Subitement aucune tonnalité sur les lignes

[cedricscha]

As-tu comme te la suggere jean controle l'ip de ton serveur dans la config de tes telephone?

[alexfos]

Bjr
l'@ip de mon serveur dans la config de mes pap2 ou mon softphone pangolin est bon.
et mon fournisseur n'a bloqué aucun port sur mon serveur.(ce serveur est hebergé sur ovh)
Je viens mm de modifier mon port sip, mais toujours pas de tonnalité.

Je viens mm de reinstaller asterisk avec freepbx, mais tjrs rien.
une question: est la mauvaise config de freepbx et mysql peut causer ce pb, je demande ceci parce que l'installation d'amportal me renvoie une erreur:
[ERROR] queues access failed, Queues module may not be installed: DB Error: no such table
Please update your modules and reload Asterisk by visiting http://x.x.x.x/admin

et a l'interface freepbx j'ai aussi cette erreur:
SELECT * FROM cronmanager [nativecode=1146 ** Table 'asterisk.cronmanager' doesn't exist]SQL -
SELECT * FROM cronmanager

Added 20 minutes ago
(cron_manager.FATAL)

svp je ne sais plus quoi tester!

[alexfos]

Voici mon fichier sip.conf avant la modification du port:

;--------------------------------------------------------------------------------;
; 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]

; 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

et apres avoir spécifier le port 5222:

;--------------------------------------------------------------------------------;
; 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]
port=5222
bindport=5222
; 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

je spécifie que mes extensions s'enregistre plutot dans le fichier sip_additional.conf, voici un extrait:

[3333]
type=friend
secret=3333
qualify=yes
port=5222
pickupgroup=
nat=yes
mailbox=3333@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/3333
context=from-internal
canreinvite=no
callgroup=
callerid=device <3333>
accountcode=
call-limit=50

[cedricscha]

je déplace ton post dans la catégorie distribution packagée, tu auras, peut etre plus de réponse la bas.