(gelöst) Smartnode 1400, Ausgehend bei Rufannahme Ende

[rootkid2]

Schönen guten Tag,

nachdem ich seit laaanger Zeit durch dieses Forum passiv alle meine Probleme lösen konnte und zwischenzeitlich sogar meine Zugangsdaten vergessen habe ;-) habe ich nun ein Problem bei dem ich ratlos bin:

Wenn ich von Intern nach Extern einen Telefonanruf starte, klingelt es bei der Gegenstelle wie gewünscht. Sobald aber die Gegenstelle den Anruf an nimmt beendet sich der Anruf.

Im Log des Smartnode steht leider auch nichts. Eingehende Anrufe können normal entgegen genommen werden. Auch interne Anrufe (von SIP zum ISDN Telefon am NT Port) funktionieren einwandfrei.

Nachdem ich durch Recherche im Forum schon meine anderen Probleme klären konnte (u.A. falsche/keine CallerID), habe ich hier leider nicht den blassesten Schimmer woran das liegen könnte oder wonach ich suchen kann. Hat jemand eine Idee?

Schonmal vielen Dank fürs durchlesen ;-)

/wurzelkind


Meine Konfiguration:

ISDN Telefon--> NT Port SmartNode 1400 --> Asterisk --> bei 0 wieder per SIP an SmartNode 1400 (nötig wegen CDR) --> TE Port ins ISDN Netz

bzw. wenn vom SIP Telefon aus telefoniert wird:
SIP Telefon --> Asterisk --> SmartNode 1400 --> TE Port ins ISDN Netz

Versionen: 
Asterisk       1.6.1.10
SmartNode R3.20 2007-07-13 SIP

Cli Log Asterisk:

    -- Called isdn1/XXXXXXXXXXXX
    -- SIP/isdn1-0000001f is ringing
    -- SIP/isdn1-0000001f is ringing
    -- SIP/isdn1-0000001f is making progress passing it to SIP/me-0000001e
    -- SIP/isdn1-0000001f answered SIP/me-0000001e
    -- Native bridging SIP/me-0000001e and SIP/isdn1-0000001f
  == Spawn extension (default, XXXXXXXXXXXX, 1) exited non-zero on 'SIP/me-0000001e'

Konfiguration SmartNode:

#----------------------------------------------------------------#
#                                                                #
# SN1400                                                         #
# R3.20 2007-07-13 SIP                                           #
# 2010-02-17T12:52:52                                            #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
gui type basic
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary 192.168.4.10 port 123 version 4
sntp-client gmt-offset + 01:00:00

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 0 350 -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 192.168.4.9 255.255.255.0
  lease 2 hours
  default-router 1 192.168.4.9
  domain-name patton.com
  domain-name-server 1 192.168.4.9

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 192.168.4.9 255.255.255.0
    icmp router-discovery

subscriber ppp SUB_PPPOE
  dial out
  authentication chap
  authentication pap
  bind interface IF_IP_WAN router

context cs switch
  national-prefix 0
  international-prefix 00

  routing-table called-e164 RT_SPEED_DIAL
    route T dest-table RT_NR_BLOCKING MT_SPEED_DIAL

  routing-table called-e164 RT_NR_BLOCKING
    route default dest-table RT_CALL_ROUTING MT_SET_CNPN

  routing-table called-e164 RT_CALL_ROUTING
    route default dest-service SER_HG_PSTN_FALLBACK

  routing-table called-e164 RT_INCOMING
    route default dest-interface IF_S0_01
    route 0 dest-interface IF_S0_00

  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_SERVICE

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

  interface isdn IF_DEV0

  interface sip IF_SIP_SERVICE
    bind gateway GW_SIP
    service default
    route call dest-table RT_INCOMING
    use profile voip VOIP

  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

gateway sip GW_SIP
  bind interface IF_IP_LAN router

  service default
    authentication isdn1 password Otx2vJCEWP+8Bb6tqoGkwA== encrypted
    default-server 192.168.4.10 loose-router
    registrar 192.168.4.10
    user isdn1

gateway sip GW_SIP
  no shutdown

port ethernet 0 0
  medium 10 half
  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
  medium 10 half
  bind interface IF_IP_LAN router
  no shutdown

port bri 0 0
  clock slave
  encapsulation q921

  q921
    permanent-layer2
    protocol pmp
    uni-side user
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      encapsulation cc-isdn
      bind interface IF_S0_00 switch

port bri 0 0
  no shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    protocol pmp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side net
      encapsulation cc-isdn
      bind interface IF_S0_01 switch

port bri 0 1
  no shutdown

LÖSUNG:

Ich habe jetzt einfach mal rumgesucht was meine Config von der anderer unterscheidet und dann Try&Error gemacht, gefehlt hat letztendlich

    early-connect
    early-disconnect

im Bereich interface sip IF_SIP_SERVICE