Hi,
I am running Asterisk 11.7 and trying to setup direct media between the end points (Analog phones).
I tried directmedia = yes but it doesn’t seem to work. Please help
sudo cat /etc/asterisk/sip.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;
; This file is part of FreePBX.
;
; FreePBX is free software: you can redistribute it and/or modify
; it under the terms of the GNU General Public License as published by
; the Free Software Foundation, either version 2 of the License, or
; (at your option) any later version.
;
; FreePBX is distributed in the hope that it will be useful,
; but WITHOUT ANY WARRANTY; without even the implied warranty of
; MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
; GNU General Public License for more details.
;
; You should have received a copy of the GNU General Public License
; along with FreePBX. If not, see http://www.gnu.org/licenses/.
;
; Copyright © 2004 Coalescent Systems Inc (Canada)
; Copyright © 2006 Why Pay More 4 Less Pty Ltd (Australia)
; Copyright © 2007 Astrogen LLC (USA)
[general]
directmedia=yes
; These files will all be included in the [general] context
;
#include sip_general_additional.conf
;sip_general_custom.conf is the proper file location for placing any sip general
;options that you might need set. For example: enable and force the sip jitterbuffer.
;If these settings are desired they should be set the sip_general_custom.conf file.
;
; jbenable=yes
; jbforce=yes
;
;It is also the proper place to add the lines needed for sip nat’ing when going
;through a firewall. For nat’ing you’d need to add the following lines:
; nat=yes , externip= , localhost= , and optionally fromdomain= .
;
#include sip_general_custom.conf
;sip_nat.conf is here for legacy support reasons and for those that upgrade
;from previous versions. If you have this file with lines in it please make
;sure they are not duplicated in sip_general_custom.conf, if so remove them
;from sip_nat.conf as sip_general_custom.conf will have precedence.
#include sip_nat.conf
;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
;
#include sip_registrations_custom.conf
#include sip_registrations.conf
; These files should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf
;sip_custom_post.conf If you have extra parameters that are needed for a
;extension to work to for example, those go here. So you have extension
;1000 defined in your system you start by creating a line 1000 in this
;file. Then on the next line add the extra parameter that is needed.
;When the sip.conf is loaded it will append your additions to the end of
;that extension.
sudo cat /etc/asterisk/sip_custom.conf
subscribe to Message Waiting
sip explicit mwi subscription: 1
local_secure
directmedia=yes
sudo cat /etc/asterisk/sip_general_additional.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;
vmexten=5000
accept_outofcall_messages=yes
auth_message_requests=no
outofcall_message_context=dpma_message_context
disallow=all
allow=ulaw
allow=alaw
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-2.11.0(11.7.0)
directmedia=yes
sudo cat /etc/asterisk/sip_general_custom.conf
bindaddr=::
directmedia=yes
allowsubscribe=yes