Merci, voici le fichier sip.conf.
J'en ai coupé une partie car ce sont plusieurs fois la même configuration pour nos télépones/user. La configuration est plutôt primaire, j'utilise Asterisk 1.4.21 des dépôts Debian.
Je viens d'activer le Debug SIP, quelles informations sont intéressante pour mon cas ?
Code:
[general]
defaultexpirey = 1800
dtmfmode = auto
qualify = yes
context = others
port = 5060
bindaddr = 0.0.0.0
register => 095027****:****@freephonie.net
allow = all
register => 095020****:****@freephonie.net
allow = all
register => USER:****@voip.wengo.fr/user
allow = all
[freephonie-out]
type = peer
host = freephonie.net
username = 095027****
fromuser = 095027****
secret = ****
nat = yes
language = fr
callerid ="User" <014366****>
insecure = very
qualify = yes
[freephonie-out2]
type = peer
host = freephonie.net
username = 095020****
fromuser = 095020****
secret = ****
nat = yes
language = fr
callerid ="User" <014366****>
insecure = very
qualify = yes
[neuftalk-out]
type=friend
host=voip.wengo.fr
username=user
fromuser=user
secret=****
nat=yes
callerid="User" <014366****>
[neuftalk-in]
type=user
context=mainmenu
language=fr
qualify=yes
allow=all
[fp-in]
type = peer
context = mainmenu
host = freephonie.net
language=fr
[11]
type = friend
context = phones
host = dynamic
secret = ****
language=fr
callerid = "JD" <11>
display-name = "JD" <11>
... SPLIT 9 autre compte SIP similaire ...
[41]
username=41
type = friend
context = phones
host = dynamic
nat=0
dtmfmode=auto
secret = ****
language=fr
canreinvite=no
callerid = "Conference" <41>
display-name = "Conference" <41>
qualify=yes
incominglimit=1
outgoinglimit=1
call-limit=1
busylevel=1