Verzweifle an SN 4112 FAX A/B Config

[heiniheini]

folgender Aufbau SN4960 -- ASTERIK 1.6.2 -- SN4112 -- FAX

Ich verzweifle total an der Konfiguration des SN 4122,
eingehend klappt es scheinbar.

Ausgehend klappt es überhaupt nicht.

in der sip.conf ist t38pt_udptl = yes gesetzt.

Der Wählpan passt auch.


Vielen Dank für eure Hilfe.

#----------------------------------------------------------------#
#                                                                #
# SN4112/JS/EUI                                                  #
# R5.2 2009-01-14 H323 SIP FXS FXO                               #
# 2010-03-29T12:09:36                                            #
# SN/00A0BA05276E                                                #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
webserver port 80 language en
sntp-client
sntp-client server primary 129.132.2.21 port 123 version 4

system

  ic voice 0
    low-bitrate-codec g729

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 no-silence-suppression
  fax transmission 1 relay t38-udp

profile pstn default

profile ringing-cadence default
  play 1 1000
  pause 2 4000

profile sip default

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface eth0
    ipaddress dhcp
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

context cs switch
  digit-collection timeout 2

  routing-table called-e164 fax_to_sip
    route .T dest-interface if_sip

  routing-table called-e164 sip_to_fax
    route .T dest-interface FAX

  interface sip if_sip
    bind context sip-gateway sip
    route call dest-table sip_to_fax
    remote asterisk.work.local
    early-connect
    early-disconnect

  interface fxs FAX
    route call dest-table fax_to_sip

context cs switch
  no shutdown

authentication-service user24
  username user24 password /GI3OoxG9/w= encrypted

location-service asterisk
  domain 1 asterisk.work.local 5060

  identity user24
    display-name "Fax Mödling"

    authentication outbound
      authenticate 1 authentication-service user24 username user24

    registration outbound
      registrar asterisk.work.local 5060
      lifetime 3600
      register auto
      retry-timeout on-system-error 10
      retry-timeout on-client-error 10
      retry-timeout on-server-error 10

    call outbound

    call inbound

context sip-gateway sip

  interface lan
    bind interface eth0 context router port 5060

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

port ethernet 0 0
  medium auto
  encapsulation ip
  bind interface eth0 router
  no shutdown

port fxs 0 0
  encapsulation cc-fxs
  bind interface FAX switch
  no shutdown

port fxs 0 1
  shutdown

 

[heiniheini]

wenn ich in der sip.conf

dem fax user

insecure=invite erlaube, dann gehts, doch das muss doch auch sauberer gehen oder?

t38 geht auch leider nicht