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Discussion: Probleme configuration Patton SN4554

  1. #1
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    Probleme configuration Patton SN4554

    Bonjour a tous.
    Nous avons une passerelle Patton SN4554 (192.168.12.164) ou arrive 2 T0 (avec 10 SDA)
    Puis un IPBX Asterisk (Trixbox) (192.168.12.163).

    Le problème est qu'on arrive bien a émettre des appels mais pas a en recevoir.

    Voici la conf du Patton (prise sur un forum)

    cli version 3.20
    clock local offset -04:00
    dns-client server 192.168.23.5
    dns-relay
    webserver port 80 language fr
    sntp-client
    sntp-client server primary 192.168.15.254 port 123 version 4

    system

    ic voice 0
    low-bitrate-codec g729

    system
    clock-source 1 bri 0 0
    clock-source 2 bri 0 1

    profile ppp default

    profile call-progress-tone defaultDialtone
    play 1 1000 440 0

    profile call-progress-tone defaultAlertingtone
    play 1 1500 440 -7
    pause 2 3500

    profile call-progress-tone defaultBusytone
    play 1 500 440 -7
    pause 2 500

    profile tone-set default

    profile voip default
    codec 1 g711alaw64k rx-length 20 tx-length 20
    codec 2 g711ulaw64k rx-length 20 tx-length 20
    fax transmission 1 relay t38-udp

    profile pstn default

    profile sip default

    profile aaa default
    method 1 local
    method 2 none

    context ip router

    interface IF_IP_LAN
    ipaddress 192.168.12.164 255.255.255.0
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

    context ip router
    route 0.0.0.0 0.0.0.0 192.168.12.254 0

    context cs switch
    national-prefix 0
    international-prefix 00

    routing-table called-e164 RT_ISDN_TO_SIP_0
    route T2 dest-interface IF_SIP_0 MAPPING_INCOMING_CALLS

    routing-table called-e164 RT_ISDN_TO_SIP_1
    route T2 dest-interface IF_SIP_1 MAPPING_INCOMING_CALLS

    mapping-table calling-pi to calling-e164 MAP_REMOVE_BLANK_CALLERID
    map restricted to ""

    mapping-table calling-e164 to calling-e164 MAP_LEADING_ZERO
    map (.%) to \1

    complex-function MAPPING_INCOMING_CALLS
    execute 1 MAP_REMOVE_BLANK_CALLERID
    execute 2 MAP_LEADING_ZERO

    interface isdn IF_ISDN_0
    route call dest-table RT_ISDN_TO_SIP_0
    caller-name
    user-side-ringback-tone

    interface isdn IF_ISDN_1
    route call dest-table RT_ISDN_TO_SIP_1
    caller-name
    user-side-ringback-tone

    interface sip IF_SIP_0
    bind context sip-gateway GW_SIP_0
    route call dest-interface IF_ISDN_0
    remote 192.168.12.163
    early-disconnect
    address-translation outgoing-call request-uri user-part fix 10000 host-part to-header target-param none
    address-translation outgoing-call diversion-header host-part call
    address-translation incoming-call calling-e164 fix 0596646867

    interface sip IF_SIP_1
    bind context sip-gateway GW_SIP_1
    route call dest-interface IF_ISDN_1
    remote 192.168.12.163
    early-disconnect
    address-translation outgoing-call request-uri user-part fix 10001 host-part to-header target-param none
    address-translation incoming-call calling-e164 fix 0596646867

    context cs switch
    no shutdown

    authentication-service AS_ALL_LINES
    realm 1 ELASTIX
    username 10000 password LbFCDNv4/Fk= encrypted
    username 10001 password dZ8edXkjFnM= encrypted

    location-service LS_10000
    domain 1 192.168.12.163
    domain 2 192.168.23.25

    identity-group default

    authentication outbound
    authenticate 1 authentication-service AS_ALL_LINES username 10000

    identity 10000

    authentication outbound
    authenticate 1 authentication-service AS_ALL_LINES

    registration outbound
    registrar 192.168.12.163
    lifetime 300
    register auto

    location-service LS_10001
    domain 1 192.168.12.163

    identity-group default

    authentication outbound
    authenticate 1 authentication-service AS_ALL_LINES username 10001

    identity 10001

    authentication outbound
    authenticate 1 authentication-service AS_ALL_LINES

    registration outbound
    registrar 192.168.12.163
    lifetime 300
    register auto

    context sip-gateway GW_SIP_0

    interface LAN
    bind interface IF_IP_LAN context router port 5060

    context sip-gateway GW_SIP_0
    bind location-service LS_10000
    no shutdown

    context sip-gateway GW_SIP_1

    interface LAN
    bind interface IF_IP_LAN context router port 5062

    context sip-gateway GW_SIP_1
    bind location-service LS_10001
    no shutdown

    port ethernet 0 0
    encapsulation ip
    bind interface IF_IP_LAN router
    no shutdown

    port bri 0 0
    clock auto
    encapsulation q921

    q921
    uni-side auto
    encapsulation q931

    q931
    protocol dss1
    uni-side user
    bchan-number-order ascending
    encapsulation cc-isdn
    bind interface IF_ISDN_0 switch

    port bri 0 0
    no shutdown

    port bri 0 1
    clock auto
    encapsulation q921

    q921
    uni-side auto
    encapsulation q931

    q931
    protocol dss1
    uni-side user
    bchan-number-order ascending
    encapsulation cc-isdn
    bind interface IF_ISDN_1 switch

    port bri 0 1
    no shutdown


    Puis sur Asterisk on a créé 2 trunk

    nom du trunk: 10000
    Detail du peer:

    canreinvite=no
    context=from-pstn
    host=192.168.12.164
    dtmfmode=auto
    port=5060
    qualify=yes
    type=friend
    username=10000
    secret=10000
    allow=alaw&ulaw&gsm


    nom du trunk: 10001
    Detail du peer:

    canreinvite=no
    context=from-pstn
    host=192.168.12.164
    dtmfmode=auto
    port=5060
    qualify=yes
    type=friend
    username=10001
    secret=10001
    allow=alaw&ulaw&gsm


    on a creer une route entrant qui route tout vers le poste 6701
    et une route sortant generale vers la patton

    lorsqu'on appelle un SDA depuis l'exterieur on arrive bien a l'IPBX mais pas sur le poste 6701 on arrive sur un message comme quoi le poste n'existe pas.

    Voici les log asterisk: (asterisk -r)

    == 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 [10001@from-sip-external:1] NoOp("SIP/5062-000000b7", "Received incoming SIP connection from unknown peer to 10001") in new stack
    -- Executing [10001@from-sip-external:2] Set("SIP/5062-000000b7", "DID=10001") in new stack
    -- Executing [10001@from-sip-external:3] Goto("SIP/5062-000000b7", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/5062-000000b7", "0?from-trunk,10001,1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/5062-000000b7", "TIMEOUT(absolute)=15") in new stack
    Channel will hangup at 2010-11-24 09:53:42.000 RET.
    -- Executing [s@from-sip-external:3] Answer("SIP/5062-000000b7", "") in new stack
    -- Executing [s@from-sip-external:4] Wait("SIP/5062-000000b7", "2") in new stack
    -- Executing [s@from-sip-external:5] Playback("SIP/5062-000000b7", "ss-noservice") in new stack
    -- <SIP/5062-000000b7> Playing 'ss-noservice.gsm' (language 'en')
    -- Executing [s@from-sip-external:6] PlayTones("SIP/5062-000000b7", "congestion") in new stack
    -- Executing [s@from-sip-external:7] Congestion("SIP/5062-000000b7", "5") in new stack
    == Spawn extension (from-sip-external, s, 7) exited non-zero on 'SIP/5062-000000b7'
    -- Executing [h@from-sip-external:1] NoOp("SIP/5062-000000b7", "Hangup") in new stack
    -- Executing [h@from-sip-external:2] Set("SIP/5062-000000b7", "DID=s") in new stack
    -- Executing [h@from-sip-external:3] Goto("SIP/5062-000000b7", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/5062-000000b7", "0?from-trunk,s,1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/5062-000000b7", "TIMEOUT(absolute)=15") in new stack
    Channel will hangup at 2010-11-24 09:53:51.000 RET.
    -- Executing [s@from-sip-external:3] Answer("SIP/5062-000000b7", "") in new stack
    == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/5062-000000b7'
    trixbox1*CLI>

    j'avoue que c'est un peu du charabia pour moi

    merci de votre aide.

  2. #2
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    fait un essai en modifiant tes trunks avec ces paramètres :
    Code:
    type=peer
    context=from-trunk

  3. #3
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    c'est exactement la même chose

    Merci

    == 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 [10001@from-sip-external:1] NoOp("SIP/5062-000000c4", "Received incoming SIP connection from unknown peer to 10001") in new stack
    -- Executing [10001@from-sip-external:2] Set("SIP/5062-000000c4", "DID=10001") in new stack
    -- Executing [10001@from-sip-external:3] Goto("SIP/5062-000000c4", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/5062-000000c4", "0?from-trunk,10001,1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/5062-000000c4", "TIMEOUT(absolute)=15") in new stack
    Channel will hangup at 2010-11-24 14:23:12.000 RET.
    -- Executing [s@from-sip-external:3] Answer("SIP/5062-000000c4", "") in new stack
    -- Executing [s@from-sip-external:4] Wait("SIP/5062-000000c4", "2") in new stack
    -- Executing [s@from-sip-external:5] Playback("SIP/5062-000000c4", "ss-noservice") in new stack
    -- <SIP/5062-000000c4> Playing 'ss-noservice.gsm' (language 'en')
    -- Executing [s@from-sip-external:6] PlayTones("SIP/5062-000000c4", "congestion") in new stack
    -- Executing [s@from-sip-external:7] Congestion("SIP/5062-000000c4", "5") in new stack
    == Spawn extension (from-sip-external, s, 7) exited non-zero on 'SIP/5062-000000c4'
    -- Executing [h@from-sip-external:1] NoOp("SIP/5062-000000c4", "Hangup") in new stack
    -- Executing [h@from-sip-external:2] Set("SIP/5062-000000c4", "DID=s") in new stack
    -- Executing [h@from-sip-external:3] Goto("SIP/5062-000000c4", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/5062-000000c4", "0?from-trunk,s,1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/5062-000000c4", "TIMEOUT(absolute)=15") in new stack
    Channel will hangup at 2010-11-24 14:23:23.000 RET.
    -- Executing [s@from-sip-external:3] Answer("SIP/5062-000000c4", "") in new stack
    == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/5062-000000c4'

  4. #4
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    Actuellement je suis en R5.6 sur la Patton SN 4554, j'ai vu des tuto avec des version R4.2 je vais downgrader en R4.2 je fait des essai et je reviens.

    et


    je n'arrive pas a la downgrader

    Dernière modification par rock17 ; 24/11/2010 à 13h17.

  5. #5
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    je trouve ces deux lignes bizarre:

    Code:
    -- Executing [10001@from-sip-external:3] Goto("SIP/5062-000000c4", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    y'a pas de raisons de faire un goto "s" - c'est ca qui déclenche le message à la con et le raccroché

  6. #6
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    et si tu regardes plus en détail,

    l'appel arrive sur l'ext 1001, bascule via goto sur s (normalement utilisé quand il n'y a pas de no d'extension fourni), tombe sur un congestion, bascule sur h (logique, raccroché), et refait un goto "s".... c'est pas sain, il n'y a pas de logique à ce que le h renvoie sur le s (ni qu'une exten valide renvoie sur le s)

  7. #7
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    finalement ca fonctionne, tomarch avait presque la reponse, c'etait tout simplement un probleme de configuration du Trunk, j'en ai profité pour faire un tuto:

    http://www.asterisk-france.org/showt...-%28Trixbox%29

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