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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
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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 ?
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probablement un problème de nat, regarde dans ma signature !
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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
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Je vous remercie, j'ai pu résoudre mon problème :), mon routeur ne routait pas les ports :)
