Nisu mi bas najjasniiji konfiguracioni fajlovi kod Elastixa. Naime imamo tri vrste sip.conf fajla: sip.conf, sip_custom.conf i sip_additional.conf.
Ovako izgleda
sip.conf na elastixu
Code:
;--------------------------------------------------------------------------------;
; 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: http://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 (C) 2004 Coalescent Systems Inc (Canada)
; Copyright (C) 2006 Why Pay More 4 Less Pty Ltd (Australia)
; Copyright (C) 2007 Astrogen LLC (USA)
[general]
; 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.
;
#include sip_custom_post.conf
A ovako s
ip_additional.conf, sa popisom svih registrovanih ekstenzija i trunkova koje smo preko gui definisali
Code:
;--------------------------------------------------------------------------------;
; 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: http://freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;
[100]
deny=0.0.0.0/0.0.0.0
secret=100voip
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/100
accountcode=
mailbox=100@device
permit=0.0.0.0/0.0.0.0
callerid=device <100>
call-limit=50
faxdetect=no
.
.
.................
sip_custom.conf je prazan! Dakle sve modifikacije koje je potrebno uraditi , a ticu se sip.conf fajla radimo u sip_custom.conf? Isto vazi i za extension.conf!?
Postoji jos i sip.conf.old_freePBX....sadrzaj mu je defaultni kao kod samog asteriska.
@JOJO847
Provjereno justvoip daje 2 kanala...mozda i vise-to ne znam.!?
Hvala na preporuci za Nomado, ali daleko je skuplji u odnosu na betamaxove klonove. Recimo poziv za Germany preko Nomada je 0.19eurcenti+vat, a preko justvoipa 0.08eurcenti +vat. Pa uporedite.!
Kako da konektujem Asterisk na VoIP provajdera ?