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
Extensions.conf (réduit à l'essentiel)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
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
et extconfig.confCode:[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
Le appels internes et externes marchent en enregistrant les extensions statiques, aucune de la base de donnée n'est chargée.Code:[settings] sippeers => odbc,general,sip_buddies extensions => odbc,general,extensions
Toute aide ou suggestion est la bienvenue.