Bin zu blöd für smartnode 4638 und asterisk

easyhugo · 16 Mrz 2011

Hi an alle

Erst mal SORRY für das alte Thema, aber ich komm einfach nicht weiter.
Bin schon drei Tage durch die Welt von Patton und schaffs einfach nicht richtig bzw versteht ichs nicht.
Komisch ist nur dass ich Telefonieren kann rein wie raus.
Erst mal die Configs

Code:

#----------------------------------------------------------------#
#                                                                #
# SN4638/5BIS                                                    #
# R5.7 2011-01-17 H323 SIP BRI                                   #
# 2011-03-15T19:42:16                                            #
# SN/00A0BA06191F                                                #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
clock local default-offset +00:00
dns-client server 208.67.222.222
dns-client server 208.67.220.220
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary 192.168.2.240 port 123 version 4
sntp-client server secondary 129.132.2.21 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
  clock-source 3 bri 0 2
  clock-source 4 bri 0 3
  clock-source 5 bri 0 4

profile napt NAPT_WAN

profile ppp default

profile tone-set default

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

profile pstn default

profile sip default
  no autonomous-transitioning

profile dhcp-server DHCPS_LAN
  network 192.168.1.0 255.255.255.0
  include 1 192.168.1.10 192.168.1.99
  lease 2 hours
  default-router 1 192.168.1.1
  domain-name-server 1 192.168.1.1

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface WAN
    ipaddress dhcp
    use profile napt NAPT_WAN
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

  interface LAN
    ipaddress 192.168.2.10 255.255.255.0
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

context ip router
  dhcp-server use profile DHCPS_LAN
  route 0.0.0.0 0.0.0.0 192.168.2.240 0

context cs switch
  national-prefix 0
  international-prefix 00

  interface isdn IF-BRI-1
    route call dest-interface IF-SIP-1

  interface isdn IF-BRI-2
    route call dest-interface IF-SIP-1

  interface isdn IF-BRI-3
    route call dest-interface IF-SIP-1

  interface isdn IF-BRI-4
    route call dest-interface IF-SIP-1

  interface sip IF-SIP-1
    bind context sip-gateway GW_ASTERISK_1
    route call dest-service HG-ISDN
    remote 192.168.2.240 5060
    early-disconnect
    privacy

  service hunt-group HG-ISDN
    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-BRI-1
    route call 2 dest-interface IF-BRI-2
    route call 3 dest-interface IF-BRI-3
    route call 4 dest-interface IF-BRI-4

context cs switch
  no shutdown

authentication-service AUTH_ASTERISK
  username isdn password 5XCeM+qgu4M= encrypted

location-service LOC_SERV_ASTERISK_1
  domain 1 192.168.2.240

  identity isdn

    authentication outbound
      authenticate 1 authentication-service AUTH_ASTERISK username isdn

    registration outbound
      registrar 192.168.2.240 5060
      lifetime 60
      register auto
      retry-timeout on-system-error 10
      retry-timeout on-client-error 10
      retry-timeout on-server-error 10

context sip-gateway GW_ASTERISK_1

  interface IF_GW_SIP_ASTERISK_1
    bind interface LAN context router port 5060

context sip-gateway GW_ASTERISK_1
  bind location-service LOC_SERV_ASTERISK_1
  no shutdown

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

port ethernet 0 1
  medium auto
  encapsulation ip
  bind interface LAN router
  no shutdown

port bri 0 0
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF-BRI-1 switch

port bri 0 0
  no shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF-BRI-2 switch

port bri 0 1
  no shutdown

port bri 0 2
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF-BRI-3 switch

port bri 0 2
  no shutdown

port bri 0 3
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF-BRI-4 switch

port bri 0 3
  no shutdown

port bri 0 4
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn

port bri 0 4
  shutdown

die sip conf:

Code:

username=isdn
type=friend
secret=xxxxx
qualify=1200
insecure=very
host=192.168.2.10
dtmfmode=rfc2833
disallow=all
allow=alaw

und da beginnts schon bei host sollte doch dynamic rein wenn ich da nicht die ip vom smartnode drinnen hab kann ich nicht telefonieren

die sip debug schaut schaut so aus dazu

Code:

Really destroying SIP dialog '[email protected]' Method: OPTIONS

<--- SIP read from UDP:192.168.2.10:5060 --->
REGISTER sip:192.168.2.240:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK1d209edca508c8b97
Max-Forwards: 70
From: <sip:[email protected]>;tag=ff9fffe939
To: <sip:[email protected]>
Call-ID: 53ef1dd932879ed4
CSeq: 14013 REGISTER
Contact: <sip:[email protected]:5060>
Expires: 60
User-Agent: Patton SN4638 5BIS 00A0BA06191F R5.7 2011-01-17 H323 SIP BRI M5T SIP Stack/4.0.30.30
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.2.10 : 5060 (NAT)

<--- Transmitting (NAT) to 192.168.2.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK1d209edca508c8b97;received=192.168.2.10
From: <sip:[email protected]>;tag=ff9fffe939
To: <sip:[email protected]>
Call-ID: 53ef1dd932879ed4
CSeq: 14013 REGISTER
Server: FPBX-2.8.1(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (NAT) to 192.168.2.10:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK1d209edca508c8b97;received=192.168.2.10
From: <sip:[email protected]>;tag=ff9fffe939
To: <sip:[email protected]>;tag=as4061bcd9
Call-ID: 53ef1dd932879ed4
CSeq: 14013 REGISTER
Server: FPBX-2.8.1(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7249932d"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '53ef1dd932879ed4' in 32000 ms (Method: REGISTER)
Scheduling destruction of SIP dialog '53ef1dd932879ed4' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.2.10:5060 --->
REGISTER sip:192.168.2.240:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bKadee578764d785155
Max-Forwards: 70
From: <sip:[email protected]>;tag=ff9fffe939
To: <sip:[email protected]>
Call-ID: 53ef1dd932879ed4
CSeq: 14014 REGISTER
Authorization: Digest username="isdn",realm="asterisk",nonce="7249932d",uri="sip:192.168.2.240:5060",response="62427845d8dbc914ff737bffeeaea9e3",algorithm=MD5
Contact: <sip:[email protected]:5060>
Expires: 60
User-Agent: Patton SN4638 5BIS 00A0BA06191F R5.7 2011-01-17 H323 SIP BRI M5T SIP Stack/4.0.30.30
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.2.10 : 5060 (NAT)

<--- Transmitting (NAT) to 192.168.2.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bKadee578764d785155;received=192.168.2.10
From: <sip:[email protected]>;tag=ff9fffe939
To: <sip:[email protected]>
Call-ID: 53ef1dd932879ed4
CSeq: 14014 REGISTER
Server: FPBX-2.8.1(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (NAT) to 192.168.2.10:5060 --->
SIP/2.0 403 Forbidden (Bad auth)
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bKadee578764d785155;received=192.168.2.10
From: <sip:[email protected]>;tag=ff9fffe939
To: <sip:[email protected]>;tag=as4061bcd9
Call-ID: 53ef1dd932879ed4
CSeq: 14014 REGISTER
Server: FPBX-2.8.1(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

könnt Ihr Profis mal so nett sein und drüberschaun wo ich nen Schmarn drinn hab
Ich dank euch schon mal

mfg
Markus Scherer

foschi · 16 Mrz 2011

Ohne dass ich Deinem Thread-Titel jetzt direkt zustimme: vielleicht magst Du uns überhaupt erst einmal schildern wo Dein Problem liegt?

easyhugo · 17 Mrz 2011

OK
vor lauter Aufregung vergessen zuviel geschrieben

Ich kann zwar Telefonieren aber wenn mann im Sipdebug schaut gibt da ein

<--- Transmitting (NAT) to 192.168.2.10:5060 --->
SIP/2.0 403 Forbidden (Bad auth)
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bKadee578764d785155;received=192.168.2.10
From: <sip:[email protected]>;tag=ff9fffe939
To: <sip:[email protected]>;tag=as4061bcd9
Call-ID: 53ef1dd932879ed4
CSeq: 14014 REGISTER
Server: FPBX-2.8.1(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

vielleicht ist ja auch nur ein Schönheitsfehler

Das nächste ist die sipconf da sollte doch bei host ein dynamic rein nur ich brauch da die ip vom patton sonst geht nix wie schon gesagt vielleicht ist es nur ein Schönheitsfehler

darum die Bitte an euch ob sich jemand mal die sachen anschauen könnte und darum auch die ganzen confs und log

thx max

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Bin zu blöd für smartnode 4638 und asterisk

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