Aucun appel sortant vers ligne RTC: passerelle VOIP SPA3102

**valerys** · 25/10/2011, 13h57

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

Code:

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** · 25/10/2011, 14h55

-- 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** · 25/10/2011, 15h28

sip show peers pstn donne:

Code:

* 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** · 25/10/2011, 22h34

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** · 26/10/2011, 13h01

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

33118058

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

**astux** · 26/10/2011, 15h39

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

**valerys** · 26/10/2011, 16h14

C'est un numéro du Cameroun je suis au cameroun et les numéros commence ici par 2, 3, et 4 et ont tous 8 chiffre d'ou le numéros 33 11 80 58. Merci

**valerys** · 27/10/2011, 11h23

Bonjour jean, merci pour ton aide précieuse.
j'ai effectué les actions que tu m'as suggéré. J'ai remarqué que le syslog du SPA 3102 gueule lorsque les appels vers la ligne RTC sont fait à partir du poste connecté à lui et pas du tout lorsque les appels sont passé à partir des autres postes ou des softphones. Je joins aussi ma config (extentions_additional.conf et sip_additional.conf)si jamais ca peut vous guider à m'aider un peu plus merci de votre disponibilité

Edit: extensions_additional.conf

[from-trunk-sip-pstn]
include => from-trunk-sip-pstn-custom
exten => _.,1,Set(GROUP()=OUT_3)
exten => _.,n,Goto(from-trunk,${EXTEN},1)

; end of [from-trunk-sip-pstn]

[outrt-001-9_outside]
include => outrt-001-9_outside-custom
exten => _9.,1,Macro(user-callerid,SKIPTTL,)
exten => _9.,n,Set(_NODEST=)
exten => _9.,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _9.,n,Macro(dialout-trunk,1,${EXTEN:1},)
exten => _9.,n,Macro(outisbusy,)

; end of [outrt-001-9_outside]

[outrt-003-pstn-camtel]
include => outrt-003-pstn-camtel-custom
exten => _22XXXXXX,1,Macro(user-callerid,SKIPTTL,)
exten => _22XXXXXX,n,Set(_NODEST=)
exten => _22XXXXXX,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _22XXXXXX,n,Macro(dialout-trunk,3,${EXTEN},,)
exten => _22XXXXXX,n,Macro(outisbusy,)
exten => _33XXXXXX,1,Macro(user-callerid,SKIPTTL,)
exten => _33XXXXXX,n,Set(_NODEST=)
exten => _33XXXXXX,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _33XXXXXX,n,Macro(dialout-trunk,3,${EXTEN},,)
exten => _33XXXXXX,n,Macro(outisbusy,)

; end of [outrt-003-pstn-camtel]

------------------------------

Edit: sip_additional.conf

[2100] ; extenxion de la passerelle SPA 3102
deny=0.0.0.0/0.0.0.0
type=friend
secret=21002100
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=2100@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/2100
context=from-internal
canreinvite=no
callgroup=
callerid=device <2100>
accountcode=
call-limit=50

[pstn] ; trunk de la passerelle SPA3102
canreinvite=no
context=from-pstn
host=192.168.1.20
nat=no
port=5060
user=pstn
secret=pstnpass
type=friend
qualify=yes
dtmfmode=rfc2833

**jean** · 27/10/2011, 16h23

il faudrait comparer les données du syslog entre un appel qui marche et un qui marche pas..

**valerys** · 27/10/2011, 16h39

Bsr jean en fait aucun appel ne sort avec tous les postes et softphone, le message dans le combiné est: Toute les lignes sont actuellement OQP veuillez recommencer ultérieurement".
La diférence avec le syslog du SPA 3102 est: lorsque l'appel est emis depuis le poste connecté directement sur lui, le syslog enregistre les logs ci-desous et pour les autres postes, il n'y rien dans le syslog:
Oct 27 15:47:19 192.168.1.20
Oct 27 15:49:33 192.168.1.20 2. Report digit 3 (1)(40 ms)
Oct 27 15:49:33 192.168.1.20 2. Report digit 3 (1)(40 ms)
Oct 27 15:49:34 192.168.1.20 2. Report digit 1 (1)(40 ms)
Oct 27 15:49:39 192.168.1.20 2. Report digit 1 (1)(40 ms)
Oct 27 15:49:40 192.168.1.20 2. Report digit 8 (1)(40 ms)
Oct 27 15:49:40 192.168.1.20 2. Report digit 0 (1)(40 ms)
Oct 27 15:49:40 192.168.1.20 2. Report digit 5 (1)(40 ms)
Oct 27 15:49:46 192.168.1.20 2. Report digit 8 (1)(40 ms)
Oct 27 15:49:47 192.168.1.20 Calling:33118058@192.168.1.10:0
Oct 27 15:49:48 192.168.1.20 [0:0]AUD ALLOC CALL (port=16384)
Oct 27 15:49:48 192.168.1.20 [0:0]RTP Rx Up
Oct 27 15:49:49 192.168.1.20 AUD:Stop PSTN Tone
Oct 27 15:49:49 192.168.1.20 [0:0]ENC INIT 0
Oct 27 15:49:50 192.168.1.20 [0:0]RTP Tx Up (pt=0->c0a8010a:14620)
Oct 27 15:49:50 192.168.1.20 [0:0]RTCP Tx Up
Oct 27 15:49:51 192.168.1.20 CC:CallProgress
Oct 27 15:49:52 192.168.1.20 [0:0]RTP Rx 1st PKT @16384(2)
Oct 27 15:49:52 192.168.1.20 [0:0]DEC INIT 0
Oct 27 15:49:53 192.168.1.20 [0:0]AUD Rel Call
Oct 27 15:49:53 192.168.1.20 CC:Failed w/ Calling
Oct 27 15:49:56 192.168.1.20 FXO:PSTN Disconnect Tone
Oct 27 15:49:57 192.168.1.20 AUD:Stop PSTN Tone
Oct 27 15:49:58 192.168.1.20 AUD:Stop PSTN Tone
Oct 27 15:49:58 192.168.1.20 FXO:Stop CNDD

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