Aucun appel sortant vers ligne RTC: passerelle VOIP SPA3102

[valerys]

Bonjour à tous,
j'ai installé un serveur Trixbox v2.8.0.4. les appels interne passent sans problème. Mais j'ai un problèmes pour la configuration de la passerelle linksys SPA 3102. je reçois les appel venant de la ligne RTC mais je n'arrive à émettre aucun appel vers cette ligne. Le message vocal dit:"toutes les lignes sont actuellement occupés. Veuillez renouveler votre appel ultérieurement" vous verrez ci-dessous les log de la CLI asterix. Merci de me venir en aide

Verbosity was 3 and is now 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 6
    -- Executing [33118058@from-internal:1] Macro("SIP/2100-0000000d", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/2100-0000000d", "AMPUSER=2100") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/2100-0000000d", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/2100-0000000d", "1?Set(REALCALLERIDNUM=2100)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/2100-0000000d", "AMPUSER=2100") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/2100-0000000d", "AMPUSERCIDNAME=Passerelle SPA") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/2100-0000000d", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/2100-0000000d", "AMPUSERCID=2100") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/2100-0000000d", "CALLERID(all)="Passerelle SPA" <2100>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/2100-0000000d", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/2100-0000000d", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/2100-0000000d", "Using CallerID "Passerelle SPA" <2100>") in new stack
    -- Executing [33118058@from-internal:2] Set("SIP/2100-0000000d", "_NODEST=") in new stack
    -- Executing [33118058@from-internal:3] Macro("SIP/2100-0000000d", "record-enable,2100,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/2100-0000000d", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/2100-0000000d", "recordingcheck,20111025-130212,1319544132.13") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
 recordingcheck,20111025-130212,1319544132.13: Outbound recording not enabled
    -- <SIP/2100-0000000d>AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/2100-0000000d", "") in new stack
    -- Executing [33118058@from-internal:4] Macro("SIP/2100-0000000d", "dialout-trunk,3,33118058,,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/2100-0000000d", "DIAL_TRUNK=3") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/2100-0000000d", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/2100-0000000d", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/2100-0000000d", "DIAL_NUMBER=33118058") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/2100-0000000d", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/2100-0000000d", "OUTBOUND_GROUP=OUT_3") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/2100-0000000d", "0?nomax") in new stack
    -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/2100-0000000d", "0?chanfull") in new stack
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/2100-0000000d", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/2100-0000000d", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/2100-0000000d", "outbound-callerid,3") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/2100-0000000d", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/2100-0000000d", "0?Set(REALCALLERIDNUM=2100)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/2100-0000000d", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/2100-0000000d", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/2100-0000000d", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/2100-0000000d", "TRUNKOUTCID=GTS <33429786>") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/2100-0000000d", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/2100-0000000d", "1?Set(CALLERID(all)=GTS <33429786>)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/2100-0000000d", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/2100-0000000d", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/2100-0000000d", "1?AGI(fixlocalprefix)") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
       > fixlocalprefix: Using pattern 33+XXXXXX
       > fixlocalprefix: Using pattern 22+XXXXXX
    -- <SIP/2100-0000000d>AGI Script fixlocalprefix completed, returning 0
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/2100-0000000d", "OUTNUM=33118058") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/2100-0000000d", "custom=SIP/pstn") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/2100-0000000d", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/2100-0000000d", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/2100-0000000d", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/2100-0000000d", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/2100-0000000d", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/2100-0000000d", "SIP/pstn/33118058,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 6
    -- Called pstn/33118058
    -- SIP/pstn-0000000e is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s@macro-dialout-trunk:20] Goto("SIP/2100-0000000d", "s-CONGESTION,1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/2100-0000000d", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,3)
    -- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/2100-0000000d", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
    -- Executing [33118058@from-internal:5] Macro("SIP/2100-0000000d", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Playback("SIP/2100-0000000d", "all-circuits-busy-now,noanswer") in new stack
    -- <SIP/2100-0000000d> Playing 'all-circuits-busy-now.gsm' (language 'fr')
    -- Executing [s@macro-outisbusy:2] Playback("SIP/2100-0000000d", "pls-try-call-later,noanswer") in new stack
    -- <SIP/2100-0000000d> Playing 'pls-try-call-later.gsm' (language 'fr')

[jean]

-- Executing [s@macro-dialout-trunk:19] Dial("SIP/2100-0000000d", "SIP/pstn/33118058,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
-- Called pstn/33118058
-- SIP/pstn-0000000e is circuit-busy

que donne sip show peer pstn ?

[valerys]

sip show peers pstn donne:

* Name       : pstn
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : from-pstn
  Subscr.Cont. : <Not set>
  Language     : fr
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  Mailbox      : 
  VM Extension : *97
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : No
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : no
  Nat          : RFC3581
  ACL          : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: Yes
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : 192.168.1.20
  Addr->IP     : 192.168.1.20 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Transport    : UDP
  Def. Username: 
  SIP Options  : (none)
  Codecs       : 0x28000c (ulaw|alaw|h263|h264)
  Codec Order  : (ulaw:20,alaw:20)
  Auto-Framing :  No 
  100 on REG   : No
  Status       : OK (9 ms)
  Useragent    : 
  Reg. Contact : 
  Qualify Freq : 60000 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs

[jean]

le statut du peer est ok, donc, il est vraisemblable que c'est la passerelle qui jette l'appel et pas un pbm entre asterisk et la passerelle

pour confirmer, un sip set debug on
avant l'etablissement d'appel serait utile

je connais pas cette passerelle en particulier, mais il doit y avoir un moyen d'envoyer des traces de la passerelle (genre, via le syslog de ton serveur). il faudrait faire ca

un coup de google 'syslog spa3102' donne
http://www.zultron.com/2008/11/spa31...freepbx-howto/

[therebel23]

Ca vient peut etre du numéro que tu composes ???

33118058

C'est quoi ce numéro ? Essaie au format national : 0XXXXXXXXX

[astux]

http://www.asterisk-france.org/showt...-spa3102/page2