Patton Asterisk Hold (gespräch halten)

cstux · 21 Mai 2007

Hi,

ich hab noch folgendes Problem! Telefonieren funktioniert einbandfrei.
Jedoch wenn ich ein Gespräch halten will. Wird die Verbindung am SIP Telefon abgebaut und der ISDN Teilnehmer verweilt für immer in der Warteschleife. Auch MUSICONHOLD on ist nicht zu hören.

Code:

    -- Executing [380962@DID_TEST:1] Goto("SIP/sipgate-b6f02a00", "default|1111|1") in new stack
    -- Goto (default,1111,1)
    -- Executing [1111@default:1] Macro("SIP/sipgate-b6f02a00", "stdexten|1111|SIP/1111") in new stack
    -- Executing [s@macro-stdexten:1] Dial("SIP/sipgate-b6f02a00", "SIP/1111|20") in new stack
    -- Called 1111>
    -- SIP/1111-0a497220 is ringing
    -- Call on SIP/1111-0a497220 left from hold
    -- SIP/1111-0a497220 answered SIP/sipgate-b6f02a00
    -- Native bridging SIP/sipgate-b6f02a00 and SIP/1111-0a497220
    -- Started music on hold, class 'default', on SIP/sipgate-b6f02a00
    -- Got SIP response 500 "Server Internal Error" back from 10.0.0.1
  == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/sipgate-b6f02a00' in macro 'stdexten'
  == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/sipgate-b6f02a00'
    -- Stopped music on hold on SIP/sipgate-b6f02a00

Code:

#----------------------------------------------------------------#
#                                                                #
# SN4634/3BIS/UI                                                 #
# R3.20 2006-11-17 H323 SIP BRI                                  #
# 1970-01-11T21:11:16                                            #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
dns-relay
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 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 voip VOIP-ASTERISK
  codec 1 g729 rx-length 20 tx-length 20
  codec 2 g711ulaw64k rx-length 20 tx-length 20
  codec 3 g711alaw64k rx-length 20 tx-length 20
  codec 4 g723-6k3 rx-length 30 tx-length 30
  dejitter-max-delay 200

profile pstn default

profile sip default

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 10.0.0.1 255.255.255.0
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

context ip router
  route 0.0.0.0 0.0.0.0 10.0.0.10 0

context cs switch
  national-prefix 00
  international-prefix 000

  routing-table called-e164 ISDN_2_SIP
    route "" dest-interface IF-ASTERISK

  interface isdn IF-PBX
    route call dest-table ISDN_2_SIP
    caller-name

  interface sip IF-ASTERISK
    bind gateway ASTERISK
    service default
    route call dest-interface IF-PBX
    early-connect
    early-disconnect
    use profile voip VOIP-ASTERISK

context cs switch
  no shutdown

gateway sip ASTERISK
  call-signaling-port 5070
  bind interface LAN router

  service default
    authentication sipgate password nx/MNIzoskpr9r/EvL457A== encrypted
    default-server 130.10.10.217 loose-router
    registration-lifetime 1800
    registrar 130.10.10.217 5070
    user sipgate display-name sipgate

gateway sip ASTERISK
  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
  power-feed

  q921
    permanent-layer2
    protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      encapsulation cc-isdn
      bind interface IF-PBX 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

port bri 0 1
  shutdown

port bri 0 2
  clock auto
  encapsulation q921

  q921
    protocol pmp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side net

port bri 0 2
  shutdown

Weiterhin bekomme ich noch folgenden Fehler:
Registration for '[email protected]' timed out, trying again (Attempt #44)

Hier noch die Ausgabe der Patton

Code:

10.0.0.1#10:24:14  SIP_TR> [GW] < Stack: INVITE sip:[email protected]:5060 SIP/2.0
10:24:14  SIP_TR> [GW] > Stack: SIP/2.0 100 Trying
10:24:14  SIP_SI> [EP IF-ASTERISK-0090c7f8 SES 0xe941b8] < Stack: ReINVITE
10:24:14  SIP_DP> [EP IF-ASTERISK-0090c7f8/0 AUDIO] Configuring datapath termination: RTP
10:24:14  SIP_DP> [EP IF-ASTERISK-0090c7f8/0 AUDIO]   Local Address:  10.0.0.1/4870
10:24:14  SIP_DP> [EP IF-ASTERISK-0090c7f8/0 AUDIO]   Remote Address: 130.10.10.217/17646
10:24:14  SIP_DP> [EP IF-ASTERISK-0090c7f8/0 AUDIO]   Codec:          G.711 A-law (20 ms)
10:24:14  SIP_DP> [EP IF-ASTERISK-0090c7f8/0 AUDIO]   Media Type:     audio
10:24:14  SIP_DP> [EP IF-ASTERISK-0090c7f8/0 AUDIO]   Payload:        Voice=8, SID=13, NTE-Local=101, NTE-Remote=101, NSE-L
SE-Remote=0
10:24:14  SIP_DP> [EP IF-ASTERISK-0090c7f8/0 AUDIO]   SSRC:           15285880
10:24:14  SIP_DP> [EP IF-ASTERISK-0090c7f8/0 AUDIO]   TOS:            0
10:24:14  SIP_SI> [EP IF-ASTERISK-0090c7f8 SES 0xe941b8] > Stack: 200 OK
10:24:14  SIP_TR> [GW] > Stack: SIP/2.0 200 OK
10:24:14  SIP_TR> [GW] < Stack: INVITE sip:[email protected]:5060 SIP/2.0
10:24:14  SIP_TR> [GW] > Stack: SIP/2.0 500 Server Internal Error
10:24:15  SIP_TR> [GW] < Stack: ACK sip:[email protected]:5060 SIP/2.0
10:24:15  SIP_TR> [GW] > Stack: SIP/2.0 200 OK
10:24:15  SIP_TR> [GW] > Stack: REGISTER sip:130.10.10.217:5060 SIP/2.0
10:24:15  SIP_TR> [GW] < Stack: SIP/2.0 404 Not found
10:24:15  SIP_SI> [EP IF-ASTERISK-0090c7f8] Connection Broken
10:24:16  SIP_TR> [GW] < Stack: REGISTER sip:10.0.0.1 SIP/2.0
10:24:16  SIP_TR> [GW] > Stack: SIP/2.0 200 OK
10:24:18  SIP_TR> [GW] > Stack: SIP/2.0 200 OK

Hat jemand Asterisk 1.4 und Patton schon am laufen wo die Weiterleitung von Gesprächen funktioniert?

Mit dem [SIP] Transfer Patch von Digium funtioniert es auch nocht nicht
http://bugs.digium.com/view.php?id=9305

Nur die Fehlermeldung hat sich geändert.

Code:

22:05:26  SIP_SI> [EP IF-ASTERISK-00963e60 SES 0x96d558] < Stack: ReINVITE
22:05:26  SIP_SI> [EP IF-ASTERISK-00963e60 SES 0x96d558] > Stack: 488 Not Acceptable Here
22:05:26  SIP_TR> [GW] > Stack: SIP/2.0 488 Not Acceptable Here
22:05:26  SIP_TR> [GW] < Stack: ACK sip:[email protected]:5070 SIP/2.0
22:05:26  SIP_TR> [GW] < Stack: BYE sip:[email protected]:5070 SIP/2.0
22:05:26  SIP_TR> [GW] > Stack: SIP/2.0 200 OK
22:05:26  SIP_SI> [EP IF-ASTERISK-00963e60 SES 0x96d558] < Stack: BYE (or CANCEL)
22:05:26  CC    > [EP IF-ASTERISK-00963e60/active] Drop call 0098cf08
22:05:26  CC    > [EP IF-ASTERISK-00963e60/active] Set call-leg property: Provides Data -> false
22:05:26  CC    > [EP IF-ASTERISK-00963e60/active] Set call-leg property: Cause -> Normal call clearing
22:05:26  CC    > [EP IF-ASTERISK-00963e60/active] Set call-leg property: State -> RELEASED
22:05:26  SIP_SI> [EP IF-ASTERISK-00963e60] Finished
22:05:26  CC    > [EP IF-PBX-0097e738/active] Set call-leg property: Cause -> Normal call clearing
22:05:27  CC    > [EP IF-PBX-0097e738/active] Drop call 0098cf08
22:05:27  CC    > [EP IF-PBX-0097e738/active] Set call-leg property: Provides Data -> false
22:05:27  CC    > [EP IF-PBX-0097e738/active] Set call-leg property: State -> RELEASED
22:05:29  SIP_TR> [GW] > Stack: REGISTER sip:10.0.0.1:5070 SIP/2.0

Edit Guard-X: Beiträge zusammengeführt.

fabian.hoppe · 21 Sep 2007

Das Problem ist bei Patton bekannt und sollte eigentlich mit der letzten Firmware behoben worden sein. Wie wir leidvoll feststellen mussten, ist dies leider nicht richtig, die Gespräche brechen immer noch ab.

Mir ist von anderer Seite aus gemeldet worden, dass der Verzicht auf ein Reinvite seitens des Pattons das Problem löst - mir hilft das aus netzwerkarchitektonischen Gründen leider nicht weiter; ich bin auf ein Reinvite angewiesen.

Ich bin mit Patton im Kontakt um den Fehler weiter zu untersuchen.

Cheers, Fabian

Tecia · 28 Sep 2007

I have the same problem, can you help me?

Code:

#----------------------------------------------------------------#
#                                                                #
# SN4552/2BIS/EUI                                                #
# R3.20 2006-11-17 SIP                                           #
# 1970-01-07T01:57:32                                            #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
gui type basic
administrator admin password MtMEIhgUMpET5CJB7J3JQA== encrypted
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary 129.132.2.21 port 123 version 4

system

  ic voice 0

profile ppp default

profile call-progress-tone IT_Dialtone
  play 1 200 425 -12
  pause 2 200
  play 3 600 425 -12
  pause 4 1000

profile call-progress-tone IT_Alertingtone
  play 1 1000 425 -12
  pause 2 4000

profile call-progress-tone IT_Busytone
  play 1 500 425 -12
  pause 2 500

profile tone-set default

profile tone-set IT
  map call-progress-tone dial-tone IT_Dialtone
  map call-progress-tone ringback-tone IT_Alertingtone
  map call-progress-tone busy-tone IT_Busytone
  map call-progress-tone release-tone IT_Busytone
  map call-progress-tone congestion-tone IT_Busytone

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

profile voip NBLADE
  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

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

profile pstn default

profile sip default

profile sip nblade

profile dhcp-server DHCPS_LAN
  network 192.168.0.151 255.255.255.0
  lease 2 hours
  default-router 1 192.168.0.1
  domain-name patton.com
  domain-name-server 1 192.168.0.1

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface IF_IP_WAN
    ipaddress dhcp
    no napt-inside

  interface IF_IP_LAN
    ipaddress 192.168.0.151 255.255.255.0
    icmp router-discovery

context cs switch
  digit-collection timeout 4
  national-prefix 0
  international-prefix 00

  routing-table called-e164 RT_2_ISDN
    route .%T dest-service HUNTING MT_ITC

  routing-table called-e164 RT_ISDN_2_SIP
    route 99[1-9].T3 dest-interface IF_S0_PSTN
    route 0[1-9].T3 dest-service SER_HUNT_OUT
    route 00[1-9].T3 dest-service SER_HUNT_OUT
    route default dest-service SER_HUNT_OUT

  mapping-table itc to itc MT_ITC
    map default to speech

  interface isdn IF_S0_PSTN
    route call dest-interface IF_S0_PHONE
    dtmf-dialing

  interface isdn IF_S0_PHONE1
    route call dest-table RT_ISDN_2_SIP
    dtmf-dialing
    use profile tone-set IT

  interface isdn IF_S0_PHONE2
    route call dest-table RT_ISDN_2_SIP
    dtmf-dialing
    use profile tone-set IT

  interface isdn IF_S0_PHONE3
    route call dest-table RT_ISDN_2_SIP
    dtmf-dialing
    use profile tone-set IT

  interface isdn IF_S0_PHONE4
    route call dest-table RT_ISDN_2_SIP
    dtmf-dialing
    use profile tone-set IT

  interface isdn IF_S0_01

  interface sip IF_SIP_NBLADE
    bind gateway GW-NBLADE
    service default
    route call dest-table RT_2_ISDN
    early-disconnect
    remote-party-id called-party
    remote-party-id calling-party
    address-translation outgoing-call from-header user-part fix 210 host-part call
    use profile voip NBLADE

  interface sip IF_SIP_SERVICE
    service default
    address-translation outgoing-call from-header user-part fix patton host-part call

  service hunt-group SER_HUNT_OUT
    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
    route call 1 dest-interface IF_SIP_NBLADE
    route call 2 dest-interface IF_S0_PSTN

  service hunt-group HUNTING
    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 user-busy
    route call 1 dest-interface IF_S0_PHONE1
    route call 2 dest-interface IF_S0_PHONE2
    route call 3 dest-interface IF_S0_PHONE3
    route call 4 dest-interface IF_S0_PHONE4

  service hunt-group SER_HG_PSTN_FALLBACK
    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_SIP_SERVICE

context cs switch
  no shutdown

gateway sip GW-NBLADE
  bind interface IF_IP_LAN router

  service default
    domain 192.168.0.150
    realm 192.168.0.150
    authentication patton password MtMEIhgUMpET5CJB7J3JQA== encrypted
    default-server 192.168.0.150 5060 loose-router
    registrar 192.168.0.150 5060 use-default-server
    user patton
    session-timer 1800

gateway sip GW-NBLADE
  no shutdown

gateway sip GW_SIP
  call-signaling-port 5062

  service default
    authentication patton password MtMEIhgUMpET5CJB7J3JQA== encrypted default
    contact-address nat-address
    default-server 192.168.0.150 loose-router
    registrar 192.168.0.150
    user patton display-name patton

gateway sip GW_SIP
  no shutdown

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

port ethernet 0 1
  encapsulation ip
  bind interface IF_IP_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
      encapsulation cc-isdn
      bind interface IF_S0_PHONE1 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_PSTN switch

port bri 0 1
  no shutdown

my firmware is 3.21

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Patton Asterisk Hold (gespräch halten)

cstux

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Tecia

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