Aucun appel sortant vers ligne RTC: passerelle VOIP SPA3102

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25/10/2011, 13h57
valerys

Aucun appel sortant vers ligne RTC: passerelle VOIP SPA3102

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')
25/10/2011, 14h55
jean

Citation:

-- 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 ?
25/10/2011, 15h28
valerys

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
25/10/2011, 22h34
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/
26/10/2011, 13h01
therebel23

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

33118058

C'est quoi ce numéro ? Essaie au format national : 0XXXXXXXXX
26/10/2011, 15h39
astux

http://www.asterisk-france.org/showt...-spa3102/page2
26/10/2011, 16h14
valerys

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
27/10/2011, 11h23
valerys

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
27/10/2011, 16h23
jean

il faudrait comparer les données du syslog entre un appel qui marche et un qui marche pas..
27/10/2011, 16h39
valerys

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
27/10/2011, 16h46
jean

on est d'accord, le num appelé est bien: 33118058 ?

Citation:

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

- on dirait que c'est le réseau qui relache l'appel. Est ce qu'un poste analogique connecté à la prise ou est connecté le SPA fonctionne ?

- si oui, il faut rechercher le paramétrage du spa, voir si il n'y a pas des trucs qui dépendent du pays - je commencerais par utiliser les paramètres des différents impérialises ;-) france, usa...