Patton 4554 - Anrufe aus PSTN gehen nicht auf Durchwahl, von GSM OK

[Heysi]

Hallo zusammen!

Wir haben da ein merkwürdiges Problem!
Eingehende Anrufe auf das Patton gehen vom GSM Netz einwandfrei direkt auf Nebenstellen.
Jedoch wird von einen Festnetz Anschluss angerufen, landen die Calls immer auf der Hauptnummer, und Durchwahlen gehen nicht
Hat jemand eine Idee was das sein könnte?
Wenn ich das debugging aufdrehe scheint's so als ob die Durchwahl nicht mitgebeben wird. Evtl. fehlt irgendwo ein Timeout?
Das einzige was wir routingtechnisch machen ist eingehende Anrufe eine 0 vorstellen und dann auf 999 schicken (Hauptnummer) bzw. an 999[Durchwahl]

Hoffe jemand von euch hat eine Idee, anbei die Konfig:

#----------------------------------------------------------------#
#                                                                #
# SN4554/2BIS/EUI                                                #
# R5.3 2009-03-18 SIP                                            #
# 2009-06-19T12:08:48                                            #
# SN/00A0BA0467CB                                                #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
clock local offset +02:00
webserver port 80 language en
sntp-client
sntp-client server primary 10.20.20.1 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 420 -6

profile call-progress-tone defaultAlertingtone
  play 1 1000 420 -13
  pause 2 5000

profile call-progress-tone defaultBusytone
  play 1 400 420 -7
  pause 2 400

profile call-progress-tone defaultReleasetone
  play 1 200 420 -7
  pause 2 200

profile call-progress-tone defaultCongestiontone
  play 1 200 420 -7
  pause 2 200

profile tone-set default

profile voip default
  codec 1 g711alaw64k rx-length 20 tx-length 20 no-silence-suppression
  codec 2 g711ulaw64k rx-length 20 tx-length 20 no-silence-suppression
  rtp traffic-class local-default
  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 10.20.20.7 255.255.0.0
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

context cs switch
  digit-collection timeout 5 set-address-complete-indication
  digit-collection terminating-char # set-address-complete-indication
  digit-collection full-match set-address-complete-indication
  national-prefix 0
  international-prefix 00

  routing-table called-e164 RT_ISDN_TO_SIP_0
    route default dest-interface IF-ASTERISK 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

  mapping-table called-e164 to called-e164 999
    map () to 999\1

  mapping-table calling-e164 to calling-e164 add_zero
    map (.%) to 0\1

  mapping-table calling-e164 to calling-name anonym
    map () to anonym

  complex-function MAPPING_INCOMING_CALLS
    execute 1 999
    execute 2 MAP_REMOVE_BLANK_CALLERID
    execute 3 MAP_LEADING_ZERO
    execute 4 add_zero

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

  interface isdn IF_ISDN_1
    route call dest-table RT_ISDN_TO_SIP_0
    user-side-ringback-tone

  interface sip IF-ASTERISK
    bind context sip-gateway asterisk
    route call dest-service isdnports
    remote 10.20.20.1 5060

  service sip-location-service 10.20.20.1
    bind location-service 10.20.20.1
    mode hunt
    hunt-timeout 20

  service sip-location-service ASTERISK_SRV
    bind location-service ASTERISK_SRV
    mode hunt
    hunt-timeout 20

  service hunt-group isdnports
    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
    route call 1 dest-interface IF_ISDN_0
    route call 2 dest-interface IF_ISDN_1

  service hunt-group tosip
    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
    route call 1 dest-service ASTERISK_SRV

context cs switch
  no shutdown

authentication-service patton
  realm 1 asterisk
  username patton password Otx2vJCEWP+8Bb6tqoGkwA== encrypted

location-service ASTERISK_SRV
  domain 1 10.20.20.1 5060
  domain 2 asterisk 5060
  match-any-domain

  identity-group default

    authentication outbound
      authenticate none

    authentication inbound
      authenticate 1 authentication-service patton username patton

    registration inbound
      contact 10.20.20.1 5060 switch IF-ASTERISK priority 1000

  identity patton
    alias name patton

    authentication outbound
      authenticate 1 authentication-service patton username patton

    authentication inbound
      authenticate 1 authentication-service patton username patton

    registration outbound
      registrar 10.20.20.1 5060
      proxy none
      lifetime 3600
      register auto
      retry-timeout on-system-error 10
      retry-timeout on-client-error 10
      retry-timeout on-server-error 10

    registration inbound
      contact 10.20.20.1 5060 switch IF-ASTERISK priority 1

    call outbound
      use profile tone-set default
      use profile voip default
      use profile sip default
      preferred-transport-protocol udp
      invite-transaction-timeout 32
      non-invite-transaction-timeout 32

    call inbound
      use profile tone-set default
      use profile voip default
      use profile sip default

context sip-gateway asterisk

  interface asterisk
    bind interface IF_IP_LAN context router port 5060

context sip-gateway asterisk
  bind location-service ASTERISK_SRV
  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
    permanent-layer2
    protocol pp
    uni-side user
    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
    permanent-layer2
    protocol pp
    uni-side user
    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

 

[matgoe]

Hallo, ich stehe gerade vor einem ähnlichen Problem.

Hast du schon eine Lösung dafür gefunden?

 

[Heysi]

Hy matoge!

Ich habs inzwischen gelöst. Das Problem ist dass du bei den Routingregeln noch T4 hinten dran hängen muss. Die Anlagen in Europa schicken manchmal die Durchwahlen erst nach, somit muss Patton auf die Durchwahlen warten, bevor es weiterroutet.

Hier meine komplette Konfig, die macht folgendes:
- 1 Trunk zu Asterisk für beide ISDN Ports
- Eingehenden Anrufen wird immer eine 999 vorangestellt
- Der Callerid wird eine 0 vorangestellt
- Kommt ein anonymer Anruf rein, ist die Callerid 0
- Bei rausgehenden Anrufen wird zuerst ISDN Port 1 genommen, wenn voll
dann ISDN Port 2

Musst die Konfig halt deinen wünschen nach anpassen...

#----------------------------------------------------------------#
#                                                                #
# SN4554/2BIS/EUI                                                #
# R5.3 2009-03-18 SIP                                            #
# 2009-06-19T12:08:48                                            #
# SN/00A0BA0467CB                                                #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
clock local offset +02:00
webserver port 80 language en
sntp-client
sntp-client server primary 10.20.20.1 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 420 -6

profile call-progress-tone defaultAlertingtone
  play 1 1000 420 -13
  pause 2 5000

profile call-progress-tone defaultBusytone
  play 1 400 420 -7
  pause 2 400

profile call-progress-tone defaultReleasetone
  play 1 200 420 -7
  pause 2 200

profile call-progress-tone defaultCongestiontone
  play 1 200 420 -7
  pause 2 200

profile tone-set default

profile voip default
  codec 1 g711alaw64k rx-length 20 tx-length 20 no-silence-suppression
  codec 2 g711ulaw64k rx-length 20 tx-length 20 no-silence-suppression
  rtp traffic-class local-default
  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 IP_IP_WAN
    ipaddress 10.20.20.7 255.255.0.0
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

context cs switch
  national-prefix 0
  international-prefix 00


  routing-table called-e164 RT_ISDN_TO_SIP_0
    route T4 dest-interface IF-ASTERISK MAPPING_INCOMING_CALLS

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

  mapping-table called-e164 to called-e164 999
    map (.%) to 999\1

  mapping-table calling-e164 to calling-e164 add_zero
    map (.%) to 0\1

  complex-function MAPPING_INCOMING_CALLS
    execute 1 999
    execute 2 MAP_REMOVE_BLANK_CALLERID
    execute 3 add_zero

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

  interface isdn IF_ISDN_1
    route call dest-table RT_ISDN_TO_SIP_0
    user-side-ringback-tone

  interface sip IF-ASTERISK
    bind context sip-gateway asterisk
    route call dest-service isdnports
    remote 10.20.20.1 5060

  service sip-location-service 10.20.20.1
    bind location-service 10.20.20.1
    mode hunt
    hunt-timeout 20

  service sip-location-service ASTERISK_SRV
    bind location-service ASTERISK_SRV
    mode hunt
    hunt-timeout 20

  service hunt-group isdnports
    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
    route call 1 dest-interface IF_ISDN_0
    route call 2 dest-interface IF_ISDN_1

  service hunt-group tosip
    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
    route call 1 dest-service ASTERISK_SRV

context cs switch
  no shutdown

authentication-service patton
  realm 1 asterisk
  username patton password Otx2vJCEWP+8Bb6tqoGkwA== encrypted

location-service ASTERISK_SRV
  domain 1 10.20.20.1 5060
  domain 2 asterisk 5060
  match-any-domain

  identity-group default

    authentication outbound
      authenticate none

    authentication inbound
      authenticate 1 authentication-service patton username patton

    registration inbound
      contact 10.20.20.1 5060 switch IF-ASTERISK priority 1000

  identity patton
    alias name patton

    authentication outbound
      authenticate 1 authentication-service patton username patton

    authentication inbound
      authenticate 1 authentication-service patton username patton

    registration outbound
      registrar 10.20.20.1 5060
      proxy none
      lifetime 3600
      register auto
      retry-timeout on-system-error 10
      retry-timeout on-client-error 10
      retry-timeout on-server-error 10

    registration inbound
      contact 10.20.20.1 5060 switch IF-ASTERISK priority 1

    call outbound
      use profile tone-set default
      use profile voip default
      use profile sip default
      preferred-transport-protocol udp
      invite-transaction-timeout 32
      non-invite-transaction-timeout 32

    call inbound
      use profile tone-set default
      use profile voip default
      use profile sip default

context sip-gateway asterisk

  interface asterisk
    bind interface IP_IP_WAN context router port 5060

context sip-gateway asterisk
  bind location-service ASTERISK_SRV
  no shutdown

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

port bri 0 0
  clock auto
  encapsulation q921

  q921
    permanent-layer2
    protocol pp
    uni-side user
    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
    permanent-layer2
    protocol pp
    uni-side user
    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