SmartNode 4552 Layer 3 Link State: down.

hotroot · 20 Jun 2009

Hallo zusammen,

ich bin gerade dabei mir meinen neuen Asterisk aufzubauen. Die Anbindung zum ISDN soll ein SmartNode 4552 übernehmen. Konfiguriert habe ich das ganze am internen ISDN von meinem Asterisk.
Wenn ich jetzt den SmartNode an den NTBA von der Telekom anschliesse leuchtet die LED Line nur sporadisch. Am Kabel kann es nicht liegen. Ich habe es schon mit 4 verschiendenen Kablen (2x LAN, 2x ISDN) versucht. Am internen ISDN vom Asterisk geht die LED sofort an, wenn ich den Stecker einstecke. Stecke ich das Kabel dann in den NTBA der Telekom um, bleibt die LED in 90% der Fälle aus. Auch der Port-Status sieht nicht gerade ermutigend aus:

Code:

Logical Isdn Driver: 0 0
========================

  Slot:                                 0
  Number of Ports:                      2

  Statistics
  ----------

    Leased buffers:                     18
    Max. leased buffers:                48
    Next Call Key:                      4

  Logical Port: 0 0 0
  -------------------

    Admin State:                        Open
    Real State:                         Open

    Operating Layer:                    3
      Link State:                       down

    Layer 2
      Permanent Layer 2:                off
      Protocol:                         PointToMultiPoint
      UniSide:                          User

    Layer 3
      Protocol:                         Dss1
      UniSide:                          User
      MinChannel:                       0
      MaxChannel:                       1
      MaxCalls:                         2
      Hunt Mode:                        Ascending
      Signalling Mode:                  Etsi

Hier meine aktuelle Config.

Code:

#----------------------------------------------------------------#
#                                                                #
# SN4552/2BIS/EUI                                                #
# R5.3 2009-05-20 SIP                                            #
# 2009-06-20T22:32:58                                            #
# SN/00A0BA048992                                                #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
gui type basic
clock local offset +02:00
dns-client server 192.168.10.11
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary 192.168.10.11 port 123 version 4

system

  ic voice 0

profile acl ACL_WAN_PERMIT_ALL_MGMT
  permit 1 ip any any ""

profile acl ACL_WAN_PERMIT_SEL_MGMT
  deny 1 tcp any any eq 23 ""
  deny 2 tcp any any eq 80 ""
  deny 3 udp any any eq 161 ""
  permit 4 ip any any ""

profile acl ACL_WAN_BLOCK_ALL_MGMT
  deny 1 tcp any any eq 23 ""
  deny 2 tcp any any eq 80 ""
  deny 3 udp any any eq 161 ""
  permit 4 ip any any ""

profile service-policy SP_WAN_OUT
  rate-limit 100000 header-length 18 voice-margin 0

  source traffic-class local-voice
    priority

  source traffic-class default
    priority

profile service-policy SP_WAN_IN
  rate-limit 100000 header-length 18 voice-margin 200

  source traffic-class local-voice
    priority

  source traffic-class default
    queue-limit 4

profile napt NAPT_WAN

profile ppp default

profile call-progress-tone US_DIAL_TONE
  play 1 10 350 -13 440 -13

profile call-progress-tone US_RB_TONE
  play 1 2000 440 -19 480 -19
  pause 2 4000

profile call-progress-tone US_BUSY_TONE
  play 1 500 480 -24 620 -24
  pause 2 500

profile call-progress-tone US_CONGESTION_TONE
  play 1 250 480 -24 620 -24
  pause 2 250

profile tone-set default
profile tone-set Europe
profile tone-set UnitedStates
  map call-progress-tone dial-tone US_DIAL_TONE
  map call-progress-tone ringback-tone US_RB_TONE
  map call-progress-tone busy-tone US_BUSY_TONE
  map call-progress-tone release-tone US_BUSY_TONE
  map call-progress-tone congestion-tone US_CONGESTION_TONE

profile voip default
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 g711ulaw64k rx-length 20 tx-length 20

profile voip VOIP
  codec 1 g729 rx-length 20 tx-length 20
  codec 2 g711alaw64k rx-length 20 tx-length 20
  codec 3 g711ulaw64k rx-length 20 tx-length 20
  dejitter-mode static
  dejitter-max-delay 120

profile pstn default

profile sip default

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface IF_IP_WAN
    ipaddress dhcp
    use profile acl ACL_WAN_PERMIT_ALL_MGMT in
    use profile service-policy SP_WAN_IN in
    use profile service-policy SP_WAN_OUT out
    use profile napt NAPT_WAN
    tcp adjust-mss rx 582
    tcp adjust-mss tx 1440

  interface IF_IP_LAN
    ipaddress 192.168.10.29 255.255.255.0
    icmp router-discovery

context ip router
  route 0.0.0.0 0.0.0.0 192.168.10.11 0

subscriber ppp SUB_PPPOE
  dial out
  no multilink
  authentication chap
  authentication pap
  bind interface IF_IP_WAN router

context cs switch
  national-prefix 0
  international-prefix 00

  routing-table called-e164 RT_ASTERISK
    route 2. dest-interface IF_S0_01
    route XXX65 dest-interface IF_S0_01
    route XXX316 dest-interface IF_S0_01
    route default dest-interface IF_S0_00
    route XXX298[1235] dest-interface IF_S0_01

  routing-table called-e164 RT_OVERLAPDIAL
    route .T dest-interface IF_SIP_ASTERISK

  mapping-table called-e164 to called-e164 MT_SPEED_DIAL
  mapping-table calling-e164 to calling-e164 MT_SET_CNPN
  mapping-table called-e164 to called-e164 MT_NR_TRANSLATION

  interface isdn IF_S0_00
    route call dest-interface IF_SIP_ASTERISK

  interface isdn IF_S0_01
    route call dest-table RT_OVERLAPDIAL
    use profile tone-set Europe
    isdn-date-time

  interface sip IF_SIP_ASTERISK
    bind context sip-gateway GW_ASTERISK
    route call dest-table RT_ASTERISK
    remote 192.168.10.19 5060

  service hunt-group SER_HG_PSTN_FALLBACK
    timeout 6
    drop-cause normal-unspecified
    drop-cause no-circuit-channel-available
    drop-cause network-out-of-order
    drop-cause temporary-failure
    drop-cause switching-equipment-congestion
    drop-cause access-info-discarded
    drop-cause circuit-channel-not-available
    drop-cause resources-unavailable
    drop-cause no-route-to-destination
    route call 1 dest-interface IF_SIP_SERVICE
    route call 2 dest-interface IF_S0_00

context cs switch
  no shutdown

authentication-service AUTH_SVC

location-service LOCATION_SVC

  identity-group default

    authentication outbound
      authenticate 1 authentication-service AUTH_SVC

    registration outbound
      register auto

    call outbound

context sip-gateway GW_SIP

  interface SIP_WAN
    bind interface IF_IP_WAN context router port 5060

context sip-gateway GW_SIP
  bind location-service LOCATION_SVC
  no shutdown

context sip-gateway GW_ASTERISK

  interface IF_LAN
    bind interface IF_IP_LAN context router port 5060

context sip-gateway GW_ASTERISK
  no shutdown

port ethernet 0 0
  bind interface IF_IP_WAN router

  pppoe

    session SES_PPPOE
      bind subscriber SUB_PPPOE
      shutdown

port ethernet 0 0
  no shutdown

port ethernet 0 1
  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_S0_00 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 net
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_01 switch

port bri 0 1
  no shutdown

Ich vermute, dass damit auch mein Problem zusammen hängt, dass ich bei ausgehenden Gesprächen (Telefon->Patton (Port Phone)->Asterisk->Patton (Port Line) -> ISDN) nichts höre.

Kann mir jemand sagen, ob ich in der Config etwas übersehen habe oder ob ich den Patton wieder einpacken und zurückschicken kann?

Danke und Gruss

hotroot

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SmartNode 4552 Layer 3 Link State: down.

hotroot

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