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