FreePBX Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)

service asterisk stop chown -R asterisk:asterisk /var/run/asterisk



FreePBX Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)

Exception: Unable to connect to Asterisk through the CLI in file /var/lib/asterisk/bin/retrieve_conf on line 24

Error Upgrading Modules: GPG Verify File check failed

   mkdir /home/asterisk

    chown -R asterisk  /home/asterisk
 
  sudo -u asterisk gpg --keyserver pgp.mit.edu --recv-key 3DDB2122FE6D84F7


http://community.freepbx.org/t/error-upgrading-modules-gpg-verify-file-check-failed/35340/5

http://community.freepbx.org/t/error-upgrading-modules-gpg-verify-file-check-failed/35340

Autodestruct on dialog

han_sip.c: Autodestruct on dialog '7c1ea63f33a10e3128bbab276168c768@94.188.133.70' with owner SIP/omnisip-outgoing-0000001a in place (Method: BYE). Rescheduling destruction for 10000 ms

Then you most likely have a 'stuck' channel in Asterisk. That typically occurs when the channel should be destroyed, but something is still referencing the channel. This is usually sign of a bug in some off nominal condition.

As it is, Asterisk 11.9.x is pretty old - over two years now - and bugs involving stuck channels have certainly been fixed within that time frame. I'd recommend upgrading to the latest in the 11.x series to see if that resolves the issue.

https://community.asterisk.org/t/calls-not-disconnected-over-ami/67606

Asterisk AMI DTMF

<?php
error_reporting (E_ALL);
set_time_limit(60);
ob_implicit_flush(false);

$ip_asterisk = "127.0.0.1";
$channel = $_GET[c];
$dtmf = $_GET[d];

$oSocket = fsockopen($ip_asterisk, 5038, $errnum, $errdesc) or
die("Connection to host failed");
    fputs($oSocket, "Action: login\r\n");
    fputs($oSocket, "Username: admin\r\n");
    fputs($oSocket, "Secret: 1456\r\n\r\n");
    fputs($oSocket, "Action: PlayDTMF\r\n");
    fputs($oSocket, "Channel: $channel\r\n");
    fputs($oSocket, "Digit: $dtmf\r\n\r\n");
/*
   usleep(500000);
    fputs($oSocket, "Action: PlayDTMF\r\n");
    fputs($oSocket, "Channel: $canal\r\n");
    fputs($oSocket, "Digit: 8\r\n\r\n");
    usleep(500000);
    fputs($oSocket, "Action: Logoff\r\n\r\n");
*/
    // Carga toda la respuesta recibida en un string
    $loaded = "";
    while (!feof($oSocket)){
        $buffer = fgets($oSocket, 4096);
        $loaded .= $buffer;
    }

    $vec = explode("<br>", $loaded);
    $len = count($vec);
    print_r($vec);
?>
http://165.181.11.23/dtmf.php?c=SIP/400-000000df&d=1


http://lists.digium.com/pipermail/asterisk-users/2009-October/238570.html

freepbx 13 centos 6

[root@FreePBX-13 freepbx]# ./install -n --dbpass 123456


http://wiki.freepbx.org/pages/viewpage.action?pageId=1048598

Feature Code Call Transfers

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• Creado por Rusty Newton, modificado por última vez en jun 29, 2015

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Overview of Feature Code Call Transfers

A call transfer is when one party of a call directs Asterisk to connect the other party to a new location on the system.

Transfer types supported by the Asterisk core:

• Blind transfer
• Attended transfer
  • Variations on attended transfer behavior

Transfer features provided by the Asterisk core are configured in features.conf and accessed with feature codes.

Channel driver technologies such as chan_sip and chan_pjsip have native capability for various transfer types. That native transfer functionality is independent of this core transfer functionality. The core feature code transfer functionality is channel agnostic.

how to provisioning polycom

Getting Started

This is a publicly accessible server intended only for use of provisioning, upgrading, or downgrading the software of Polycom IP phones with stock Polycom software and configuration files.

Multiple 'stock' directories exist with different versions of SIP, UC, and BootROM software, and are named according to the SIP or UC software version contained within. These directories are completely stock; no changes should be made to any configuration files within.

Each directory contains an ABOUT.TXT to indicate which SIP and BootROM software versions exist within that directory.

To Provision Your Polycom VoIP Phone:

1.       From the phone's local menu interface, press Menu > Settings > Advanced (default password: 456) > Administration Settings > Network Configuration > Server Menu.

2.       Configure your DHCP menu:

On 3.x Phones: Main Menu > Settings > Advanced > Admin Settings > Network Configuration > DHCP Menu
On 4.x/5.x phones: Main Menu > Settings > Advanced > Admin Settings > Network Configuration > Provisioning Server > DHCP Menu set the following:

Boot Server: Static

BootSrv Type: IP Address������ (Once phone is upgraded, please remove the IP address from the phone!!!)

3.       From the Provisioning Server menu set:�� Server Type: HTTP

4.      Next input Server Address as voipt2.polycom.com or IP�� Example to load the latest SIP 4.06 = voipt2.polycom.com/406

a.       Note voipt2.polycom.com can be entered in IP form, enter:� Example 140.242.64.35/406 for convenience.