iptables startup file using:

service iptables save

The freepbx firewall can be started using:

fwconsole firewall start

...however this flushes all of my existing rules and replaces them with the freepbx firewall rules.

Is there a method of starting the freepbx firewall which can preserve my default rules without flushing them?

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@asteroid4u wrote:

Hi,

We are using freepbx and I want to blacklist outbound and In bound calls. I am trying to install blacklist module but it is failing with below error

Please wait while module actions are performed
Upgrading blacklist to 13.0.14 from track stable
Downloading blacklist 44757 of 44757 (100%)

Error(s) downloading blacklist:

File Integrity failed for /var/www/html/admin/modules/_cache/blacklist-13.0.14.tgz.gpg - aborting (GPG Verify File check failed)

FreePBX 13.0.1beta3.45 'VoIP Server'
CentOS 6

Please help

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@shoobi wrote:

can u create ivr menu system in freepbx? help needed !

Screenshot_5.png1313x256 13.5 KB

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@cappilio wrote:

Hi Guys, Sorry if this has already been discussed in details but i cant seem to search the forum topics.
I did some previous google searches and find a few topics that help describe what i want to do, but i cant seem to get the solution working so im hoping someone here can highlight what i am doing wrong.

i'm simply trying to get the phone directory button on Cisco 7941 Phones working,
I have been looking at these two sets of instructions to help me complete;

CANT POST LINKS [LINK REMOVED]
CANT POST LINKS [LINK REMOVED]

My environment is two freepbx servers;

Server 1;
PBX Firmware: 1.810.210.57-1
PBX Service Pack: 1.0.0.0

Server 2
PBX Firmware: 1.813.210.58-1
PBX Service Pack: 1.0.0.0

So i require the directory button to probe both servers extension list, according to the first resource i need to do this via php script, now i have literally 0 skills with scripting or coding at any level so i already feel out of my comfort zone. but through a process of copy and paste and replacing the blanks i think i managed to get the script working

now on each server when i visit /localhost/search.php i see an output of all the users and extension numbers like;


Joe Bloggs- 4119
4119


Joey Bloggs- 4118
4118


Jospeh Bloggs- 5031
5031


Jerry Bloggs- 4103
4103

so im assuming the script is doing as it should,

now i have to add that configuration to my phones; according too CANT POST LINKS [LINK REMOVED] i do this in SIPDefault.cnf

looking at my SIPDefault.cnf the only line i see that relates to directory is;
directory_url: "CANT POST LINKS [LINK REMOVED]/menu2.xml"

this xml file looks like the following;

Phone Directory
Enter The Search Criteria
CANT POST LINKS [LINK REMOVED]/ciscodir/search.php

Enter Name
sn
U

so this includes the search.php originally created which looks like the following;
<?php
header ("content-type: text/xml");
// Created by Intuit
// with credit to JOYCE CR, s.r.o.CANT POST LINKS [LINK REMOVED]
// Make sure you configure the allowable settings only
// This script directly integrates with FreePBX and picksup the asterisk.users table
// Should work for both device-user mode or extensions mode
// Works by searching from anywhere of the person's name
// feedback to CANT POST LINKS [LINK REMOVED]

// Change here to match the webaddress absolute path
$URL = 'server1ip/ciscodir/';

// Choose how many results to return if search term produces a lot of output
$per_page = '100';

// Change here to match your own passwords
$mysql_conn = mysql_connect('server2ip','freepbxuser','password');
$mysql_conn2 = mysql_connect('server1ip','freepbxuser','password', true);

// Dont change anything from here unless you know what you are doing
mysql_select_db('asterisk', $mysql_conn );
mysql_select_db('asterisk', $mysql_conn2 );

$NAME=$_GET["sn"];
$FROM=$_GET["FROM"];
$TO=$_GET["TO"];
if ( ($FROM=='') and ($TO=='') )
{
//check to see how many
$result= mysql_query("SELECT count(phones.Remark) as total
FROM phones
WHERE phones.Remark LIKE '%$NAME%' ", $mysql_conn);
$howmany = mysql_fetch_row($result);
$result2= mysql_query("SELECT count(users.name) as total
FROM users
WHERE users.name LIKE '%$NAME%' ", $mysql_conn2);
$howmany2 = mysql_fetch_row($result2);

if (($howmany[0] > $per_page) || ($howmany2[0] > $per_page))
{
print("\n");
print("\n");

$start = 0; 
$index = 0; 
$total = $howmany[0]; 
$remain = $per_page; 
while ($start < ($total + 1)) 
{ 
  $limitstart = 'LIMIT '.$start.','.$per_page; 
  $result = mysql_query("SELECT Remark,username 
                         FROM phones 
                         WHERE Remark LIKE '%$NAME%' ORDER BY Remark $limitstart", $mysql_conn);

  $row = mysql_fetch_row($result); 
  $from = $row[0]; 
  if (($total - $start) < $per_page) { $remain = $total - $start; } 
  for ($i = 1; $i < $remain; ++$i) { $row = mysql_fetch_row($result); } 
  $to = $row[0]; 

  print("<SoftKeyItem>\n"); 
  print("\t<Name>"); 
  print($index); 
  print("</Name>\n"); 
  print("\t<URL>"); 
  print($URL."search.php?FROM=".$from."&TO=".$to); 
  print("</URL>\n"); 
  print("</SoftKeyItem>\n");

  $start = $start + $per_page; 
  $index = $index+1;


}

$start2 = 0; 
$index2 = 0; 
$total2 = $howmany2[0]; 
$remain2 = $per_page; 
while ($start2 < ($total2 + 1)) 
{ 
  $limitstart2 = 'LIMIT '.$start2.','.$per_page2; 
  $result2 = mysql_query("SELECT name,extension 
                         FROM users 
                         WHERE name LIKE '%$NAME%' ORDER BY name $limitstart2", $mysql_conn2);

  $row2 = mysql_fetch_row($result2); 
  $from2 = $row2[0]; 
  if (($total2 - $start2) < $per_page) { $remain2 = $total2 - $start2; } 
  for ($i2 = 1; $i2 < $remain2; ++$i2) { $row2 = mysql_fetch_row($result2); } 
  $to2 = $row2[0]; 

  print("<SoftKeyItem>\n"); 
  print("\t<Name>"); 
  print($index2); 
  print("</Name>\n"); 
  print("\t<URL>"); 
  print($URL."search.php?FROM=".$from2."&TO=".$to2); 
  print("</URL>\n"); 
  print("</SoftKeyItem>\n");

  $start2 = $start2 + $per_page2; 
  $index2 = $index2+1;


}
print("</CiscoIPPhoneDirectory>\n");

} else {
print("\n");
print("\n");
$result = mysql_query("SELECT Remark,username,username
FROM phones
WHERE phones.Remark LIKE '%$NAME%'
ORDER BY Remark ", $mysql_conn);

  
while($row = mysql_fetch_row($result)) 
{ 
  print("<DirectoryEntry>\n"); 
  print("\t<Name>"); 
  print($row[0]."- ".$row[1] );

  print("</Name>\n"); 
  print("\t<Telephone>"); 
  print($row[2]); 
  print("</Telephone>\n"); 
  print("</DirectoryEntry>\n"); 
}

$result2 = mysql_query("SELECT name,extension,extension 
                       FROM users 
                       WHERE users.name LIKE '%$NAME%' 
                       ORDER BY name ", $mysql_conn2);

  
while($row2 = mysql_fetch_row($result2)) 
{ 
  print("<DirectoryEntry>\n"); 
  print("\t<Name>"); 
  print($row2[0]."- ".$row2[1] );

  print("</Name>\n"); 
  print("\t<Telephone>"); 
  print($row2[2]); 
  print("</Telephone>\n"); 
  print("</DirectoryEntry>\n"); 
}	
print("</CiscoIPPhoneDirectory>\n");

}

} else {
print("\n");
print("\n");
print("Intuittech Directory\n");
print("Intuittech Directory\n");

$result = mysql_query("SELECT Remark,username,username 
                     FROM phones 
                     WHERE Remark>='$FROM' AND Remark<='$TO' 
                     ORDER BY Remark", $mysql_conn);

while($row = mysql_fetch_row($result))
{
print("\n");
print("\t");
print($row[0]."- ".$row[1] );
print("\n");
print("\t");
print($row[2]);
print("\n");
print("\n");
}

$result2 = mysql_query("SELECT name,extension,extension 
                     FROM users 
                     WHERE name>='$FROM' AND name<='$TO' 
                     ORDER BY name", $mysql_conn2);

while($row2 = mysql_fetch_row($result2))
{
print("\n");
print("\t");
print($row2[0]."- ".$row2[1] );
print("\n");
print("\t");
print($row2[2]);
print("\n");
print("\n");
}

print("\n");
}

?>

I then do a factory reset of a cisco 7941, hoping when it comes back up, pulls its config from the tftp server the directory button works, but i don't get anything the directory buttons takes you to the directory menu but i only have options for missed/received and placed.

im obviously doing something wrong but with all the documentation im reading i cant figure out for the life of me what.
hopefully some braniac here will be able to help.

Sorry for the long post i just wanted to make it as detailed as possible.

Regards
Cap

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@markcrobinson wrote:

So my trunk provider, Vitelity, detects unusual Intl call volume and shuts off International. Phew. Their call logs show about $50 worth of international calls from my IP 12/14/16 18:18.
I've checked my asterisk logs for the time and CDR and found no reference to that number.
What else can I check?
I was hoping to find the extension they went through or some clue.
Is there a "Tighten FreePBX Security Guide?"

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@mvogel4949 wrote:

If I run a complete update all the way to the newest version of FreePBX13, which I believe is -17, will that also update all of my modules to the newest release version? And if I happen to be at 13-1 and update all of my modules to the newest release version is that essentiall putting me at -17? I'm just trying to fully understand the difference between the two. Thanks!

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@astbox wrote:

So I have an old Epygi gateway, with one pri and one fxs port. The fxs port registers to my FreePBX box.
For the pri port I have a sip trunk.

I get a digest authentication error, the device send as username the one that I have filled in my sip trunk but asterisk compares it with the one that the extensions has. Something like this

WARNING...: chan_sip.c:16702 check_auth: username mismatch, have <937>, digest has <10000>

I setup my trunk with host as dynamic and I register one more ip account from the gateway to use this and avoid ip authentication but whatever I try asterisk just goes and compares the digest with the last one that have read.

The Epygi gateway cannot listen to different port for each voip account so ip and port authentication is out of the way, but why asterisk keeps reading wrong the invite packet?
Is there anythin else to try. My sip trunk options are the following

host=dynamic
username=10000
secret=10000
canreinvite=no
insecure=no
qualify=yes
disallow=all
allow=alaw&ulaw
nat=no
type=friend
context=from-pstn

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@FryDaddy6 wrote:

We are running FreePBX 13.0.190.7 on Asterisk 13.12.2. We have just installed 6 EXP1240H and 4 EXP1240B from Uniden. The are running awesome with great coverage and voice clarity in a manufacturing facility. However several of the phones lose registration in FreePBX, but still show registered in the Uniden Web GUI. These phones can make calls (anonymously, I am guessing) but cannot receive calls when the lose registration. Rebooting the phones several times fixed it for about 20 -30 minutes then they lose registration again. It may be they lose it when the roam between uniden base stations, but I don't know for sure. This only seems to be a problem with 3 specific phones and extensions. We are using a dedicated extension for each phone. The DECT tree is configured properly in the Uniden GUI. Anyone have any ideas?

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@vianneyjs wrote:

Hi guys,

I registered the mac address of my remote Sangoma S500 @Sangoma Portal, at this point the phone pulls its config from its deployment, but it does not pull its VPN tar file. Redirection type being used is HTTPS/FQDN along with the port number (83)

I created a template on the EPM for remote Sangoma phone and set the Provision Server Protocol to HTTP.

If I manually upload the VPN tar file to the phone, after restarting it, it will register to the PBX with its corresponding settings defined on EPM.

The phone is upgraded to the latest firmware available: 2.0.4.21
This FreePBX deployment is up-to-date, including the commercial modules: System Admin and EPM.

Any idea why this remote phone is not pulling its VPN tar file?

Thank you.

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@DarkQuark wrote:

Greetings all. This might be a misconfig on my end, if it is please point it out.

I have 3 extensions set to record (via force option on the extensions) inbound internal and inbound external calls. The idea is to capture customer conversations but they transfer calls around (hence the internal recording). Call flow is inbound call -> ring group -> someone answers and then transfers to the right person.

They complain they never see the customer recording. What it appears is going on is that they are doing a supervised transfer to the final destination of the call telling them what's up before the call gets transferred. I have recordings from the initial call and the supervised part of the transfer. But when they call is sent to the final destination there is no recording.

Thanks!

Official Distro
FreePBX v13
Asterisk v13

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@nubleet wrote:

I have a trunk that is setup specifically to forward outside calls to a certain internal extension back out a SIP trunk to another external number. Is there a way to append CID information to these calls, so the number I am forwarding these calls to can see that they are coming from the internal extension?

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@sorvani wrote:

My certificate failed to renew the last couple days, so I checked the modules and updated certificate manage form 13.0.34 to 13.0.34.4. No change.

There was an error updating the certificate: Verification ended with error: {"identifier":{"type":"dns","value":"fpbx.domain.com"},"status":"invalid","expires":"2016-12-23T16:43:50Z","challenges":[{"type":"dns-01","status":"pending","uri":"https:\/\/acme-v01.api.letsencrypt.org\/acme\/challenge\/<snippysnip>\/NNNNNNNN","token":"<snip>"},{"type":"tls-sni-01","status":"invalid","error":{"type":"urn:acme:error:unauthorized","detail":"Incorrect validation certificate for TLS-SNI-01 challenge. Requested <snip>.acme.invalid from XXX.XXX.XXX.XXX:443. Received certificate containing 'fpbx.domain.com'","status":403},"uri":"https:\/\/acme-v01.api.letsencrypt.org\/acme\/challenge\/<snippysnip>\/NNNNNNNN","token":"<snip>","keyAuthorization":"<snip>","validationRecord":[{"hostname":"fpbx.domain.com","port":"443","addressesResolved":["XXX.XXX.XXX.XXX"],"addressUsed":"XXX.XXX.XXX.XXX"}]},{"type":"http-01","status":"pending","uri":"https:\/\/acme-v01.api.letsencrypt.org\/acme\/challenge\/<snippysnip>\/NNNNNNNN","token":"<snip>"}],"combinations":[[0],[2],[1]]}

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@nettel wrote:

Hello, have a few workstations for our customer service department where the users change shift throughout the day. The department leader would like to begin getting stats/reports on each individual user. Is there a method to do this? We are open to commercial/paid modules that help us accomplish this along with reporting. Thank you!

AsteriskNow
Cent6.5
Asterisk 13.2.0
Freepbx 12.0.76.2

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@dcitelecom wrote:

I use SRV sub domain record to point UDP and TCP traffic to port 5060. e.g.
sub1.domain.com = UDP port 5060
sub2.domain.com = TCP port 5060

Do I need to create another record for TLS and point it to port 5061? e.g.
sub3.domain.com = TLS port 5061

TLS seems to work with sub2.domain.com pointing at port 5060 but maybe it would work better with sub3.domain.com pointing at port 5061

If hope someone on this board is familiar with SRV records. Thanks.

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@dears1208 wrote:

I am working with a client that would like calls to ring, if unanswered they would need to hear a message, then go to MOH while holding for an agent.
Caller side. "4 Rings -> "Thank you for calling, we are busy but stay on the line or press 1 at any time to leave a message" -> MOH until answered.
Agent side. Ring until answered.

I have tried setting up 2 queues (601,602) 601 has ringing routes through a queue priority to increase the priority of the call as it enters the 602 queue, 602 has an then MOH. This is 90% what I was looking for, the issue is the call stops ringing at the agents phones between the two queues.
I am also not sure if the call is in the 602 queue, will the agent get all of the 602 calls before the 601 calls.
Is there a way to do this that I am not thinking of.
The only other way that I can think of is to record what I would like as MOH and just have the MOH play the Ringing and the message.
Any help would be great.

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