Eingehende ISDN Anrufe an Patton 4552

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av0n

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Hallo,

ich habe mich schon seit Tagen durch dieses Forum gelesen und auch sonst wie wild gesucht, aber ich komme einfach nicht weiter.
Mein Hardware Aufbau ist wie folgt:

ISDN Mehrgeräteanschluss --> Patton --> Asterisk
bzw. umgekehrt und natürlich auch VOIP rein und rausgehend.

Ich habe eine gute Seite gefunden, wo die Konfiguration anhand von Beispielen erklärt wurde.
Abgehend Telefonieren über ISDN und auch über VOIP funktioniert einwandfrei.
Eingehend über VOIP funktioniert auch.

Was ich leider nicht hinbekommen habe ist, eingehende Anrufe von ISDN an Asterisk weiterzuleiten.
Mir kommt es so vor, als ob der Patton die Anrufe zwar registriert, aber sonst nichts weiter tut.
Asterisk "bemerkt" keine ISDN Anrufe.

Für den Anrufer aus dem Festnetz oder GSM Netz kommt nur die Nachricht, dass die Nummer nicht vergeben sei.
Genauso, als wenn ich in einem ISDN Telefon keine MSN eingetragen hätte.

Hier meine Patton Config:

Code: 
cli version 3.20
gui type basic
clock local offset +01:00
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary 10.16.1.254 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 dhcp-server DHCPS_LAN
  network 10.16.99.1 255.240.0.0
  lease 2 hours
  default-router 1 10.16.99.1
  domain-name patton.com
  domain-name-server 1 10.16.99.1

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 10.16.99.1 255.240.0.0
    icmp router-discovery

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

context cs switch
  no digit-collection terminating-char
  digit-collection full-match set-address-complete-indication
  national-prefix 0
  international-prefix 00

  routing-table called-e164 RT_FROM_PSTN
    route 826425 dest-interface IF_SIP_ASTERISK
    route 8264.. dest-interface IF_SIP_ASTERISK
    route default dest-interface IF_SIP_ASTERISK

  interface isdn IF_TE_00
    route call dest-table RT_FROM_PSTN
    call-reroute accept
    call-reroute emit
    diversion accept
    diversion emit
    dtmf-dialing
    use profile tone-set Europe
    isdn-date-time
    caller-name

  interface sip IF_SIP_ASTERISK
    bind context sip-gateway GW_ASTERISK
    route call dest-interface IF_TE_00
    remote 10.16.1.49
    use profile tone-set Europe

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_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_TE_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

port bri 0 1
  no shutdown


Und hier das Log von "debug call-control":
Code: 
10.16.99.1(cfg)#debug call-control
10.16.99.1(cfg)#09:13:57  CC    > [EP IF_TE_00-009c5560/active] Set call-leg pro
perty: E164-Number -> 041*****1
09:13:57  CC    > [EP IF_TE_00-009c5560/active] Set call-leg property: Type-Of-N
umber -> Unknown
09:13:57  CC    > [EP IF_TE_00-009c5560/active] Set call-leg property: Numbering
-Plan -> ISDN/Telephony numbering plan
09:13:57  CC    > [EP IF_TE_00-009c5560/active] Set call-leg property: Presentat
ion-Indicator -> Presentation allowed
09:13:57  CC    > [EP IF_TE_00-009c5560/active] Set call-leg property: Screening
-Indicator -> User provided, not screened
09:13:57  CC    > [EP IF_TE_00-009c5560/active] Set call-leg property: Name ->
09:13:57  CC    > [EP IF_TE_00-009c5560/active] Set call-leg property: Supports
Overlap-Sending -> false
09:13:57  CC    > [EP IF_TE_00-009c5560/active] Set call-leg property: Unique Id
entifier -> 19
09:13:57  CC    > [EP IF_TE_00-009c5560/active] Set call-leg property: Quality-O
f-Service -> MOS 4.50, DS0
09:13:57  CC    > [EP IF_TE_00-009c5560/active] Set call-leg property: Network -
> IF_TE_00
09:13:57  CC    > [EP IF_TE_00-009c5560/active] Set call-leg property: Call-Leg-
ID -> 0x009d9048
09:13:57  CC    > [EP IF_TE_00-009c5560/active] Set call-leg property: State ->
CONNECTED
09:13:57  CC    > [Call 00e4ed78] Set call property: Context -> 0x00000009
09:13:57  CC    > [Call 00e4ed78] Set call property: Information-Transfer-Capabi
lity -> 3.1kHz Audio
09:13:57  CC    > [Call 00e4ed78] Set call property: Hops -> 0x00000010
09:13:57  CC    > [EP IF_TE_00-009c5560/active] Dial to provider router (IF_TE_0
0-precall-service) using call 00e4ed78
09:13:57  CC    > [EP router-00e7df70/incoming] Accept call 00e4ed78
09:13:57  CC    > [EP router-00e7df70/incoming] Set call-leg property: E164-Numb
er -> 826425
09:13:57  CC    > [EP router-00e7df70/incoming] Set call-leg property: Type-Of-N
umber -> Subscriber number
09:13:57  CC    > [EP router-00e7df70/incoming] Set call-leg property: Numbering
-Plan -> ISDN/Telephony numbering plan
09:13:57  CC    > [EP router-00e7df70/incoming] Set call-leg property: Name ->
09:13:57  CC    > [EP router-00e7df70/incoming] Set call-leg property: Network -
> router
09:13:57  CC    > [EP router-00e7df70/incoming] Set call-leg property: Call-Leg-
ID -> 0x009db780
09:13:57  CC    > [EP router-00e7df70/incoming] Set call-leg property: State ->
TRYING
09:13:57  CC    > [EP router-00e7df70] Start route-lookup
09:13:57  CC    > [EP router-00e7df70] Route found; immediately place call
09:13:57  CC    > [EP router-00e7df70] Route to provider 'IF_SIP_ASTERISK'
09:13:57  CC    > [EP router-00e7df70/outgoing] Set call-leg property: E164-Numb
er -> 041*******1
09:13:57  CC    > [EP router-00e7df70/outgoing] Set call-leg property: Type-Of-N
umber -> Unknown
09:13:57  CC    > [EP router-00e7df70/outgoing] Set call-leg property: Numbering
-Plan -> ISDN/Telephony numbering plan
09:13:57  CC    > [EP router-00e7df70/outgoing] Set call-leg property: Presentat
ion-Indicator -> Presentation allowed
09:13:57  CC    > [EP router-00e7df70/outgoing] Set call-leg property: Screening
-Indicator -> User provided, not screened
09:13:57  CC    > [EP router-00e7df70/outgoing] Set call-leg property: Name ->
09:13:57  CC    > [EP router-00e7df70/outgoing] Set call-leg property: Supports
Overlap-Sending -> false
09:13:57  CC    > [EP router-00e7df70/outgoing] Set call-leg property: Unique Id
entifier -> 19
09:13:57  CC    > [EP router-00e7df70/outgoing] Set call-leg property: Network -
> router
09:13:57  CC    > [EP router-00e7df70/outgoing] Set call-leg property: Call-Leg-
ID -> 0x00e7d488
09:13:57  CC    > [EP router-00e7df70/outgoing] Set call-leg property: State ->
CONNECTED
09:13:57  CC    > [Call 009e2080] Set call property: Context -> 0x00000009
09:13:57  CC    > [Call 009e2080] Set call property: Information-Transfer-Capabi
lity -> 3.1kHz Audio
09:13:57  CC    > [Call 009e2080] Set call property: Hops -> 0x0000000f
09:13:57  CC    > [EP router-00e7df70/outgoing] Dial to provider IF_SIP_ASTERISK
 () using call 009e2080
09:13:57  CC    > [EP IF_SIP_ASTERISK-00e7fc68/active] Accept call 009e2080
09:13:57  CC    > [EP IF_SIP_ASTERISK-00e7fc68/active] Set call-leg property: E1
64-Number -> 826425
09:13:57  CC    > [EP IF_SIP_ASTERISK-00e7fc68/active] Set call-leg property: Ty
pe-Of-Number -> Subscriber number
09:13:57  CC    > [EP IF_SIP_ASTERISK-00e7fc68/active] Set call-leg property: Nu
mbering-Plan -> ISDN/Telephony numbering plan
09:13:57  CC    > [EP IF_SIP_ASTERISK-00e7fc68/active] Set call-leg property: Na
me ->
09:13:57  CC    > [EP IF_SIP_ASTERISK-00e7fc68/active] Set call-leg property: Ne
twork -> GW_ASTERISK
09:13:57  CC    > [EP IF_SIP_ASTERISK-00e7fc68/active] Set call-leg property: Ca
ll-Leg-ID -> 0x009e24f8
09:13:57  CC    > [EP IF_SIP_ASTERISK-00e7fc68/active] Set call-leg property: St
ate -> TRYING
09:13:57  CC    > [EP IF_SIP_ASTERISK-00e7fc68/active] Set call-leg property: Su
pported Codecs -> Voice: G.711 A-law[20/20], G.711 u-law[20/20]
09:13:57  CC    > [EP IF_SIP_ASTERISK-00e7fc68/active] Set call-leg property: Ne
twork -> GW_ASTERISK/10.16.99.1
09:13:57  CC    > [Call 00e4ed78] Set call property: Hops -> 0x0000000f
09:13:57  CC    > [EP router-00e7df70/outgoing] Set call-leg property: Quality-O
f-Service -> MOS 4.50, DS0
09:13:57  CC    > [EP router-00e7df70/incoming] Set call-leg property: Supported
 Codecs -> Voice: G.711 A-law[20/20], G.711 u-law[20/20]
09:13:57  CC    > [EP IF_SIP_ASTERISK-00e7fc68/active] Drop call 009e2080
09:13:57  CC    > [EP IF_SIP_ASTERISK-00e7fc68/active] Set call-leg property: Ca
use -> Unallocated number (404)
09:13:57  CC    > [EP IF_SIP_ASTERISK-00e7fc68/active] Set call-leg property: St
ate -> RELEASED
09:13:57  CC    > [EP router-00e7df70/incoming] Set call-leg property: Cause ->
Unallocated number (404)
09:13:57  CC    > [EP router-00e7df70/incoming] Set call-leg property: State ->
RELEASED
09:13:57  CC    > [EP router-00e7df70/outgoing] Drop call 009e2080
09:13:57  CC    > [EP router-00e7df70/outgoing] Set call-leg property: Cause ->
Normal call clearing
09:13:57  CC    > [EP router-00e7df70/outgoing] Set call-leg property: State ->
RELEASED
09:13:57  CC    > [EP IF_TE_00-009c5560/active] Set call-leg property: Cause ->
Unallocated number
09:13:57  CC    > [EP IF_TE_00-009c5560/active] Set call-leg property: Cause ->
Normal call clearing
09:13:57  CC    > [EP IF_TE_00-009c5560/active] Drop call 00e4ed78
09:13:57  CC    > [EP IF_TE_00-009c5560/active] Set call-leg property: State ->
RELEASED


Leider weiss ich da nicht wirklich weiter.
Vielleicht kann mir ja jemand helfen.
Das wäre super.

Grüße aus dem Norden
 
Unglaublich, der Patton war doch richtig konfiguriert.
Das Problem war in der Trixbox.
Es fehlte der Eintrag im SIP Trunk context=from-pstn

....
Gottseidank wieder eine Baustelle weniger.
 
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