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Vega 60G Bri ports

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@tcmtelecom_stu wrote:

Hi
I have a Vega 60G with 4 Bri interfaces - it is linking a cloud instance of freepbx to a Panasonic KXTDE 100 via 4 Bri ports on the Panasonic. Periodically (sometimes daily) one or two of the Bri ports becomes disconnected. A reboot of the Vega gateway does not cure this, but if i log in to the vega with the web gui, go into Bri port setup, untick the NT box, save and save - then go back in, tick the NT box, save and save they come up. BTW as of Sunday i am on the latest firmware but this was happening long before the upgrade.
Any help would be appreciated as i am having to log in daily to check if all ports are up.
Thank you.

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iSymphony call button issue

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@Raven5650 wrote:

hello guys,
i have installed the last stable version of freepbx and activated a trial of isymphony (as base internal module of freepbx) with sipml5 as softphone for isymphony.
i have configured webrtc pjsip extension(let’s say 101,102,103) and they work fine with sipml5 alone. I can do outbound and inbound calls. Instead, let’s say i log in with 101 in isymphony client and sipml5, if i use the “call” button of isymphony, sipml5 receive a call from the extension 101 itself whatever number i digit (other extension or external number).
Asterisk write the following log:

[2020-04-23 15:52:49] VERBOSE[30880] res_rtp_asterisk.c: DTLS ECDH initialized (automatic), faster PFS enabled
[2020-04-23 15:52:49] VERBOSE[30880] netsock2.c: Using SIP RTP TOS bits 184
[2020-04-23 15:52:49] VERBOSE[30880] netsock2.c: Using SIP RTP CoS mark 5
[2020-04-23 15:52:49] VERBOSE[30882] dial.c: Called 101
[2020-04-23 15:52:50] VERBOSE[30882] dial.c: SIP/101-00000049 is ringing

So i assume that isymphony is dialing 101
Is that a known issue or am i doing something wrong?
Thanks for help.

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FreePBX repository's down?

Yum: repos unavailable this AM

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@bitbanger wrote:

Another Sangoma repo access problem, this time with yum.
Got an email from my PBX at 08:19 EDT today:

This is an automatic notification from your FreePBX (VoIP Server) server. Could not retrieve mirrorlist http://mirrorlist.sangoma.net/?release=7&arch=x86_64&repo=os&dist=$dist&staging=$staging error was 12: Timeout on http://mirrorlist.sangoma.net/?release=7&arch=x86_64&repo=os&dist=$dist&staging=$staging: (28, ‘Connection timed out after 30001 milliseconds’) One of the configured repositories failed (Unknown), and yum doesn’t have enough cached data to continue.

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Digium phones looking for the Wrong PBX Server IP

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@froglevelmc wrote:

I recently bought a Sangoma FreePBX Appliance 40. I am new to this whole thing and not terrible familiar with phone systems short of online videos. Anyway I have set this thing up and and am having trouble getting some Digium D40 and D70 phones to connect. DHCP gave the device an IP of 192.168.100.139 and then changed it to a static IP of 192.168.100.7. The first phone I plugged into the network started looking for 192.168.100.139 (the wrong address).
I can manually set the IP in the phone and it finds all the available extensions. When I select one it looks like it is reconfiguring the phone and restarts. As soon as it restarted is starts looking for the wrong IP again. I have factory defaulted the thing and tried it again but it still does the same thing. Where is the phone getting that old IP from?

Ultimately, my intent is to be able to provision the phones with the endpoint manager. but when I try adding a phone with the extension mapping and save / reconfig / update nothing happens. I am sure there is a config mistake somewhere.

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Cant download distro

Isymphony not working

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@Scion wrote:

FreePBX Distro 1910-2
Version: FreePBX 15.0.16.49
Asterisk: 16.9.0

When going to admin - Isymphony - Status says:

Failed to contact the iSymphony server.
Verify that your iSymphony server is installed and running and that the server API host and port are correct in the fields below.
If you have SSL enabled below and are using the SSL port for the API connection you need to enable SSL in the iSymphony server’s security.xml file for the communication_manager servlet

Cli command to restart:

[root@freepbx ~]# service isymphony restart
Redirecting to /bin/systemctl restart isymphony.service
Failed to restart isymphony.service: Unit not found.

Debug data:

/

var

/

www

/

html

/

admin

/

modules

/

cxpanel

/

lib

/

util.php

  1. $char = substr($possible, mt_rand(0, $maxlength-1), 1);
  • if (!strstr($password, $char)) {
  1. $password .= $char;
  2. $i++;
  3. }
  4. }
  • return $password;
  1. }
  • /**
    • Reads the contents of a file and returns it as a string
    • @param String $file path to file that needs to be read
  1. */
  2. function cxpanel_read_file($file) {
  3. $contents = “”;
  4. if(($contentFile = fopen($file, ‘r’)) !== false) {
  5. while (!feof($contentFile)) {
  6. $contents .= fgets($contentFile, 4096);
  7. }
  8. fclose($contentFile);
  9. }
  10. return $contents;
  11. }
  • /**
    • Converts a 2 dimensional array into a table
    • @param Array $array the array to convert
    • @param String $tagAdditions content to add to the <table> tag
  1. */
  2. function cxpanel_array_to_table_2d($array, $tagAdditions = “”) {
  3. $table = “<table $tagAdditions>”;
  4. $header = true;
  5. foreach($array as $a) {
  6. if($header) {

Arguments

  1. “fopen(/var/spool/asterisk/cxpanel/modify.log): failed to open stream: No such file or directory”

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A200 Card Isn't Working

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@MC1 wrote:

Hello

I’ve had this issue in the past and was only able to rectify it by paying for tech support from Sangoma. I’m hoping to resolve this with the aid of this forum.

Issue: the FXO & FXS ports aren’t being recognized by the system.

I’ve run trough the command line configuration as well as the DAHDI GUI with no improvement. I’ve disabled and re-enabled the DAHDI module.

Any ideas as to how I can get the card to work? Thanks in advance for any help that can be provided.

Michael

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Sangoma Appliance 40 new interface not responding

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@froglevelmc wrote:

I just purchased a Sangoma Appliance 40. The local LAN is 192.168.1.0/24 the phones are on VLAN 192.168.100.0/24. On eth0 I set a static IP of 192.168.1.7 and I set eth1 to static 192.168.100.3. The eth1 interface will not respond to the phones or ping or anything. I whitelisted the both subnets in Intrusion Prevention and added the networks to the firewall for NAT. Any ideas why eth1 is not talking? It does not even show up on a LAN scan.

[root@freepbx ~]# clear
[root@freepbx ~]# cat /etc/sysconfig/network-scripts/ifcfg-eth1
DEVICE=eth1
BOOTPROTO=static
ONBOOT=‘yes’
IPADDR=192.168.100.3
NETMASK=255.255.255.0
GATEWAY=192.168.100.1
ZONE=internal
PEERDNS=no
IPV6_PEERDNS=no
DESCRIPTION=“unset”

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Call recording

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@JDEC01 wrote:

Hi,

I saw similar posts but no answer.

*1 works for call recording but the “record” button on D50 do not.
DPMA record option is set to “enabled” but still no record is done when pressing it.

I am under Freepbx 14 and Asterix 14.

What am I missing ?

Many thanks.

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PBXact New System 502 bad gateway error

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@sprater wrote:

After setting up a new PBXact system, running system updates, and setting up the basic zulu server, just configuring the advanced options based on the wiki article. After that and restarting I am getting a 502 bad gateway error after logging in. The system can reach the internet and I can manipulate it by SSH but I can’t seem to find anything regarding this issue or solutions. Any thoughts would be greatly appreciated.

/var
/www
/html
/admin
/modules
/soundlang
/Soundlang.class.php

        }
        if(!empty($exceptions)) {
            $message = '';
            $code = '';
            foreach($exceptions as $e) {
                if($this->FreePBX->Config->get('MODULE_REPO') !== $this->FreePBX->Config->get_conf_default_setting('MODULE_REPO')) {
                    $this->FreePBX->Config->reset_conf_settings(array('MODULE_REPO'),true);
                    $code = 500;
                    $message = _("The mirror server did not return the correct response and had been previously changed from the default server(s), it has now been reset back to the standard default. Please try again");
                } else {
                    $code = $e->getCode();
                    $msg = $e->getMessage();
                    $message .= !empty($msg) ? $msg : sprintf(_("Error %s returned from remote servers %s"),$code,json_encode($mirrors['mirrors']));
                    $message .= ", ";
                }
 
            }
            $message = rtrim(trim($message),",");
 
            throw new \Exception($message,$code);
        } else {
            throw new \Exception(sprintf(_("Unknown Error. Response was empty from %s"),json_encode($mirrors['mirrors'])),0);
        }
    }
    public function getRightNav($request) {
        return load_view(dirname(__FILE__).'/views/rnav.php',array());
    }
}

Arguments

"""
<html><body><h1>502 Bad Gateway</h1>\n
The server returned an invalid or incomplete response.\n
</body></html>\n
"""

Environment & details:
GET Data empty
POST Data empty
Files empty
Cookies
lang

“en_US”

PHPSESSID

“on6cf6k1g0cpemdds9vpmt8837”

Session
langdirection

“ltr”

module_name

“”

module_page

“”

AMP_user

ampuser {}

Server/Request Data
UNIQUE_ID

“Xqsm85GNnQMKDCCk2NAyggAAAAU”

HTACCESS

“on”

HTTPS

“on”

SSL_SERVER_S_DN_C

“–”

SSL_SERVER_S_DN_ST

“SomeState”

SSL_SERVER_S_DN_L

“SomeCity”

SSL_SERVER_S_DN_O

“SomeOrganization”

SSL_SERVER_S_DN_OU

“SomeOrganizationalUnit”

SSL_SERVER_S_DN_CN

“localhost.localdomain”

SSL_SERVER_S_DN_Email

“root@localhost.localdomain”

SSL_SERVER_I_DN_C

“–”

SSL_SERVER_I_DN_ST

“SomeState”

SSL_SERVER_I_DN_L

“SomeCity”

SSL_SERVER_I_DN_O

“SomeOrganization”

SSL_SERVER_I_DN_OU

“SomeOrganizationalUnit”

SSL_SERVER_I_DN_CN

“localhost.localdomain”

SSL_SERVER_I_DN_Email

“root@localhost.localdomain”

SSL_VERSION_INTERFACE

“mod_ssl/2.4.6”

SSL_VERSION_LIBRARY

“OpenSSL/1.0.2k-fips”

SSL_PROTOCOL

“TLSv1.2”

SSL_SECURE_RENEG

“true”

SSL_COMPRESS_METHOD

“NULL”

SSL_CIPHER

“ECDHE-RSA-AES128-GCM-SHA256”

SSL_CIPHER_EXPORT

“false”

SSL_CIPHER_USEKEYSIZE

“128”

SSL_CIPHER_ALGKEYSIZE

“128”

SSL_CLIENT_VERIFY

“NONE”

SSL_SERVER_M_VERSION

“3”

SSL_SERVER_M_SERIAL

“7708”

SSL_SERVER_V_START

“Apr 17 14:11:29 2020 GMT”

SSL_SERVER_V_END

“Apr 17 14:11:29 2021 GMT”

SSL_SERVER_S_DN

“emailAddress=root@localhost.localdomain,CN=localhost.localdomain,OU=SomeOrganizationalUnit,O=SomeOrganization,L=SomeCity,ST=SomeState,C=–”

SSL_SERVER_I_DN

“emailAddress=root@localhost.localdomain,CN=localhost.localdomain,OU=SomeOrganizationalUnit,O=SomeOrganization,L=SomeCity,ST=SomeState,C=–”

SSL_SERVER_A_KEY

“rsaEncryption”

SSL_SERVER_A_SIG

“sha256WithRSAEncryption”

SSL_SESSION_ID

“49a778b12e782ac21a5cb2830a2642f1b53184ccff895f2300df30cc1ec68f0e”

SSL_SESSION_RESUMED

“Resumed”

HTTP_HOST

“10.200.4.10”

HTTP_USER_AGENT

“Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:75.0) Gecko/20100101 Firefox/75.0”

HTTP_ACCEPT

“text/html,application/xhtml+xml,application/xml;q=0.9,image/webp,/;q=0.8”

HTTP_ACCEPT_LANGUAGE

“en-US,en;q=0.5”

HTTP_ACCEPT_ENCODING

“gzip, deflate, br”

HTTP_DNT

“1”

HTTP_CONNECTION

“keep-alive”

HTTP_COOKIE

“lang=en_US; PHPSESSID=on6cf6k1g0cpemdds9vpmt8837”

HTTP_UPGRADE_INSECURE_REQUESTS

“1”

PATH

“/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin”

SERVER_SIGNATURE

“”

SERVER_SOFTWARE

“Apache/2.4.6 (Sangoma) OpenSSL/1.0.2k-fips PHP/5.6.40”

SERVER_NAME

“10.200.4.10”

SERVER_ADDR

“10.200.4.10”

SERVER_PORT

“443”

REMOTE_ADDR

“10.200.1.3”

DOCUMENT_ROOT

“/var/www/html”

REQUEST_SCHEME

“https”

CONTEXT_PREFIX

“”

CONTEXT_DOCUMENT_ROOT

“/var/www/html”

SERVER_ADMIN

“root@sangoma.localhost”

SCRIPT_FILENAME

“/var/www/html/admin/config.php”

REMOTE_PORT

“18087”

GATEWAY_INTERFACE

“CGI/1.1”

SERVER_PROTOCOL

“HTTP/1.1”

REQUEST_METHOD

“GET”

QUERY_STRING

“”

REQUEST_URI

“/admin/config.php”

SCRIPT_NAME

“/admin/config.php”

PHP_SELF

“/admin/config.php”

REQUEST_TIME_FLOAT

1588274931.774

REQUEST_TIME

1588274931

Environment Variables empty
Registered Handlers
0. Whoops\Handler\PrettyPageHandler

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iSymphony queue paused agent visibility

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@Raven5650 wrote:

Hi
in the agent widget did anyone know if there is a way to see if the other agent in same queue are in pause?
It seem to me that i can only see the logged in but it would be useful see even the paused ones

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Accessing the Console Port from a Lap Top LAN Port

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@MC1 wrote:

Hello

I don’t have a USB to Serial Console cable and wanted to know if there is a way to access the Console port on my FreePBX 60 via the LAN port on my Lap Top? Is there a piece of software that I can download to make this happen?

Thanks in advance for any help that can be provided.
Michael

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Unable to purchase Commercial Module

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@kaiserpr wrote:

We want to add a commercial module (Queue PRO) to our PBXact appliance. The deployment id match the sangoma portal and if I run fwconsole command. The checkout process cant find my deployment id no matter what I put on the field. This appliance was purchase from VOIPSUPPLY. Can we purchase module directly from sangoma?

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Vega 60 to Free PBX No CID

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@Croenbaugh wrote:

I have a configuration problem with my Vega 60g to freepbx.

When the Vega gets an incoming call from the PSTN the caller id that shows up is just the trunk name “Vega” in this case. The inbound route directs to a ring group. The phones ring once then stop for two seconds then ring again.

I’m lost.

Incoming SIP Trunk Settings

username=vega
type=peer
trustrpid=yes
sendrpid=yes
secret=XXXXXXXX
qualify=yes
insecure=very
host=176.16.0.69
dtmfmode=rfc2833
disallow=all
context=from-trunk
allow=ulaw

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A200 Line Cord Extension

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@Altocloud1 wrote:

The wiki shows that the card uses an RJ11 Narrow to RJ14. I need to either extend these current cables or replace. I don’t deal with analog connections often can someone point me in the right direction?

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Activation

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@freebs wrote:

Looking to get reactivated… I have moved the server once again… and no Zend resets left.
ID is 44838064

Thanks!

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Please help me configure TE133 Single-Span T1/E1 Card use FreePBX

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@truongtv wrote:

Hi guys,
I have tried the configuration a lot for TE133 Single-Span cards on FreePBX, but it still doesn’t work properly, please help me configure it.

I have setup a ISDN card and use FreePBX 15.0.16.49.

I use 2 TE133 cards on 2 FreePBX, temporarily called FreePBX A and FreeBPX B.

The call from FreePBX A comes in FreePBX B, the phone rang, and when I answered, the call only lasted for 31 seconds, then it automatically hangup. I have noticed its giving > Span 1: Channel 0/1 got hangup request, cause 16
Below is the log part of FreePbx B:
–PJSIP/100-00000001 answered DAHDI/i1/101-2

– Channel PJSIP/100-00000001 joined ‘simple_bridge’ basic-bridge <64fc141e-1016-4fd7-9111-46f619672291>
– Channel DAHDI/i1/101-2 joined ‘simple_bridge’ basic-bridge <64fc141e-1016-4fd7-9111-46f619672291>
– Span 1: Channel 0/1 got hangup request, cause 16
– Channel DAHDI/i1/101-2 left ‘simple_bridge’ basic-bridge <64fc141e-1016-4fd7-9111-46f619672291>
== Spawn extension (macro-dial-one, s, 56) exited non-zero on ‘DAHDI/i1/101-2’ in macro ‘dial-one’
== Spawn extension (macro-exten-vm, s, 26) exited non-zero on ‘DAHDI/i1/101-2’ in macro ‘exten-vm’
== Spawn extension (ext-local, 100, 3) exited non-zero on ‘DAHDI/i1/101-2’
– Executing [h@ext-local:1] Macro(“DAHDI/i1/101-2”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“DAHDI/i1/101-2”, “1?theend”) in new stack
– Channel PJSIP/100-00000001 left ‘simple_bridge’ basic-bridge <64fc141e-1016-4fd7-9111-46f619672291>
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“DAHDI/i1/101-2”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“DAHDI/i1/101-2”, "PJSIP/100-00000001 montior file= ") in new stack
– Executing [s@macro-hangupcall:5] GotoIf(“DAHDI/i1/101-2”, “1?skipagi”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [s@macro-hangupcall:7] Hangup(“DAHDI/i1/101-2”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘DAHDI/i1/101-2’ in macro ‘hangupcall’
== Spawn extension (ext-local, h, 1) exited non-zero on ‘DAHDI/i1/101-2’
– Hungup ‘DAHDI/i1/101-2’

On both FreePBX A and B, I created and used:

  1. Extension: PJSIP
  2. Inbound
  3. Outbound
  4. DAHDi Trunk!

This is configuration information for the TE133 card on FreePBX B. The configuration on FreePBX A is the same as on FreePBX B. They differ only from Signaling

Note: Signaling on FreePBX A is PRI - CPE , and Signaling on FreePBX B is PRI - NET. . If I use the same Signaling, the call won’t work between FreePBX A and FreePBX B, I don’t know why either

Please help me to see where I am getting the wrong configuration.

Thank you so much.

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Is isymphony dead?

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@joethegeek wrote:

I have logged a support ticket with them and tried to contact them via email, but no response.
Is isymphony still “alive”

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Yealink T42g forwarding to cell phone even though call forward isn't enabled

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@jkalber wrote:

So I have a rather strange one here. I have a user that submitted a ticket stating his desk phone no longer rings and only his cell phone rings. We are running FreePBX version 2.9.0.15.

I have confirmed that call forward is disabled on his phone from the display menu settings. I have also confirmed that DND is not enabled. I have connected to the Yealink phone portal for this specific phone and have confirmed under Features > Forward&DND that “Always Forward” and “DND Status” are both set to off.

I do have find me follow me enabled for his phone and his find me follow me settings are listed below;

  • Initial Ring Time: 10

  • Ring Strategy: ringallv2 (so that it rings his primary extension first)

  • Ring Time: 20

  • Follow-Me List: his primary extension is first and just beneath that is his cell phone number followed with a # sign.

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