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Discussion: asterisk: configuration pour pré décroché automatique.

  1. #1
    Membre Junior
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    septembre 2016
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    asterisk: configuration pour pré décroché automatique.

    Bonjour à tous et toutes,

    Je suis sur une distribution Mageia 5 avec un bureau KDE.

    Je viens d'installer via les dépôts asterisk (11.23.1) et les softphones ekiga et jitsi. Les installations se sont bien passées je pense puisque les softphones se lancent correctement: je me vois connecté avec ma ligne sip.
    Le but d'une telle installation est de me fournir un standard pour recevoir des appels, lesquels auront un pré décroché d'attente.

    En console, j'ai cette info lorsque je tape asterisk -r et je reload ensuite :

    localhost*CLI> reload
    [Nov 1 15:08:30] WARNING[711]: cel.c:361 do_reload: Could not load cel.conf
    [Nov 1 15:08:30] NOTICE[711]: app_queue.c:7953 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action.
    [Nov 1 15:08:30] NOTICE[711]: pbx_ael.c:164 pbx_load_module: Starting AEL load process.
    [Nov 1 15:08:30] NOTICE[711]: pbx_ael.c:177 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'.
    [Nov 1 15:08:30] NOTICE[711]: pbx_ael.c:180 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'.
    [Nov 1 15:08:30] NOTICE[711]: pbx_ael.c:187 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'.
    [Nov 1 15:08:30] NOTICE[711]: pbx_ael.c:192 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'.
    [Nov 1 15:08:30] NOTICE[711]: pbx_ael.c:195 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'.
    localhost*CLI>
    Disconnected from Asterisk server
    Asterisk cleanly ending (0).
    Executing last minute cleanups
    [root@localhost mand]#
    En console, j'ai après avoir taper "sip show users" :
    localhost*CLI> sip show user
    Usage: sip show user <name> [load]
    Shows all details on one SIP user and the current status.
    Option "load" forces lookup of peer in realtime storage.
    localhost*CLI> sip show users
    Username Secret Accountcode Def.Context ACL Forcerport
    101 azerty local No Yes
    102 azerty local No Yes
    6000 1234 work No No
    localhost*CLI>
    Comme j'ai essentiellement bidouillé dans les fichiers extensions.conf et sip.conf, je ne comprends pas le problème évoquant le fichier extensions.ael
    Je mets ici les deux fichiers :

    extensions.conf (un peu long, désolé):
    [general]
    static=yes
    writeprotect=no
    autofallthrough=yes
    clearglobalvars=no

    [globals]
    CONSOLE=Console/dsp ; Console interface for demo
    ;CONSOLE=DAHDI/1
    ;CONSOLE=Phone/phone0
    IAXINFO=guest ; IAXtel username/password
    ;IAXINFO=myuser:mypass
    TRUNK=DAHDI/G2 ; Trunk interface

    TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
    ; X - any digit from 0-9
    ; Z - any digit from 1-9
    ; N - any digit from 2-9
    ; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)

    ;[context]
    ;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
    ; <time range>,<days of week>,<days of month>,<months>[,<timezone>]
    ;include => daytime,9:00-17:00,mon-fri,*,*
    ;include => weekend,*,sat-sun,*,*
    ;include => weeknights,17:02-8:58,mon-fri,*,*
    ;ignorepat => 9

    [dundi-e164-canonical]
    ;exten => 12564286000,1,Gosub(6000,stdexten(IAX2/foo))
    ;exten => 12564286000,n,Goto(default,s,1) ; exited Voicemail
    ;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})

    [dundi-e164-customers]
    ;exten => _12564286000,1,Dial(SIP/customer1)
    ;exten => _12564286001,1,Dial(IAX2/customer2)

    [dundi-e164-via-pstn]
    ;exten => _1256428XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428
    ;exten => _1256325XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Ditto for 256-325

    [dundi-e164-local]
    include => dundi-e164-canonical
    include => dundi-e164-customers
    include => dundi-e164-via-pstn

    [dundi-e164-switch]

    switch => DUNDi/e164

    [dundi-e164-lookup]

    include => dundi-e164-local
    include => dundi-e164-switch

    [macro-dundi-e164]

    exten => s,1,Goto(${ARG1},1)
    include => dundi-e164-lookup

    [iaxtel700]
    exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)

    [iaxprovider]
    ;switch => IAX2/user:[key]@myserver/mycontext

    [trunkint]
    exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
    exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${FILTER(0-9,${EXTEN:${GLOBAL(TRUNKMSD)}})})

    [trunkld]
    exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
    exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

    [trunklocal]
    exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

    [trunktollfree]
    exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
    exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
    exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
    exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

    [international]
    ignorepat => 9
    include => longdistance
    include => trunkint

    [longdistance]
    ignorepat => 9
    include => local
    include => trunkld

    [local]
    ignorepat => 9
    include => default
    include => trunklocal
    include => iaxtel700
    include => trunktollfree
    include => iaxprovider
    include => parkedcalls
    ; switch => IAX2/user:password@bigserver/local
    ; lswitch => Loopback/12${EXTEN}@othercontext
    ; eswitch => IAX2/context@${CURSERVER}
    ; This is the dialing hook. use:
    ; include => outbound-freenum

    [outbound-freenum]
    exten => _X*X!,1,Goto(outbound-freenum2,${EXTEN},1)
    exten => _XX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
    exten => _XXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
    exten => _XXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
    exten => _XXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
    exten => _XXXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)

    [outbound-freenum2]
    exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
    same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)}) ; make sure the suffix is all digits as well
    same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1)
    ; filter out bad characters per the README-SERIOUSLY.best-practices.txt document
    same => n,Set(TIMEOUT(absolute)=10800)
    same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freen um.org)}) ; perform our lookup with freenum.org
    same => n,GotoIf($["${isnresult}" != ""]?from)
    same => n,Set(DIALSTATUS=CONGESTION)
    same => n,Goto(fn-CONGESTION,1)
    same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
    same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial) ; check if we set the FREENUMDOMAIN global variable in [global]
    same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ; if we did set it, then we'll use it for our outbound dialing domain
    same => n(dial),Dial(SIP/${isnresult},40)
    same => n,Goto(fn-${DIALSTATUS},1)

    exten => fn-BUSY,1,Busy()

    exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
    same => n,Congestion()

    [macro-trunkdial]
    ; ${ARG1} - What to dial
    exten => s,1,Dial(${ARG1})
    exten => s,n,Goto(s-${DIALSTATUS},1)
    exten => s-NOANSWER,1,Hangup
    exten => s-BUSY,1,Hangup
    exten => _s-.,1,NoOp

    [stdexten]
    ; ${EXTEN} - Extension
    ; ${ARG1} - Device(s) to ring
    ; ${ARG2} - Optional context in Voicemail
    -------->suite du fichier extensions.conf dans le post suivant car trop long.


    et sip.conf :


    [general]
    context=local ;Contexte par defaut
    bindport=5060 ;UDP standard
    bindaddr=0.0.0.0 ;bind access to all
    srvlookup=yes ;activer les lookup DNS des appels
    language=fr ;MSG vocaux en FR

    [101] ;Login SIP
    secret=azerty ;Mot de passe
    callerid= »Bob_les_ponges» <101> ;Affichage lors de l appel
    context=local ;appels geres dans extension local
    mailbox=101@default ;compte de msg vocale cfr voicemail.conf
    type=friend ;allow in et out
    host=dynamic ;adresse ip du client
    nat=yes ;utiliser derriere du NAT

    [102]
    secret=azerty
    callerid= »Kiki » <102>
    context=local
    type=friend
    host=dynamic
    nat=yes
    mailbox=102@default
    Je suis connecté avec une Bbox. J'ai une ligne sip. Ekiga. J'utilise un casque avec micro pour les appels.
    En test, si je m'appelle, ça sonne côté appelant et j'ai le fichier de pré décroché qui se lance côté réception d'appel, soit exactement l'inverse de ce que je souhaite !

    Je pense que le problème n'est pas gravissisme mais vu mon niveau général.. je tourne en rond. Il y a forcément un truc que je n'ai pas indiqué...et même plusieurs.

    Merci aux intervenants qui me fileront un coup de main pour comprendre le problème.
    Dernière modification par Hub49 ; 01/11/2016 à 20h19.

  2. #2
    Membre Junior
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    septembre 2016
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    Suite du fichier extensions.conf :
    exten => _X.,50000(stdexten),NoOp(Start stdexten)
    exten => _X.,n,Set(LOCAL(ext)=${EXTEN})
    exten => _X.,n,Set(LOCAL(dev)=${ARG1})
    exten => _X.,n,Set(LOCAL(cntx)=${ARG2})
    exten => _X.,n,Set(LOCAL(mbx)=${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})})
    exten => _X.,n,Dial(${dev},20) ; Ring the interface, 20 seconds maximum
    exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
    exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
    exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
    exten => stdexten-BUSY,1,Voicemail(${mbx},b) ; If busy, send to voicemail w/ busy announce
    exten => stdexten-BUSY,n,Return() ; If they press #, return to start
    exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer
    exten => a,1,VoicemailMain(${mbx}) ; If they press *, send the user into VoicemailMain
    exten => a,n,Return()

    [stdPrivacyexten]
    ; Standard extension subroutine:
    ; ${ARG1} - Extension
    ; ${ARG2} - Device(s) to ring
    ; ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority)
    ; ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)`
    ; ${ARG5} - Context in voicemail (if empty, then "default")
    exten => _X.,60000(stdPrivacyexten),NoOp(Start stdPrivacyexten)
    exten => _X.,n,Set(LOCAL(ext)=${ARG1})
    exten => _X.,n,Set(LOCAL(dev)=${ARG2})
    exten => _X.,n,Set(LOCAL(dontcntx)=${ARG3})
    exten => _X.,n,Set(LOCAL(tortcntx)=${ARG4})
    exten => _X.,n,Set(LOCAL(cntx)=${ARG5})

    exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
    exten => _X.,n,Dial(${dev},20,p) ; Ring the interface, 20 seconds maximum, call screening
    ; option (or use P for databased call _X.creening)
    exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
    exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
    exten => stdexten-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
    exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
    exten => stdexten-BUSY,1,Voicemail(${mbx},b) ; If busy, send to voicemail w/ busy announce
    exten => stdexten-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
    exten => stdexten-BUSY,n,Return() ; If they press #, return to start
    exten => stdexten-DONTCALL,1,Goto(${dontcntx},s,1) ; Callee chose to send this call to a polite "Don't call again" script.
    exten => stdexten-TORTURE,1,Goto(${tortcntx},s,1) ; Callee chose to send this call to a telemarketer torture script.
    exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer
    exten => a,1,VoicemailMain(${mbx}) ; If they press *, send the user into VoicemailMain
    exten => a,n,Return

    [macro-page];
    ; ${ARG1} - Device to page
    exten => s,1,ChanIsAvail(${ARG1},s) ; s is for ANY call
    exten => s,n,GoToIf($[${AVAILSTATUS} = "1"]?autoanswer:fail)
    exten => s,n(autoanswer),Set(_ALERT_INFO="RA") ; This is for the PolyComs
    exten => s,n,SIPAddHeader(Call-Info: Answer-After=0) ; This is for the Grandstream, Snoms, and Others
    exten => s,n,NoOp() ; Add others here and Post on the Wiki!!!!
    exten => s,n,Dial(${ARG1})
    exten => s,n(fail),Hangup

    [demo]
    include => stdexten
    exten => s,1,Wait(1) ; Wait a second, just for fun
    exten => s,n,Answer ; Answer the line
    exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
    exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
    exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message
    exten => s,n(instruct),BackGround(demo-instruct) ; Play some instructions
    exten => s,n,WaitExten ; Wait for an extension to be dialed.
    exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
    exten => 2,n,Goto(s,instruct)
    exten => 3,1,Set(CHANNEL(language)=fr) ; Set language to french
    exten => 3,n,Goto(s,restart) ; Start with the congratulations
    exten => 1000,1,Goto(default,s,1)
    exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
    ; (but skip if channel is not up)
    exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}) )
    exten => 1234,n,Goto(default,s,1) ; exited Voicemail
    exten => 1235,1,Voicemail(1234,u) ; Right to voicemail
    exten => 1236,1,Dial(Console/dsp) ; Ring forever
    exten => 1236,n,Voicemail(1234,b) ; Unless busy
    exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
    exten => #,n,Hangup ; Hang them up.
    exten => t,1,Goto(#,1) ; If they take too long, give up
    exten => i,1,Playback(invalid) ; "That's not valid, try again"
    exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
    exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default) ; Call the Asterisk demo
    exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site
    exten => 500,n,Goto(s,6) ; Return to the start over message.
    exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
    exten => 600,n,Echo ; Do the echo test
    exten => 600,n,Playback(demo-echodone) ; Let them know it's over
    exten => 600,n,Goto(s,6) ; Start over
    exten => 76245,1,Macro(page,SIP/Grandstream1)
    exten => _7XXX,1,Macro(page,SIP/${EXTEN})
    exten => 7999,1,Set(TIMEOUT(absolute)=60)
    exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d)
    exten => 8500,1,VoicemailMain
    exten => 8500,n,Goto(s,6)
    ;exten => 1265,1,Dial(Phone/phone0,15)
    ;exten => 1265,n,Goto(s,5)

    [page]
    exten => _X.,1,Macro(page,SIP/${EXTEN})
    ;[mainmenu]
    ;exten => s,1,Answer
    ;exten => s,n,Background(thanks) ; "Thanks for calling press 1 for sales, 2 for support, ..."
    ;exten => s,n,WaitExten
    ;exten => 1,1,Goto(submenu,s,1)
    ;exten => 2,1,Hangup
    ;include => default
    ;[submenu]
    ;exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback
    ;exten => s,n,Wait,2
    ;exten => s,n,Background(submenuopts) ; "Thanks for calling the sales department. Press 1 for steve, 2 for..."
    ;exten => s,n,WaitExten
    ;exten => 1,1,Goto(default,steve,1)
    ;exten => 2,1,Goto(default,mark,2)

    [public]
    ; ATTENTION: If your Asterisk is connected to the internet and you do
    ; not have allowguest=no in sip.conf, everybody out there may use your
    ; public context without authentication. In that case you want to
    ; double check which services you offer to the world.
    ;
    include => demo
    [default]
    [local]
    exten => _1XX, 1, Dial(SIP/${EXTEN}, 15) ; Compose 101 appelle franky etc
    exten => _1XX, n, VoiceMail(${EXTEN}) ; Voicemail apres 15 secondes
    exten => 90,1,VoiceMailMain(${CALLERID(num)}) ; Messagerie
    exten => 300, 1, Meetme(300)

    include => demo
    ;exten => _41X.,1,Dial(SIP/${FILTER(0-9,${EXTEN:2})}@sipprovider,,r)
    ;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1(Joe Schmoe) ; Channel hints for presence
    ;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
    ;exten => 6245,n(dial),Dial(${HINT},20,rtT) ; Use hint as listed
    ;exten => 6245,n,Voicemail(6245,u) ; Voicemail (unavailable)
    ;exten => 6245,s+1,Hangup ; s+1, same as n
    ;exten => 6245,dial+101,Voicemail(6245,b) ; Voicemail (busy)
    ;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit
    ;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
    ;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
    ;exten => 6391,1,Dial(JINGLE/asterisk@digium.com/mogorman@astjab.org) ;Dial via jingle using asterisk as the transport and calling mogorman.
    ;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}

    ;exten => 6275,1,Gosub(${EXTEN},stdexten(${MARK}))
    ; assuming ${MARK} is something like DAHDI/2
    ;exten => 6275,n,Goto(default,s,1) ; exited Voicemail
    ;exten => mark,1,Goto(6275,1) ; alias mark to 6275
    ;exten => 6536,1,Gosub(${EXTEN},stdexten(${WIL}))
    ; Ditto for wil
    ;exten => 6536,n,Goto(default,s,1) ; exited Voicemail
    ;exten => wil,1,Goto(6236,1)
    ;exten => 6600,hint,park:701@parkedcalls
    ;exten => 6600,1,noop
    ; You can also monitor the status of a queue by providing a hint for a
    ; particular queue name.
    ;exten => 8502,hint,Queue:markq
    ;exten => 8502,1,Queue(markq)

    ;To subscribe to the availability of a free member in the 'markq' queue.
    ;Note: '_avail' is added to the QueueName
    ;exten => 8501,hint,Queue:markq_avail
    ;exten => 8501,1,Queue(markq)
    ; Some other handy things are an extension for checking voicemail via
    ; voicemailmain
    ;exten => 8500,1,VoicemailMain
    ;exten => 8500,n,Hangup
    ;exten => 8600,1,Meetme(1234)
    ;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
    ;[acme-incoming]
    ;exten => s,1,Wait(1)
    ;exten => s,n,Answer()
    ;exten => s,n(menu),Playback(acme/vm-brief-menu)
    ;exten => s,n(exten),Background(vm-enter-num-to-call)
    ;exten => s,n,WaitExten(5)
    ;exten => s,n(goodbye),Playback(vm-goodbye)
    ;exten => s,n(end),Hangup()
    ;include => acme-extens
    ;exten => i,1,Playback(vm-invalid)
    ;exten => i,n,Goto(s,exten) ; optionally, transfer to operator
    ;exten => t,1,Goto(s,goodbye)
    ; this is the context our internal SIP hardphones use (see sip.conf)
    ;[acme-internal]
    ;exten => s,1,Answer()
    ;exten => s,n(exten),Background(vm-enter-num-to-call)
    ;exten => s,n,WaitExten(5)
    ;exten => s,n(goodbye),Playback(vm-goodbye)
    ;exten => s,n(end),Hangup()
    ;include => trunkint
    ;include => trunkld
    ;include => trunklocal
    ;include => acme-extens
    ;exten => 777,1,DISA(no-password,acme-incoming)
    ;[acme-extens]
    ;include => stdexten
    ;exten => 111,1,Gosub(111,stdexten(SIP/pete_1,acme))
    ;exten => 111,n,Goto(s,exten)
    ;exten => 112,1,Gosub(112,stdexten(SIP/nancy_1,acme))
    ;exten => 112,n,Goto(s,end)
    Pour le prédécroché apparemment :
    pour le prédécroché c'est l'option "m" de la commande "dial": exten => 1,1,Dial(SIP/toto,,m(musique))
    Dernière modification par Hub49 ; 01/11/2016 à 20h32.

  3. #3
    Membre Senior
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    septembre 2010
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    0
    - le fichier extensions.ael a été installé avec les paquets, dans l'absolu, tu peux soit le vider, soit l'effacer - ca contient du code qui si utilisé, apporte des fonctionnalités d'un pabx

    - attention, softphone + asterisk sur le meme serveur = conflit pour utiliser le port 5060.... il faut t'assurer que le sofptohne n'essaie pas d'utiliser le port 5060 comme port source

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