ODBC et MySQL realtime extensions et utilisateurs

Bonjour,
Cela fait quelques jours que je tourne en rond sur ce problème et je vous remercie déja de votre aide
Voici ma configuration :Asterisk 11.13 sous Centos 6.5 avec MySQL et apache fonctionnels
J'essaie de charger les utilisateurs (et/ou extensions) depuis une base MySQL
Voici mes fichiers
Sip.conf:

Code:

---

[general]
context = DLPN_dialplan1  ; Default context for incoming calls
allowoverlap = no  ; Disable overlap dialing support. (Default is yes)
udpbindaddr = 0.0.0.0  ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
tcpenable = yes  ; Enable server for incoming TCP connections (default is no)
tcpbindaddr = 0.0.0.0  ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
transport = udp  ; Set the default transports.  The order determines the primary default transport.
srvlookup = yes  ; Enable DNS SRV lookups on outbound calls
language = fr  ; Default language setting for all users/peers
videosupport = yes  ; Turn on support for SIP video. You need to turn this
subscribecontext = default
localnet=192.168.200.0/255.255.0.0 ; RFC 1918 addresses
rtcachefriends=yes            ; Cache realtime friends by adding them to the internal list
rtupdate=yes                  ; Send registry updates to database using realtime? (yes|no)
rtautoclear=no                ; Auto-Expire friends created on the fly on the same schedule
bindport=5060
engine=asterisk 
[authentication]

[basic-options](!); a template
dtmfmode = rfc2833
context = from-office
type = friend

[natted-phone](!,basic-options); another template inheriting basic-options
directmedia = no
host = dynamic

[public-phone](!,basic-options); another template inheriting basic-options
directmedia = yes

[my-codecs](!); a template for my preferred codecs
disallow = all
allow = ilbc
allow = g729
allow = gsm
allow = g723
allow = ulaw


[ulaw-phone](!); and another one for ulaw-only
disallow = all
allow = ulaw

---

users.conf

Code:

---

[general]
dtmfmode = rfc2833
hasvoicemail = yes
hassip = yes
hasiax = yes
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
callgroup = 1
pickupgroup = 1
nat = force_rport
vmexten = 6999
bindaddr=0.0.0.0
bindport = 5060
videosupport=yes
externhost=rh.ddns.net      ; refreshed periodically
externrefresh=180              ; change the refresh interval
disallow=all
allow=gsm     
allow=alaw                     
allow=ulaw
allow=speex
allow=h264
allow=h261
allow=h263
allow=h263p
allowsubscribe=yes
asterisk sip allowoverlap=yes
caninvite=no ; These setting confirm we want the PBX handling the audio
canreinvite=no
jbenable=yes
maxcallbitrate=384
rtpcachefriends=yes
rtupdate=yes


[template](!)
type = friend
host = dynamic
dtmfmode = rfc2833
disallow = all
allow = ulaw
allow = h263

[6000](template)
fullname = Standard
username = tax
secret = 123456
context = DLPN_dialplan1
callcounter = yes
linenumber = 1
cid_number = 6000
hasvoicemail = no
vmsecret =
email =
threewaycalling = no
hasdirectory = yes
callwaiting = no
hasmanager = no
hasagent = yes
hassip = yes
hasiax = yes
nat = force_rport
canreinvite = no
insecure = no
pickupgroup =
call-limit = 100
allow = ulaw,h263
macaddress = 6000
autoprov = yes
label = 6
LINEKEYS = 1

[6001](template)
fullname = tahiana raolona
username = tax
secret = 123456
context = DLPN_dialplan1
callcounter = yes
linenumber = 1
cid_number = 6001
hasvoicemail = no
vmsecret =
email =
threewaycalling = no
hasdirectory = yes
callwaiting = no
hasmanager = no
hasagent = yes
hassip = yes
hasiax = yes
nat = force_rport
canreinvite = no
insecure = no
pickupgroup =
call-limit = 100
allow = ulaw,h263
macaddress = 6001
autoprov = yes
label = 6001
LINEKEYS = 1

[6002](template)
fullname = xperia tax
username = xperia tax
secret = 123456
context = DLPN_dialplan1
callcounter = yes
linenumber = 1
cid_number = 6002
hasvoicemail = no
vmsecret =
email =
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = no
hassip = yes
hasiax = no
nat = force_rport
canreinvite = no
insecure = no
pickupgroup =
call-limit = 100
allow = ulaw,h263
macaddress = 6002
autoprov = yes
label = 6002
LINEKEYS = 1

---

Extensions.conf (réduit à l'essentiel)

Code:

---

[CallingRule_work]
switch =>Realtime
exten => _6XXX,1,Dial(SIP/${EXTEN},20,tTkK)
exten => _6XXX,n,Playback(MSG-${EXTEN})
exten => _6XXX,n,VoiceMail(${EXTEN}@CallingRule_work)
exten => 6999,1,VoiceMailMain(${CALLERID(num)}@CallingRule_work,s)
exten => 9001,1,Answer()
exten => 9001,2,Set(TIMEOUT(response)=10)
exten => 9001,3,agi(googletts.agi,"Bienvenues chez Techmedia!",fr,any)
exten => 9001,4,agi(googletts.agi,"Qui souhaitez vous joindre?",fr,any)
exten => 9001,5,agi(googletts.agi,"Pour Tax Lenovo tapez 1",fr,any)
exten => 9001,6,agi(googletts.agi,"Pour  Tax Xperia tapez 2",fr,any)
exten => 9001,7,agi(googletts.agi,"Appuyez sur dièse si vous souhaitez réécouter ce  message",fr,any)
exten => 9001,8,WaitExten()
exten => 1,1,Goto(6001,1)
exten => 2,1,Goto(6006,1)
exten => _[3-9#],1,Goto(8001,3)
exten => t,1,Goto(8001,3)
exten => 9000,1,Goto(voicemail-msg,s,1)




[voicemail-msg]
exten => s,1,Answer
exten => s,2,agi(googletts.agi,"Bienvenue dans l'utilitaire de création de messages d'accueil.",fr,any)
exten => s,3,agi(googletts.agi,"Après le bip sonore, veillez annoncer votre message d'accueil, et validez avec dièse.",fr,any)
exten => s,4,Record(MSG-${CALLERID(num)}:ulaw)
exten => s,5,agi(googletts.agi,"Voici votre message d'accueil: ",fr,any)
exten => s,6,Playback(MSG-${CALLERID(num)})
exten => s,7,agi(googletts.agi,"Si vous souhaitez le ré enregistrer appuyez sur 1",fr,any)
exten => s,8,agi(googletts.agi,"Si vous souhaitez garder ce message vous pouvez raccrocher",fr,any)
exten => s,9,Set(TIMEOUT(response)=10)
exten => s,10,WaitExten()
exten => 1,1,Goto(voicemail-msg,s,3)
exten => _[2-9#],1,Goto(voicemail-msg,s,7)
exten => t,1,Goto(voicemail-msg,s,7)



[dongle-incoming]
exten => sms,1,Verbose(Incoming SMS from ${CALLERID(num)} ${BASE64_DECODE(${SMS_BASE64})})
exten => sms,n,System(echo '${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} - ${DONGLENAME} - ${CALLERID(num)} - ${BASE64_DECODE(${SMS_BASE64})}' >> /var/log/asterisk/sms.txt) ; be careful this may not be safe if sms/ussd contains shell code
exten => sms,n,System(php /var/www/html/sms.php ${DONGLENAME} ${CALLERID(num)} '${BASE64_DECODE(${SMS_BASE64})}')
exten => sms,n,Hangup()
exten => ussd,1,Verbose(Incoming USSD: ${BASE64_DECODE(${USSD_BASE64})})
exten => ussd,n,System(echo '${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} - ${DONGLENAME}: ${BASE64_DECODE(${USSD_BASE64})}' >> /var/log/asterisk/ussd.txt) ; be careful this may not be safe if sms/ussd contains shell code
exten => ussd,n,Hangup()
exten => s,1,NoOp(Appel entrant de: ${CALLERID(all)} to ${EXTEN})
        same => n,Dial(SIP/6004,20,tTkK)
        same => n,Playback(MSG-${EXTEN})
        same => n,VoiceMail(${EXTEN}@CallingRule_work)
        same => n,Hangup()



[dongle-outgoing]
exten => _0XXXXXXXXX,1,Dial(SIP/6000,20,tTkK)
exten => _+261XXXXXXXXX,1,Dial(SIP/6000,20,tTkK)
exten => _990XXXXXXXXX,1,Dial(Dongle/appel_externe/${EXTEN:2})



[incoming]
exten => _2XXX,1,Dial(SIP/6001&SIP/6006, 20)  ;Action lors d'un appel, dans ce cas appeler les postes: 6001, 6002, 6003 et 6004 mm tps



[DLPN_dialplan1]
include = CallingRule_work
include = dongle-incoming
include = dongle-outgoing
include = default

---

L'odbc fontionne correctement parce que j'arrive à enregistrer sans problèmes les CDRs
j'ai suivi ce guide http://www.open-voip.org/index.php?t...altime_example

qui n'es plus à jour apparemment puisque ce'est l'odbc qui est conseillé.
J'ai donc essayé de l'adapter à l'ODBC

voici le fichier res_odbc.conf

Code:

---


[ENV]

[asterisk]
enabled => yes
dsn => asterisk
username => root
password => *****
pre-connect => yes
sanitysql => select 1
idlecheck => 3600
share_connections => yes
backslash_is_escape => no


[mysql]
enabled => yes
dsn => asterisk
username => root
password => *****
pre-connect => yes
[mysql2]
enabled => no
dsn => MySQL-asterisk
username => myuser
password => mypass
pre-connect => yes

[sqlserver]
enabled => no
dsn => mickeysoft
share_connections => no
limit => 5
username => oscar
password => thegrouch
pre-connect => yes
sanitysql => select count(*) from systables
                             
backslash_is_escape => no

---

et extconfig.conf

Code:

---

[settings]
 sippeers => odbc,general,sip_buddies
 extensions => odbc,general,extensions

---

Le appels internes et externes marchent en enregistrant les extensions statiques, aucune de la base de donnée n'est chargée.

Toute aide ou suggestion est la bienvenue.