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Discussion: Perte de l'audio [OVH]

  1. #1
    Membre Junior
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    mai 2015
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    Perte de l'audio [OVH]

    Bonjour à tous,

    je viens vers vous car mon serveur asterisk a subitement perdu tout le son qu'il était capable de délivré ...

    Aujourd'hui que ce soit en appel externe, interne, avec ivr, voicemail, etc ... je n'ai plus de son ou alors 5 secondes par appel à n'importe quelle moment.

    Voici le CLI des appels entrants et sortants :

    Code:
    == Using SIP RTP CoS mark 5
        -- Executing [s@depuis-ovh:1] Answer("SIP/vers-ovh-0000001b", "") in new stack
        -- Executing [s@depuis-ovh:2] Set("SIP/vers-ovh-0000001b", "CALLERIN=01XXXX7888") in new stack
        -- Executing [s@depuis-ovh:3] GotoIf("SIP/vers-ovh-0000001b", "1?4:8") in new stack
        -- Goto (depuis-ovh,s,4)
        -- Executing [s@depuis-ovh:4] Set("SIP/vers-ovh-0000001b", "HEURE=OUVERT") in new stack
        -- Executing [s@depuis-ovh:5] GotoIf("SIP/vers-ovh-0000001b", "1?6:7") in new stack
        -- Goto (depuis-ovh,s,6)
        -- Executing [s@depuis-ovh:6] Goto("SIP/vers-ovh-0000001b", "ivr-01XXXX7888,s,1") in new stack
        -- Goto (ivr-01XXXX7888,s,1)
        -- Executing [s@ivr-01XXXX7888:1] Answer("SIP/vers-ovh-0000001b", "") in new stack
        -- Executing [s@ivr-01XXXX7888:2] AGI("SIP/vers-ovh-0000001b", "googletts.agi, "Bienvenue chez IXXXXX !",fr,any") in new stack
        -- Launched AGI Script /var/lib/asterisk/agi-bin/googletts.agi
        -- <SIP/vers-ovh-0000001b>AGI Script googletts.agi completed, returning 0
        -- Executing [s@ivr-01XXXX7888:3] AGI("SIP/vers-ovh-0000001b", "googletts.agi, "Pour joindre le service apr�s vente, tapez 1", fr, any") in new stack
        -- Launched AGI Script /var/lib/asterisk/agi-bin/googletts.agi
        -- <SIP/vers-ovh-0000001b>AGI Script googletts.agi completed, returning 0
        -- Executing [s@ivr-01XXXX7888:4] AGI("SIP/vers-ovh-0000001b", "googletts.agi, "Pour joindre le service commercial, tapez 2", fr, any") in new stack
        -- Launched AGI Script /var/lib/asterisk/agi-bin/googletts.agi
        -- <SIP/vers-ovh-0000001b>AGI Script googletts.agi completed, returning 0
        -- Executing [s@ivr-01XXXX7888:5] AGI("SIP/vers-ovh-0000001b", "googletts.agi, "Pour joindre le service informatique, tapez 3", fr, any") in new stack
        -- Launched AGI Script /var/lib/asterisk/agi-bin/googletts.agi
        -- <SIP/vers-ovh-0000001b>AGI Script googletts.agi completed, returning 0
        -- Executing [s@ivr-01XXXX7888:6] AGI("SIP/vers-ovh-0000001b", "googletts.agi, "Pour toute autre demande, tapez 4", fr, any") in new stack
        -- Launched AGI Script /var/lib/asterisk/agi-bin/googletts.agi
        -- <SIP/vers-ovh-0000001b>AGI Script googletts.agi completed, returning 0
        -- Executing [s@ivr-01XXXX7888:7] WaitExten("SIP/vers-ovh-0000001b", "") in new stack
        -- Timeout on SIP/vers-ovh-0000001b, going to 't'
        -- Executing [t@ivr-01XXXX7888:1] Goto("SIP/vers-ovh-0000001b", "ivr-01XXXX7888,s,2") in new stack
        -- Goto (ivr-01XXXX7888,s,2)
    Malgré que ce soit un IVR, il n'y a aucun sons.

    Code:
    == Using SIP RTP CoS mark 5
        -- Executing [06XXXX1740@work:1] Answer("SIP/6003-00000020", "") in new stack
           > 0x7f29bd1c8890 -- Probation passed - setting RTP source address to 192.168.1.66:3000
        -- Executing [06XXXX1740@work:2] Wait("SIP/6003-00000020", "1") in new stack
        -- Executing [06XXXX1740@work:3] Dial("SIP/6003-00000020", "SIP/vers-ovh/06XXXX1740") in new stack
      == Using SIP RTP CoS mark 5
        -- Called SIP/vers-ovh/06XXXX1740
        -- SIP/vers-ovh-00000021 is ringing
        -- SIP/vers-ovh-00000021 is making progress passing it to SIP/6003-00000020
        -- SIP/vers-ovh-00000021 is ringing
        -- SIP/vers-ovh-00000021 is making progress passing it to SIP/6003-00000020
        -- SIP/vers-ovh-00000021 answered SIP/6003-00000020
        -- Channel SIP/6003-00000020 joined 'simple_bridge' basic-bridge <e47e0cc0-cee6-4ff6-a9b7-117addebce72>
        -- Channel SIP/vers-ovh-00000021 joined 'simple_bridge' basic-bridge <e47e0cc0-cee6-4ff6-a9b7-117addebce72>
           > Bridge e47e0cc0-cee6-4ff6-a9b7-117addebce72: switching from simple_bridge technology to native_rtp
           > Locally RTP bridged 'SIP/vers-ovh-00000021' and 'SIP/6003-00000020' in stack
           > Locally RTP bridged 'SIP/vers-ovh-00000021' and 'SIP/6003-00000020' in stack
           > 0x2960750 -- Probation passed - setting RTP source address to 91.121.129.154:35316
        -- Channel SIP/vers-ovh-00000021 left 'native_rtp' basic-bridge <e47e0cc0-cee6-4ff6-a9b7-117addebce72>
        -- Channel SIP/6003-00000020 left 'native_rtp' basic-bridge <e47e0cc0-cee6-4ff6-a9b7-117addebce72>
      == Spawn extension (work, 06XXXX1740, 3) exited non-zero on 'SIP/6003-00000020'
    C'est simple durant cet appel sortant, je n'ai réussi a communiquer qu'au moment où cette ligne est apparu : > 0x2960750 -- Probation passed - setting RTP source address to 91.121.129.154:35316
    et ce durant 5 secondes.

    Je ne comprend pas le téléphone fonctionnait parfaitement hier.

    Voici tout de même mon sip.conf :

    Code:
    [general]
    language=fr
    bindport=5060
    bindaddr=0.0.0.0
    srvlookup=yes
    canreinvite=no
    defaultexpiry=3600
    registertimeout=30
    registerattempts=0
    disallow=all
    allow=ulaw
    allowguest=yes
    nat=yes
    
    register => 00331XXXX7888:XXXXXXXX@sip.ovh.fr
    register => 00331XXXX7980:XXXXXXXX@sip.ovh.fr
    register => 00331XXXX7985:XXXXXXXX@sip.ovh.fr
    register => 00331XXXX7908:XXXXXXXX@sip.ovh.fr
    register => 00331XXXX7885:XXXXXXXX@sip.ovh.fr
    
    ;Cr�ation du compte Asterisk pour OVH
    [vers-ovh]
    disallow=all
    type=friend
    secret=XXXXXXXX
    host=sip.ovh.fr
    fromdomain=sip.ovh.fr
    fromuser=00331XXXX7888
    username=00331XXXX7888
    nat=yes
    context=depuis-ovh
    insecure=invite,port
    qualify=yes
    canreinvite=no
    allow=ulaw

  2. #2
    Membre Junior
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    mai 2015
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    Voici tout de même mon extensions.conf :

    Code:
    [general]
    static=yes
    writeprotect=no
    clearglobalvars=no
    [globals]
    CONSOLE=Console/dsp                             ; Console interface for demo
    IAXINFO=guest                                   ; IAXtel username/password
    TRUNK=DAHDI/G2                                  ; Trunk interface
    TRUNKMSD=1                                      ; MSD digits to strip (usually 1 or 0)
    
    [work]
    include => parkedcalls
    
    exten => 01XXXX7980,1,Ringing(1)
    exten => 01XXXX7980,2,Set(CALLERIN=${CALLERID(num)})
    exten => 01XXXX7980,3,Answer
    exten => 01XXXX7980,4,GotoIf($[${DB_EXISTS(6001/NUMCF)} & ${DB(6001/NUMCF)} != 6001]?5:8)
    
    exten => 01XXXX7980,5,Set(NUMCF=${DB(6001/NUMCF)})
    exten => 01XXXX7980,6,Set(CALLERID(num)=${CALLERIN})
    exten => 01XXXX7980,7,Transfer(SIP/${NUMCF}@work)
    
    exten => 01XXXX7980,8,Goto(work,6001,1)
    exten => 01XXXX7980,9,Hangup(16)
    
    exten => 01XXXX7985,1,Ringing(1)
    exten => 01XXXX7985,2,Answer
    exten => 01XXXX7985,3,GotoIf($[${DB_EXISTS(6002/NUMCF)} & ${DB(6002/NUMCF)} != 6002]?4:6)
    
    exten => 01XXXX7985,4,Set(NUMCF=${DB(6002/NUMCF)})
    exten => 01XXXX7985,5,Transfer(SIP/${NUMCF}@work)
    
    exten => 01XXXX7985,6,Goto(work,6002,1)
    exten => 01XXXX7985,7,Hangup(16)
    
    exten => 01XXXX7908,1,Ringing(1)
    exten => 01XXXX7908,2,Answer
    exten => 01XXXX7908,3,GotoIf($[${DB_EXISTS(6003/NUMCF)} & ${DB(6003/NUMCF)} != 6003]?4:6)
    
    exten => 01XXXX7908,4,Set(NUMCF=${DB(6003/NUMCF)})
    exten => 01XXXX7908,5,Transfer(SIP/${NUMCF}@work)
    
    exten => 01XXXX7908,6,Goto(work,6003,1)
    exten => 01XXXX7908,7,Hangup(16)
    
    exten => 01XXXX7885,1,Ringing(1)
    exten => 01XXXX7885,2,Answer
    exten => 01XXXX7885,3,GotoIf($[${DB_EXISTS(6004/NUMCF)} & ${DB(6004/NUMCF)} != 6004]?4:6)
    
    exten => 01XXXX7885,4,Set(NUMCF=${DB(6004/NUMCF)})
    exten => 01XXXX7885,5,Transfer(SIP/${NUMCF}@work)
    
    exten => 01XXXX7885,6,Goto(work,6004,1)
    exten => 01XXXX7885,7,Hangup(16)
    
    exten => _6XXX,1,Dial(SIP/${EXTEN},20,tT)
    exten => _6XXX,2,VoiceMail(${EXTEN}@work)
    
    ;Num�ro de la boite vocale
    exten => 600,1,VoiceMailMain(@work)
    
    ;Num�ro pour le transfert on
    exten => *55,1,Goto(ivr-transfer-on,s,1)
    
    ;Num�ro pour le transfert off
    exten => #55,1,Goto(ivr-transfer-off,s,1)
    
    exten => _0[12345679]XXXXXXXX,1,Answer()
    exten => _0[12345679]XXXXXXXX,2,Wait(1)
    exten => _0[12345679]XXXXXXXX,3,Dial(SIP/vers-ovh/${EXTEN})
    exten => _0[12345679]XXXXXXXX,4,Hangup()
    
    ;Les appels entrants font sonner le 6001 (John DOE) et si pas r�ponses au bout de 20 secondes transfert sur sa boite vocale.
    [depuis-ovh]
    exten => s,1,Answer
    
    exten => s,2,Set(CALLERIN=${CUT(CUT(SIP_HEADER(To),@,1),:,2)})
    
    exten => s,3,GotoIf($[${CALLERIN}=01XXXX7888]?4:8)
    
    exten => s,4,Set(HEURE=${IFTIME(07:00-19:00,mon-fri,*,*?OUVERT:FERME)})
    exten => s,5,GotoIf($[${HEURE}=OUVERT]?6:7)
    exten => s,6,Goto(ivr-01XXXX7888,s,1)
    exten => s,7,VoiceMail(8000@work)
    
    exten => s,8,Goto(work,${CALLERIN},1)
    
    ;IVR pour la gestion des transfert d'appel
    [ivr-transfer-on]
    exten => s,1,Answer()
    exten => s,2,agi(googletts.agi, "Veuillez saisir les 10 chiffres au maximum du num�ro de transfert",fr,any)
    exten => s,3,Read(digit,,10,1)
    exten => s,4,Set(DB(${CALLERID(num)}/NUMCF)=${digit})
    exten => s,5,Wait(2)
    exten => s,6,agi(googletts.agi, "Votre transfert est actif sur le num�ro suivant",fr,any)
    exten => s,7,SayDigits(${digit})
    exten => s,8,Hangup()
    
    [ivr-transfer-off]
    exten => s,1,Answer()
    exten => s,2,Set(DB(${CALLERID(num)}/NUMCF)=${CALLERID(num)})
    exten => s,3,agi(googletts.agi, "Votre transfert est � pr�sent inactif",fr,any)
    exten => s,4,Hangup()
    
    ;IVR du menu global du 0185087888
    [ivr-0185087888]
    exten => s,1,Answer()
    exten => s,2,agi(googletts.agi, "Bienvenue chez XXXXXXXX !",fr,any)
    exten => s,3,agi(googletts.agi, "Pour joindre le service apr�s vente, tapez 1", fr, any)
    exten => s,4,agi(googletts.agi, "Pour joindre le service commercial, tapez 2", fr, any)
    exten => s,5,agi(googletts.agi, "Pour joindre le service informatique, tapez 3", fr, any)
    exten => s,6,agi(googletts.agi, "Pour toute autre demande, tapez 4", fr, any)
    exten => s,7,WaitExten()
    
    exten => 1,1,Goto(sav,s,1)
    exten => 2,1,Goto(commercial,s,1)
    exten => 3,1,Goto(informatique,s,1)
    exten => 4,1,Goto(autre,s,1)
    exten => _[5-9#],1,Goto(ivr-0185087888,s,2)
    exten => t,1,Goto(ivr-0185087888,s,2)
    
    [sav]
    exten => s,1,Answer()
    exten => s,2,Wait(1)
    exten => s,3,Dial(SIP/6002,20,tT)
    exten => s,4,Dial(SIP/6001,20,tT)
    exten => s,5,VoiceMail(8000@work)
    exten => s,6,Hangup()
    
    [commercial]
    exten => s,1,Answer()
    exten => s,2,Wait(1)
    exten => s,3,Dial(SIP/6002,20,tT)
    exten => s,4,Dial(SIP/6001,20,tT)
    exten => s,5,VoiceMail(8000@work)
    exten => s,6,Hangup()
    
    [informatique]
    exten => s,1,Answer()
    exten => s,2,Wait(1)
    exten => s,3,Dial(SIP/6003,20,tT)
    exten => s,4,Dial(SIP/6001,20,tT)
    exten => s,5,VoiceMail(8000@work)
    exten => s,6,Hangup()
    
    [autre]
    exten => s,1,Answer()
    exten => s,2,Wait(1)
    exten => s,3,Dial(SIP/6002,20,tT)
    exten => s,4,Dial(SIP/6003,20,tT)
    exten => s,5,Dial(SIP/6001,20,tT)
    exten => s,6,VoiceMail(8000@work)
    exten => s,7,Hangup()
    mon users.conf :

    Code:
    [general]
    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 = yes
    
    [template](!)
    type=friend
    dtmfmode=rfc2833
    disallow=all
    allow=ulaw
    context = work
    host = dynamic
    
    [6001](template)
    fullname = Gregoire
    username = 6001
    secret = 6001
    
    [6002](template)
    fullname = Thomas
    username = 6002
    secret = 6002
    
    [6003](template)
    fullname = Quentin
    username = 6003
    secret = 6003
    
    [6004](template)
    fullname = Salle
    username = 6004
    secret = 6004
    et mon voicemail.conf :

    Code:
    [general]
    format=wav49|gsm|wav
    serveremail=maison-voicemail@test.com
    attach=yes
    maxsilence=10
    silencethreshold=128
    maxlogins=3
    sendvoicemail=yes
    
    ;Corps du mail
    emaildateformat=%A, %d %B %Y a %H:%M:%S
    emailsubject=[ASTERIX] Nouveau message dans la boite ${VM_MAILBOX}
    emailbody=Bonjour ${VM_NAME},\n\n\tLe numero ${VM_CALLERID} a tente de vous joindre sans succes le ${VM_DATE}.\nCette personne vous a laisse un message de ${VM_DUR} secondes. Vous pouvez le consulter en appelant votre boite vocale.\n\n\tBonne journee !\n\n\t\t\t\t--Asterix\n
    pagerfromstring=[Asterix]
    pagersubject=Nouveau message vocal
    pagerbody=Nouveau message de ${VM_DUR} secondes dans la boite ${VM_MAILBOX} laisse le ${VM_DATE} par ${VM_CALLERID}.
    
    [work]
    6001 => 1234,Gregoire
    6002 => 1234,Thomas
    6003 => 1234,Quentin
    8000 => 1234,REUNION
    Je suis dans une impasse, pourriez vous me dire se que vous en pensez ?

  3. #3
    Membre Senior
    Date d'inscription
    septembre 2010
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    Where the sun shines
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    probablement un problème de nat, regarde dans ma signature !

  4. #4
    Membre Association
    Date d'inscription
    août 2010
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    région parisienne
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    si ca fonctionnai avant et que tu n'a rien touché sur l'asterisk, il s'agit probablement d'un changement d'adresse IP (pb de NAT comme le dit mon confrere) ou bien un ajout/modification de regles firewall

  5. #5
    Membre Junior
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    mai 2015
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    Je vous remercie, j'ai pu résoudre mon problème , mon routeur ne routait pas les ports

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