spa112

[davlefou]

bonjour,
j'ai un spa112 et j'arrive pas à le faire communiquer avec asterisk. Voici la trace d'un appels :

 -- Accepting AUTHENTICATED call from 41.225.110.165:
       > requested format = ulaw,
       > requested prefs = (),
       > actual format = ulaw,
       > host prefs = (ulaw|alaw|gsm),
       > priority = mine
    -- Executing [103@direction:1] Ringing("IAX2/sabi-14981", "") in new stack
    -- Executing [103@direction:2] Dial("IAX2/sabi-14981", "SIP/sabi") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/sabi
    -- Got SIP response 503 "Service Unavailable" back from 41.225.110.165:5061
    -- SIP/sabi-000000b1 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Auto fallthrough, channel 'IAX2/sabi-14981' status is 'CONGESTION'
 
And the user:

le compte sip :

[sabi]
username=sabi
secret=passwd
type=friend
host=dynamic
mailbox=sabi.nounette@rsa.com
context=direction
callerid=103
insecure=invite,port
dtmfmode=rfc2833
nat=yes
disallow=all
allow=g729
allow=ulaw
allow=alaw
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en détail :

sip show peer sabi
 
 
  * Name       : sabi
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : manager
  Subscr.Cont. : <Not set>
  Language     : fr
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  MOH Suggest  :
  Mailbox      : sabi.nounette@rsa.com
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Max forwards : 0
  Dynamic      : Yes
  Callerid     : <33484251592>
  MaxCallBR    : 384 kbps
  Expire       : 30
  Insecure     : port,invite
  Force rport  : Yes
  ACL          : No
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : Yes
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       :
  Addr->IP     : 41.225.110.165:5061
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: sabi
  SIP Options  : (none)
  Codecs       : 0x10c (ulaw|alaw|g729)
  Codec Order  : (g729:20,ulaw:20,alaw:20)
  Auto-Framing :  No
  100 on REG   : No
  Status       : Unmonitored
  Useragent    : Cisco/SPA112-1.3.1(003)
  Reg. Contact : sip:sabi@192.168.0.155:5061
  Qualify Freq : 60000 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   :
  Use Reason   : No
  Encryption   : No

Il ne sonne et les appels ne passe pas! Quand l'appel, asterisk ne réagit pas du tout.