merci a tous j'ai pu faire la redirection le client xlite se connecte mais la voix ne passe pas pendant un appel.pourtant j'ai redirigé les ports 5060 sip et 10000 à 15000 rtp vers l'adresse du pbx.
voici quelques fichiers de ma config
rtp.conf
Code:
;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
rtpstart=10000
rtpend=15000
sip_additional.conf
Code:
[204]
deny=0.0.0.0/0.0.0.0
type=friend
secret=
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=204@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/204
context=from-internal
canreinvite=no
callgroup=
callerid=device <204>
accountcode=
call-limit=50
[200]
deny=0.0.0.0/0.0.0.0
type=friend
secret=
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=200@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/200
context=from-internal
canreinvite=no
callgroup=
callerid=device <200>
accountcode=
call-limit=50
sip_nat.conf
Code:
nat=yes
externhost=82.*.*.*
localnet=192.168.100.0/255.255.255.0
sip.conf
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 ;
;--------------------------------------------------------------------------------;
;
[general]
bindport=5060
bindaddr=0.0.0.0
; 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
Merci pour votre aide.
Envoyé par davlefou
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