[Gelöst] Asterisk 1.6 + SN4638: Erste Gesprächs Sekunden fehlen bei abgehenden Gesprächen

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tb87729

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Hallo *,

wir haben nun in der Firma komplett auf VoIP Telefonie auf Basis SNOM 370/360 mit Asterisk 1.6 umgestellt. Die ISDN Anbindung läuft über Patton 4638. Wir haben an jedem unserer Standorte so eine Kombination stehen, damit wir auch im Falle eines Internen Netzausfalles noch telefonieren können. Nun ist einigen Mitarbeiter aufgefallen, dass die ersten 1 - 2 Sekunden bei ausgehenden Gesprächen fehlen - d.h. vom angerufenen Teilnehmer hört man u.U. den Namen nicht mehr. Um dem Problem auf die Spur zu kommen, habe ich im Asterisk eine Testrufnummer eingerichtet, die nach einem "Answer" direkt mit SayDigits(12345) anfängt zu zählen. Rufe ich diese Nummer von einem normalen ISDN Anschluss aus an, so höre ich eine "1 2 3 4 5". Wenn ich diese Nummer über die Patton anrufe, so höre ich nur ein "3 4 5". Der Weg geht dann also zweimal über die Pattons:
Snom 370 -> Patton A (Standort 1) -> ISDN Anlagenschluss -> Patton B (Standort2) -> Asterisk. Da ich den "Eingangsweg" per normalen ISDN Telefon ausgetestet habe, gehe ich davon aus, dass das Problem beim ausgehenden Asterisk besteht.

Ich habe den Netzwerkverkehr auf beiden Asterisk Server mitgeschnitten und mir den RTP Strom angehört (Die Pattons sind über ein dediziertes Netzwerk mit dem Asterisk Server verbunden / Cross Over). Der Asterisk Server an Patton B sendet den vollständigen RTP Strom an die Patton ("1 2 3 4 5"). Am Asterisk Server mit der Patton A bekomme ich aber nur noch per RTP "3 4 5" zu hören. Damit schließe ich ein Einfluss der Snom Telefone aus (mit einem Softphone konnte ich das Problem auch nachstellen).

Testweise habe ich einen PC mit astlinux aufgebaut und eine HFC-ISDN Karte eingebaut. Die Sprachqualität ist zwar schlechter, aber in dieser Konfiguration ist der Sprachstrom immer komplett zu hören. Allerdings musste ich im Dialplan vor dem Dial(misdn) noch ein Answer setzen, ansonsten fehlte auch dort der Gesprächsanfang. Der Answer vor dem Dial bei den Patton Umgebungen macht allerdings keinen Unterschied.
Irgendwelche Ideen, wo die erste Sekunde bleibt ? Wenn jemand meine Testnummer ausprobieren möchte: 069 981 955 309. Wäre interessant, ob ihr mit ähnlicher Konfig die volle Ansage hört.

Gruß
Thore


Hier mein Patton Config:
Code: 
#----------------------------------------------------------------#
#                                                                #
# SN4638/5BIS                                                    #
# R5.6 2011-01-17 H323 SIP BRI                                   #
# 2011-02-22T10:25:53                                            #
# SN/00A0BA048B18                                                #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
administrator admin password xxxxxxxxxxxxxxxx== encrypted
administrator administrator password xxxxxxxxxxxxxxxxx== encrypted
clock local default-offset +02:00
dns-client server 192.168.10.2
webserver port 80 language en
sntp-client
sntp-client server primary 85.214.108.169 port 123 version 4
sntp-client poll-interval 5
sntp-client local-clock-offset

syslog-client

  remote 10.10.12.11 514
    facility kernel severity debug
    facility user-level severity debug
    facility daemon severity debug

configure
system hostname brevoipgw01

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 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 aaa default
  method 1 local
  method 2 none

context ip router

  interface WAN
    ipaddress 192.168.50.5 255.255.255.0
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

  interface LAN
    ipaddress 10.10.12.2 255.255.255.0
    icmp redirect accept
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

  dyndns
    authentication none RZR+7jHDsNg= encrypted
    service custom

context ip router
  route 0.0.0.0 0.0.0.0 192.168.50.1 0

context cs switch
  digit-collection timeout 3
  digit-collection full-match set-address-complete-indication
  national-prefix 0
  international-prefix 00

  routing-table called-e164 RT_FROM_PSTN
    route .T dest-interface ast-oraise

  interface isdn BRI00
    route call dest-interface ast-oraise

  interface isdn IF_TE_00
    route call dest-table RT_FROM_PSTN

  interface isdn IF_TE_01
    route call dest-table RT_FROM_PSTN

  interface isdn IF_TE_02
    route call dest-table RT_FROM_PSTN

  interface isdn IF_TE_03
    route call dest-table RT_FROM_PSTN

  interface isdn IF_NT_04
    route call dest-service SER_HUNT_PSTN

  interface sip ast-oraise
    bind context sip-gateway asterisk-gw
    route call dest-service SER_HUNT_PSTN
    remote 10.10.12.11
    early-disconnect

  service sip-location-service SER_SIP_LOCATION
    bind location-service SER_LOCATION

  service hunt-group SER_HUNT_PSTN
    cyclic
    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_TE_00
    route call 2 dest-interface IF_TE_01
    route call 3 dest-interface IF_TE_02

context cs switch
  no shutdown

authentication-service oraise
  realm 1 10.10.12.11
  username xxxxx password xxxxxxxxxxxxxxxxxx== encrypted

location-service SER_LOCATION
  domain 1 1

  identity oraise
    alias name oraise

    authentication inbound

context sip-gateway asterisk-gw

  interface INBOUND
    bind interface LAN context router port 5060

context sip-gateway asterisk-gw
  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 user
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_TE_00 switch

port bri 0 0
  no shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side user
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_TE_01 switch

port bri 0 1
  no shutdown

port bri 0 2
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side user
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_TE_02 switch

port bri 0 2
  no shutdown

port bri 0 3
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side user
    encapsulation q931

    q931
      protocol dss1
      uni-side net
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_TE_03 switch

port bri 0 3
  shutdown

port bri 0 4
  clock auto
  power-feed
  encapsulation q921

  q921
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side net
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_NT_04 switch

port bri 0 4
  no shutdown
 
Zuletzt bearbeitet:
Update

Hallo zusammen,

habe gerade mal ein SNOM Telefon direkt an der Patton registriert. Und auch dort fehlt die erste Sekunde in der Ansage.....

Gruß
Thore
 
Hallo!

Versuch mal an den ISDN-Interfaces:

Code: 
interface isdn ISDN_0x
     route call dest-table RT_FROM_PSTN
     isdn-date-time
     inband-info accept force call-setup call-proceeding

"isdn-date-time" würd ich auch setzten auch wenn du es in deinem Fall (Asterisk) nicht benötigen wirst.

jwm
 
Hallo jwm,
macht keinen Unterschied - leider ....
 
Hallo!

Hast du es auf allen ISDN Interfaces aktiviert?
Ich hatte mal das selbe Problem was dadurch gelöst war. Hab jedoch an den Geräten noch die SW. 4.2 laufen ev. heisst der Befehl jetzt irgendwie anders.

Weiters hab ich auf dem SIP interface noch 3 Befehle:

early-disconnect
no call-transfer accept
no call-transfer emit

Vielleicht hilft das etwas.
Anbei auch noch ein Konfigauszug.

Code: 
 interface isdn ISDN_01
    route call dest-table RT_SIP
    isdn-date-time
    inband-info accept force call-setup call-proceeding

 interface sip SIP
    bind gateway GW_SIP
    service default
    route call dest-table RT_ISDN
    remote 192.168.xx.xxx
    early-disconnect
    no call-transfer accept
    no call-transfer emit

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 ISDN_01 switch

jwm
 
Hallo,

early-disconnect
habe ich schon in der Konfig, die anderen beiden werde ich am Wochende testen. Ich baue im Moment noch eine weitere Testumgebung mit eigener Patton auf, damit ich in Ruhe an der Parametern drehen kann und nicht Gefahr laufe, die "normalen" Gespräche zu beeinflussen.

Thore
 
Hallo
Eine tiefe analyse währe hier sicher nützlich.
Dazu helfen die folgenden Monitore:

Code: 
show version
show running-config
debug context sip-gateway transport detail 5
debug context sip-gateway datapath
debug ccisdn signaling
debug ccisdn datapath
debug media-gateway all

Der Ausdruck sieht dann in etwa folgendermassen aus:

Code: 
KUSI_DESK(sip-if)[IF_SIP_~]#
KUSI_DESK(sip-if)[IF_SIP_~]#14:09:30  ICC   > [IF_ISDN_S00] << Message: primitive=64
14:09:30  ICC   > [IF_ISDN_S00] Added endpoint IF_ISDN_S00-009fe800
14:09:30  ICC   > [IF_ISDN_S00] NEW CALL. Allocated Endpoint IF_ISDN_S00-009fe800
14:09:30  ICC   > [EP IF_ISDN_S00-009fe800] << [080005]
SETUP (DSS1 Ntwk)
  [04038090A3]
  Bearer capability : speech - CCITT
    circuit mode - 64kBit/s - G.711 A-law
  [6C0800A0353535323030]
  Calling party number : 1
    unknown number - unknown numbering plan
    presentation restricted - user provided not screened
  [700481353438]
  Called party number : 2
    unknown number - E.164 numbering plan
  [7D029181]
  High layer compatibility : telephony
    CCITT

14:09:30  DP    > TDM-00/00/00: registerEventCallback(0xc92254)
14:09:30  DP    > TDM-00/00/00: configure(Pstn-Config)
14:09:30  IDP   > [EP IF_ISDN_S00-009fe800] Change dtmf detection to 'disabled'
14:09:30  DP    > TDM-00/00/00: The following config package could not be converted:
SEQUENCE {
  packageId UUIDpackageId UUID = CfgToneDetection
  argument SEQUENCE {
    profile GeneralString IMPLICIT [CONTEXT 1] = 'default' OPTIONAL
    detectDtmf BOOLEAN IMPLICIT [CONTEXT 2] = false OPTIONAL
  }
}
14:09:30  IDP   > [EP IF_ISDN_S00-009fe800] Acquired b-Channel 0 (Datapath termination 1000000)
14:09:30  ICC   > [EP IF_ISDN_S00-009fe800] State: NULL, Event: TERMINAL SETUP IND
14:09:30  ICC   > IsdnEndpointDss1Net(0x9fe800)::actionDial. Subscriber number:1
14:09:30  ICC   > IsdnEndpointDss1Net(0x9fe800)::actionDial. Setting mwi by stutter dialtone for 1 to disabled (New:0 Old:0 NewUrgent:0, OldUrgent:0)
14:09:30  ICC   > [EP IF_ISDN_S00-009fe800] set call key: 8
14:09:30  ICC   > [EP IF_ISDN_S00-009fe800] AOC-S Net << Call Setup
14:09:30  ICC   > [EP IF_ISDN_S00-009fe800] AOC-S Net: Set state to AOC-S Activated
14:09:30  ICC   > [EP IF_ISDN_S00-009fe800] AOC-D Net << Call Setup
14:09:30  ICC   > [EP IF_ISDN_S00-009fe800] AOC-E Net << Call Setup
14:09:30  ICC   > [EP IF_ISDN_S00-009fe800] AOC-E Net: Set state to AOC-E Activated
14:09:30  ICC   > IsdnEndpointDss1Net(0x9fe800)::ActionDial: Stutter=0 dwNewMessages=0
14:09:30  ICC   > [EP IF_ISDN_S00-009fe800] Set state to OVERLAP SENDING
14:09:30  ICC   > IsdnEndpointDss1Net(0x9fe800)::UpdateDatapath peerProvidesData: disabled
14:09:30  IDP   > [EP IF_ISDN_S00-009fe800] Change datapath direction to Send Only.
14:09:30  DP    > TDM-00/00/00: setOperationMode(send-only)
14:09:30  TDM   > TDM-00/00/00: Acquiring DSP resource.
14:09:30  TDM   > TDM-00/00/00: Reconfigure resource.
14:09:30  TDM   > [DSP 0xe5a328] Received RTP Media Config.
14:09:30  TDM   > [DSP 0xe5a328] voice config.
14:09:30  TDM   > [DSP 0xe5a328] dejitter config.
14:09:30  TDM   > [DSP 0xe5a328] pstn config.
14:09:30  TDM   > [DSP 0xe5a328] caller-id config (None).
14:09:30  TDM   > [DSP 0xe5a328] Admin operation mode inactive -> send-only
14:09:30  TDM   > [DSP 0xe5a328] Dsp operation mode inactive -> send-only
14:09:30  TDM   > [DSP 0xe5a328]: Connected port 0 bChannel 0 to DSP timeslot 0
14:09:30  DSP   > [Port 3] State=closed: Event=openVoice
14:09:30  DSP   > [Port 3] Set tx plugin to voice.
14:09:30  DSP   > [Port 3] Opening
14:09:30  DSP   > [Port 3 (dsp 0, channel 0)] Configured for G.711aLaw on timeslot 0:
    Silence compression OFF
    Echo canceller ON (NLPM Adaptive, HybridLoss 6)
    Post filter: ON, HighPass filter: ON
    Output gain: 0dB
    Input gain:  0dB
    Transmission: Fax None / Modem: None
    Max bit rate (relay): Fax 14400 bit/s / Modem 9600 bit/s
    Fax/modem gain (relay): -9.5dB
    Fax/modem bypass codec: G.711aLaw
    Modem Dejitter buffer size: 200ms
    Fax Dejitter buffer size: 200ms
    T.38 Error correction: ON
    T.38 HDLC image tx: ON
    Fax protocol mode: T.38 UDP
    DTMF signal gain: lf -2dB, hf -1dB, mute encoder OFF
    Fax detection forced: OFF
    Caller-ID: Disabled

14:09:30  DSP   > [Channel 0 0] ACTIVATING CHANNEL
14:09:30  DSP   > [Port 3] Set tx media type 2.
14:09:30  DSP   > [Port 3] Set rx media type 2.
14:09:30  DSP   > [Port 3] New state=voice
14:09:30  TDM   > [DSP 0xe5a328] Activate VOICE dejitter configuration
14:09:30  Dejit > Input length: 80
14:09:30  Dejit > [0xe5a600] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

14:09:30  Dejit > Input length: 80
14:09:30  Dejit > [0xe5aab0] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

14:09:30  Dejit > Input length: 80
14:09:30  Dejit > [0xe5ad70] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

14:09:30  TDM   > [DSP 0xe5a328] Activate VOICE tx-buffer configuration
14:09:30  TDM   > [DSP 0xe5a328] Activating TX.
14:09:30  DSP   > Scheduler: added source processor 0xe5a77c
14:09:30  ICC   > [EP IF_ISDN_S00-009fe800] >> [08000D]
SETUP ACKNOWLEDGEMENT (DSS1 Ntwk)

14:09:30  DP    > TDM-00/00/00: addToContext(00000021)
14:09:30  ICC   > [EP IF_ISDN_S00-009fe800] State: OVERLAP SENDING, Event: PEER TRYING
14:09:30  ICC   > IsdnEndpointDss1Net(0x9fe800)::UpdateDatapath peerProvidesData: disabled
14:09:30  DSP   > [Channel 0 0] FIRST_PACKET_RECEIVED
14:09:30  ICC   > IsdnEndpointDss1Net(0x9fe800)::UpdateDatapath peerProvidesData: disabled
14:09:35  ICC   > [EP IF_ISDN_S00-009fe800] State: OVERLAP SENDING, Event: PEER TRYING
14:09:35  ICC   > IsdnEndpointDss1Net(0x9fe800)::UpdateDatapath peerProvidesData: disabled
14:09:35  SIP_TR> [STACK] > Stack: to 172.16.100.10
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.16.100.5:5060;branch=z9hG4bK5c8051fb0e3472df6
Max-Forwards: 70
From: <sip:[email protected]:5060>;tag=ec4a7644a2
To: <sip:[email protected]>
Call-ID: b4ce047c7b58faaf
CSeq: 31693 INVITE
Contact: <sip:[email protected]:5060>
Supported: replaces
User-Agent: Patton SN4634 3BIS UI 00A0BA021563 R5.3 2009-03-18 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Type: application/sdp
Content-Length: 341

v=0
o=MxSIP 0 1 IN IP4 172.16.100.5
s=SIP Call
c=IN IP4 172.16.100.5
t=0 0
m=audio 4864 RTP/AVP 8 2 0 18 101 100
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-16
a=fmtp:100 192-194
a=sendrecv

14:09:35  SIP_TR> [STACK] < Stack: from 172.16.100.10
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.16.100.5:5060;branch=z9hG4bK5c8051fb0e3472df6;received=172.16.100.10
From: <sip:[email protected]:5060>;tag=ec4a7644a2
To: <sip:[email protected]>;tag=as5a51b910
Call-ID: b4ce047c7b58faaf
CSeq: 31693 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="172.16.100.10", nonce="01cad78a"
Content-Length: 0


14:09:35  SIP_TR> [STACK] > Stack: to 172.16.100.10
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.16.100.5:5060;branch=z9hG4bK5c8051fb0e3472df6
Max-Forwards: 70
From: <sip:[email protected]:5060>;tag=ec4a7644a2
To: <sip:[email protected]>;tag=as5a51b910
Call-ID: b4ce047c7b58faaf
CSeq: 31693 ACK
User-Agent: Patton SN4634 3BIS UI 00A0BA021563 R5.3 2009-03-18 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


14:09:35  SIP_TR> [STACK] > Stack: to 172.16.100.10
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.16.100.5:5060;branch=z9hG4bK89edcaeb6d0af5f61
Proxy-Authorization: Digest username="1",realm="172.16.100.10.com",nonce="01cad78a",uri="sip:[email protected]",response="3823dd888f8b75fa2ca139c3fc5126e7",algorithm=MD5
Max-Forwards: 70
From: <sip:[email protected]:5060>;tag=ec4a7644a2
To: <sip:[email protected]>
Call-ID: b4ce047c7b58faaf
CSeq: 31694 INVITE
Contact: <sip:[email protected]:5060>
Supported: replaces
User-Agent: Patton SN4634 3BIS UI 00A0BA021563 R5.3 2009-03-18 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Type: application/sdp
Content-Length: 341

v=0
o=MxSIP 0 1 IN IP4 172.16.100.5
s=SIP Call
c=IN IP4 172.16.100.5
t=0 0
m=audio 4864 RTP/AVP 8 2 0 18 101 100
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-16
a=fmtp:100 192-194
a=sendrecv

14:09:35  SIP_TR> [STACK] < Stack: from 172.16.100.10
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.100.5:5060;branch=z9hG4bK89edcaeb6d0af5f61;received=172.16.100.10
From: <sip:[email protected]:5060>;tag=ec4a7644a2
To: <sip:[email protected]>
Call-ID: b4ce047c7b58faaf
CSeq: 31694 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


14:09:35  ICC   > [EP IF_ISDN_S00-009fe800] State: OVERLAP SENDING, Event: PEER PROCEEDING
14:09:35  ICC   > [EP IF_ISDN_S00-009fe800] Set state to OUTGOING PROCEEDING
14:09:35  ICC   > IsdnEndpointDss1Net(0x9fe800)::UpdateDatapath peerProvidesData: disabled
14:09:35  IDP   > [EP IF_ISDN_S00-009fe800] Change dtmf detection to 'enabled'
14:09:35  DP    > TDM-00/00/00: The following config package could not be converted:
SEQUENCE {
  packageId UUIDpackageId UUID = CfgToneDetection
  argument SEQUENCE {
    profile GeneralString IMPLICIT [CONTEXT 1] = 'default' OPTIONAL
    detectDtmf BOOLEAN IMPLICIT [CONTEXT 2] = true OPTIONAL
  }
}
14:09:35  IDP   > [EP IF_ISDN_S00-009fe800] Change datapath direction to Send/Receive.
14:09:35  DP    > TDM-00/00/00: setOperationMode(send/receive)
14:09:35  TDM   > [DSP 0xe5a328] Admin operation mode send-only -> send/receive
14:09:35  TDM   > [DSP 0xe5a328] Dsp operation mode send-only -> send/receive
14:09:35  TDM   > [DSP 0xe5a328] Activating RX.
14:09:35  TDM   > [DSP 0xe5a328] Reconfiguring tx-buffer.
14:09:35  TDM   > [DSP 0xe5a328] Activate VOICE tx-buffer configuration
14:09:35  ICC   > [EP IF_ISDN_S00-009fe800] >> [080002]
CALL PROCEEDING (DSS1 Ntwk)
  [1E028182]
  Progress indicator : destination address is non-ISDN
    private network serving local user - CCITT

14:09:35  SIP_TR> [STACK] < Stack: from 172.16.100.10
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.100.5:5060;branch=z9hG4bK89edcaeb6d0af5f61;received=172.16.100.10
From: <sip:[email protected]:5060>;tag=ec4a7644a2
To: <sip:[email protected]>;tag=as0cf81c40
Call-ID: b4ce047c7b58faaf
CSeq: 31694 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


14:09:35  ICC   > [EP IF_ISDN_S00-009fe800] State: OUTGOING PROCEEDING, Event: PEER ALERTING
14:09:35  ICC   > [EP IF_ISDN_S00-009fe800] Set state to CALL DELIVERED
14:09:35  ICC   > IsdnEndpointDss1Net(0x9fe800)::UpdateDatapath peerProvidesData: disabled
14:09:35  IDP   > [EP IF_ISDN_S00-009fe800] Play tone: ringback-tone
14:09:35  DP    > TDM-00/00/00: signal(Tones/Cadence-Signal)
14:09:35  ICC   > [EP IF_ISDN_S00-009fe800] >> [080001]
ALERTING (DSS1 Ntwk)
  [1E028188]
  Progress indicator : inband information available
    private network serving local user - CCITT

14:09:40  SIP_TR> [STACK] < Stack: from 172.16.100.10
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.100.5:5060;branch=z9hG4bK89edcaeb6d0af5f61;received=172.16.100.10
From: <sip:[email protected]:5060>;tag=ec4a7644a2
To: <sip:[email protected]>;tag=as0cf81c40
Call-ID: b4ce047c7b58faaf
CSeq: 31694 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 4042 4042 IN IP4 172.16.100.10
s=session
c=IN IP4 172.16.100.10
t=0 0
m=audio 10794 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

14:09:40  DP    > RTP-00/0022: registerEventCallback(0xa1165c)
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 DP] Using datapath termination 0x0200ffff
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 DP] Using VoIP profile:
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO] Configuring datapath termination: Voice
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Highpass Filter:           enabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Post Filter:               enabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Silence Suppression:       disabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Voice Update Frames:       disabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   DTMF Relay:                enabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Mute Encoder:              enabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Modem Transmission mode:   bypass
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax Transmission mode:     relay
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax Bypass Coder:          (undefined)
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Modem Bypass Coder:        G.711 A-law
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax Dejitter Max. Delay:   200
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Modem Dejitter Max. Delay: 200
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax Max. Bit Rate:         14400
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax Error Correction Mode: enabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax/Data HDLC:             enabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax Detection Mode:        cedTone
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax Bypass Method:         nse
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Modem Bypass Method:       nse
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax Volume:                14281512
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   CED Net Detection:         disabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   CED Net Detection Time:    0
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   CED Net Observation Time:  0
14:09:40  DP    > RTP-00/0022: configure(Voice-Config)
14:09:40  DP    > RTP-00/0022: event(General/ConfigChanged-Event)
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO] Configuring datapath termination: Dejitter
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Dejitter Mode:    adaptive
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Max. Delay:       120
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Max. Packet Loss: 4
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Shrink Speed:     1
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Grow Step:        1
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Grow Attenuation: 1
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Max. Delay:       120
14:09:40  DP    > RTP-00/0022: configure(Dejitter-Config)
14:09:40  DP    > RTP-00/0022: event(General/ConfigChanged-Event)
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 DP] Using tone-set profile 'default'
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 DP] Play tones: yes
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 DP] Local Media Address: 172.16.100.5
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 DP] Peer call-leg changed to state CONNECTED
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 DP] Add termination RTP-00/0022 to context 00000021
14:09:40  DP    > RTP-00/0022: addToContext(00000021)
14:09:40  DP    > RTP-00/0022: addConnector(ffffffff)
14:09:40  DP    > TDM-00/00/00: addConnector(ffffffff)
14:09:40  TDM   > [DSP 0xe5a328] Add connector notify. ID 0
14:09:40  DSP   > Scheduler: removed source processor 0xe5a77c
14:09:40  DSP   > Scheduler: added source processor 0xe5a77c
14:09:40  DP    > RTP-00/0022: getConfiguration(00010300)
14:09:40  DP    > RTP-00/0022: getConfiguration(00010b00)
14:09:40  DP    > TDM-00/00/00: configure(Voice-Config)
14:09:40  TDM   > [DSP 0xe5a328] voice config.
14:09:40  TDM   > [DSP 0xe5a328] reconfiguring DSP.
14:09:40  DSP   > [Port 3] State=voice: Event=close
14:09:40  DSP   > [Port 3] Set tx plugin to none.
14:09:40  DSP   > [Port 3] Set tx media type 2.
14:09:40  DSP   > [Port 3] Set rx media type 2.
14:09:40  DSP   > [Port 3] Closing (dsp 0, channel 0)
14:09:40  DSP   > [Channel 0 0] SENDING IDLE PACKET
14:09:40  DSP   > [Port 3] New state=closed
14:09:40  DSP   > [Port 3] State=closed: Event=openVoice
14:09:40  DSP   > [Port 3] Set tx plugin to voice.
14:09:40  DSP   > [Port 3] Opening
14:09:40  DSP   > [Port 3 (dsp 0, channel 0)] Configured for G.711aLaw on timeslot 0:
    Silence compression OFF
    Echo canceller ON (NLPM Adaptive, HybridLoss 6)
    Post filter: ON, HighPass filter: ON
    Output gain: 0dB
    Input gain:  0dB
    Transmission: Fax Relay / Modem: Bypass
    Max bit rate (relay): Fax 14400 bit/s / Modem 9600 bit/s
    Fax/modem gain (relay): -9.5dB
    Fax/modem bypass codec: G.711aLaw
    Modem Dejitter buffer size: 200ms
    Fax Dejitter buffer size: 200ms
    T.38 Error correction: ON
    T.38 HDLC image tx: ON
    Fax protocol mode: T.38 UDP
    DTMF signal gain: lf -2dB, hf -1dB, mute encoder ON
    Fax detection forced: OFF
    Caller-ID: Disabled

14:09:40  DSP   > [Port 3] Set tx media type 2.
14:09:40  DSP   > [Port 3] Set rx media type 2.
14:09:40  DSP   > [Port 3] New state=voice
14:09:40  TDM   > [DSP 0xe5a328] Activate VOICE dejitter configuration
14:09:40  Dejit > Input length: 80
14:09:40  Dejit > [0xe5a600] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

14:09:40  Dejit > Input length: 80
14:09:40  Dejit > [0xe5aab0] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

14:09:40  Dejit > Input length: 80
14:09:40  Dejit > [0xe5ad70] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

14:09:40  TDM   > [DSP 0xe5a328] Activate VOICE tx-buffer configuration
14:09:40  DP    > RTP-00/0022: getConfiguration(00010e00)
14:09:40  DP    > TDM-00/00/00: configure(Dejitter-Config)
14:09:40  TDM   > [DSP 0xe5a328] dejitter config.
14:09:40  TDM   > [DSP 0xe5a328] Reconfiguring dejitter.
14:09:40  TDM   > [DSP 0xe5a328] Activate VOICE dejitter configuration
14:09:40  Dejit > Input length: 80
14:09:40  Dejit > [0xe5a600] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 120
    max queue fill level: 12

14:09:40  Dejit > Input length: 80
14:09:40  Dejit > [0xe5aab0] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 120
    max queue fill level: 12

14:09:40  Dejit > Input length: 80
14:09:40  Dejit > [0xe5ad70] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 120
    max queue fill level: 12

14:09:40  DP    > RTP-00/0022: registerEventCallback(0xd84018)
14:09:40  DP    > TDM-00/00/00: registerEventCallback(0xd84018)
14:09:40  DSP   > [Channel 0 0] IDLE_PACKET_RECEIVED
14:09:40  DSP   > [Channel 0 0] ACTIVATING CHANNEL
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO] Configuring datapath termination: RTP
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Local Address:  172.16.100.5/4864
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Remote Address: 172.16.100.10/10794
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Codec:          G.711 u-law (20 ms)
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Media Type:     audio
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Payload:        Voice=0, SID=13, NTE-Local=101, NTE-Remote=101, NSE-Local=100, NSE-Remote=100
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   SSRC:           10514680
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   TOS:            0
14:09:40  DP    > RTP-00/0022: configure(RTP-Config)
14:09:40  RTP   > [02000022] Configure local source:  0xac106405/4864/10514680
14:09:40  RTP   > [02000022] Configure remote source: 0xd90bdcfa/10794/14806432
14:09:40  RTP   > [02000022] Next hop gateway is:     0x00000000
14:09:40  RTP   > [TERM 2000022] Config changed (codec=G.711 u-law | media=audio)
14:09:40  DP    > RTP-00/0022: event(General/ConfigChanged-Event)
14:09:40  DP    > RTP-00/0022: getConfiguration(00010300)
14:09:40  DP    > TDM-00/00/00: configure(MediaType/RTP-Config)
14:09:40  TDM   > TDM-00/00/00: Received Media Config.
14:09:40  TDM   > TDM-00/00/00: Resource can be re-used.
14:09:40  TDM   > [DSP 0xe5a328] Received RTP Media Config.
14:09:40  DSP   > [Port 3] Try codec update for G.711uLaw.
14:09:40  DSP   > [Port 3] State=voice: Event=close
14:09:40  DSP   > [Port 3] Set tx plugin to none.
14:09:40  DSP   > [Port 3] Set tx media type 2.
14:09:40  DSP   > [Port 3] Set rx media type 2.
14:09:40  DSP   > [Port 3] Closing (dsp 0, channel 0)
14:09:40  DSP   > [Channel 0 0] SENDING IDLE PACKET
14:09:40  DSP   > [Port 3] New state=closed
14:09:40  DSP   > [Port 3] State=closed: Event=openVoice
14:09:40  DSP   > [Port 3] Set tx plugin to voice.
14:09:40  DSP   > [Port 3] Opening
14:09:40  DSP   > [Port 3 (dsp 0, channel 0)] Configured for G.711uLaw on timeslot 0:
    Silence compression OFF
    Echo canceller ON (NLPM Adaptive, HybridLoss 6)
    Post filter: ON, HighPass filter: ON
    Output gain: 0dB
    Input gain:  0dB
    Transmission: Fax Relay / Modem: Bypass
    Max bit rate (relay): Fax 14400 bit/s / Modem 9600 bit/s
    Fax/modem gain (relay): -9.5dB
    Fax/modem bypass codec: G.711aLaw
    Modem Dejitter buffer size: 200ms
    Fax Dejitter buffer size: 200ms
    T.38 Error correction: ON
    T.38 HDLC image tx: ON
    Fax protocol mode: T.38 UDP
    DTMF signal gain: lf -2dB, hf -1dB, mute encoder ON
    Fax detection forced: OFF
    Caller-ID: Disabled

14:09:40  DSP   > [Port 3] Set tx media type 2.
14:09:40  DSP   > [Port 3] Set rx media type 2.
14:09:40  DSP   > [Port 3] New state=voice
14:09:40  TDM   > [DSP 0xe5a328] Activate VOICE dejitter configuration
14:09:40  Dejit > Input length: 80
14:09:40  Dejit > [0xe5a600] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 120
    max queue fill level: 12

14:09:40  Dejit > Input length: 80
14:09:40  Dejit > [0xe5aab0] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 120
    max queue fill level: 12

14:09:40  Dejit > Input length: 80
14:09:40  Dejit > [0xe5ad70] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 120
    max queue fill level: 12

14:09:40  TDM   > [DSP 0xe5a328] Activate VOICE tx-buffer configuration
14:09:40  TDM   > [DSP 0xe5a328] Reconfiguring dejitter.
14:09:40  TDM   > [DSP 0xe5a328] Activate VOICE dejitter configuration
14:09:40  Dejit > Input length: 80
14:09:40  Dejit > [0xe5a600] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 120
    max queue fill level: 12

14:09:40  Dejit > Input length: 80
14:09:40  Dejit > [0xe5aab0] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 120
    max queue fill level: 12

14:09:40  Dejit > Input length: 80
14:09:40  Dejit > [0xe5ad70] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 120
    max queue fill level: 12

14:09:40  TDM   > [DSP 0xe5a328] Reconfiguring tx-buffer.
14:09:40  TDM   > [DSP 0xe5a328] Activate VOICE tx-buffer configuration
14:09:40  DSP   > [Channel 0 0] IDLE_PACKET_RECEIVED
14:09:40  DSP   > [Channel 0 0] ACTIVATING CHANNEL
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO] Configuring datapath termination: Voice
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Highpass Filter:           enabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Post Filter:               enabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Silence Suppression:       disabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Voice Update Frames:       disabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   DTMF Relay:                enabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Mute Encoder:              enabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Modem Transmission mode:   bypass
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax Transmission mode:     relay
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax Bypass Coder:          (undefined)
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Modem Bypass Coder:        G.711 A-law
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax Dejitter Max. Delay:   200
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Modem Dejitter Max. Delay: 200
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax Max. Bit Rate:         14400
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax Error Correction Mode: enabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax/Data HDLC:             enabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax Detection Mode:        cedTone
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax Bypass Method:         nse
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Modem Bypass Method:       nse
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Fax Volume:                14280472
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   CED Net Detection:         disabled
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   CED Net Detection Time:    0
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   CED Net Observation Time:  0
14:09:40  DP    > RTP-00/0022: configure(Voice-Config)
14:09:40  DP    > RTP-00/0022: event(General/ConfigChanged-Event)
14:09:40  DP    > RTP-00/0022: getConfiguration(00010b00)
14:09:40  DP    > TDM-00/00/00: configure(Voice-Config)
14:09:40  TDM   > [DSP 0xe5a328] voice config.
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO] Datapath: Change direction to Send/Receive.
14:09:40  DP    > RTP-00/0022: setOperationMode(send/receive)
14:09:40  RTP   > [02000022] Set mode: INACTIVE -> TX/RX
14:09:40  DPMUX > Changed next processor of port 0x0022face: 0xe1f160
14:09:40  RTP   > [02000022] (BCD) Event=enable | New State=broken
14:09:40  TDM   > DATA_BUFF_TX: Update Connection. Processor: e1f220
14:09:40  DPMUX > Activating directpath port: 0x0022face
14:09:40  DPMUX >   Protocol:                 RTP
14:09:40  DPMUX >   Local transport address:  0xac106405/4864 (2 port(s))
14:09:40  DPMUX >   Remote transport address: 0xd90bdcfa/10794 (2 port(s))
14:09:40  DPMUX >   IP Interface:             3
14:09:40  RTP   > [02000022] Next hop gateway is:     0xac106001
14:09:40  NTE   > [00e1f638] Tx activation request.
14:09:40  DSP   > [Channel 0 0] FIRST_PACKET_RECEIVED
14:09:40  SIP_TR> [STACK] > Stack: to 172.16.100.10
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.16.100.5:5060;branch=z9hG4bKcf408eb5344f20d6d
Proxy-Authorization: Digest username="1",realm="172.16.100.10.com",nonce="01cad78a",uri="sip:[email protected]",response="3823dd888f8b75fa2ca139c3fc5126e7",algorithm=MD5
Max-Forwards: 70
From: <sip:[email protected]:5060>;tag=ec4a7644a2
To: <sip:[email protected]>;tag=as0cf81c40
Call-ID: b4ce047c7b58faaf
CSeq: 31694 ACK
User-Agent: Patton SN4634 3BIS UI 00A0BA021563 R5.3 2009-03-18 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


14:09:40  DP    > RTP-00/0022: event(Event/Mediatype-RTP/Connection-Established)
14:09:40  RTP   > [02000022] (BCD) Event=rx-rtp | New State=established
14:09:40  RTP   > [02000022] Rx first packet (seq=22598)
14:09:40  Dejit > Input length: 160
14:09:40  Dejit > [0xe5a600] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 120
    max queue fill level: 6

14:09:40  ICC   > [EP IF_ISDN_S00-009fe800] State: CALL DELIVERED, Event: PEER CONNECTED
14:09:40  ICC   > [EP IF_ISDN_S00-009fe800] Set state to ACTIVE
14:09:40  ICC   > IsdnEndpointDss1Net(0x9fe800)::UpdateDatapath peerProvidesData: enabled
14:09:40  IDP   > [EP IF_ISDN_S00-009fe800] Stop tone
14:09:40  DP    > TDM-00/00/00: signal(ToneStop-Signal)
14:09:40  ICC   > [EP IF_ISDN_S00-009fe800] AOC-S Net << Call Connected
14:09:40  ICC   > [EP IF_ISDN_S00-009fe800] >> [080007]
CONNECT (DSS1 Ntwk)
  [4C050180353438]
  Connected number : 2
    unknown number - E.164 numbering plan
    presentation allowed - user provided not screened

14:09:40  ICC   > IsdnEndpointDss1Net(0x9fe800)::UpdateDatapath peerProvidesData: enabled
14:09:40  ICC   > [EP IF_ISDN_S00-009fe800] State: ACTIVE, Event: PEER INBAND INFO
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 DP] Peer call-leg provides data
14:09:40  MEDIA > [Port 3] Codec=G711U | Media=VOICE | Ecan=ON | Vad=OFF
14:09:41  SIP_TR> [STACK] < Stack: from 172.16.100.10
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.100.10:5060;branch=z9hG4bK4ee45a7a;rport
From: "asterisk" <sip:[email protected]>;tag=as668479a8
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 02 Mar 2011 13:09:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


14:09:41  SIP_TR> [STACK] > Stack: to 172.16.100.10
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 172.16.100.10:5060;branch=z9hG4bK4ee45a7a;rport=5060;received=172.16.100.10
From: "asterisk" <sip:[email protected]>;tag=as668479a8
To: <sip:[email protected]:5060>;tag=2689162337
Call-ID: [email protected]
CSeq: 102 OPTIONS
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, INFO, UPDATE, REFER, REGISTER
Server: Patton SN4634 3BIS UI 00A0BA021563 R5.3 2009-03-18 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


14:09:44  RTP   > [02000022] RTCP TX (SR) Timestamp=-1690332536
14:09:44  RTP   > [02000022]  Local TX-Info: packets=190, octets=30400
14:09:44  RTP   > [02000022]  Local RX-Info: packets=189, octets=30240, lost=0, jitter=7, since last=0mS
14:09:44  DP    > RTP-00/0022: getStatistics(00020900)
14:09:44  SIP_TR> [STACK] < Stack: from 172.16.100.10
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.100.10:5060;branch=z9hG4bK03a79b04;rport
From: <sip:[email protected]>;tag=as0cf81c40
To: <sip:[email protected]:5060>;tag=ec4a7644a2
Call-ID: b4ce047c7b58faaf
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


14:09:44  SIP_TR> [STACK] > Stack: to 172.16.100.10
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.100.10:5060;branch=z9hG4bK03a79b04;rport=5060;received=172.16.100.10
From: <sip:[email protected]>;tag=as0cf81c40
To: <sip:[email protected]:5060>;tag=ec4a7644a2
Call-ID: b4ce047c7b58faaf
CSeq: 102 BYE
Server: Patton SN4634 3BIS UI 00A0BA021563 R5.3 2009-03-18 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


14:09:44  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 DP] Subtract termination RTP-00/0022 from context 00000021
14:09:44  DP    > RTP-00/0022: subtractFromContext(00000021)
14:09:44  DP    > TDM-00/00/00: unregisterEventCallback(0xd84018)
14:09:44  DP    > RTP-00/0022: unregisterEventCallback(0xd84018)
14:09:44  TDM   > DATA_BUFF_TX: Update Connection. Processor: 0
14:09:44  DP    > TDM-00/00/00: removeConnector(00000000)
14:09:44  TDM   > [DSP 0xe5a328] Remove connector notify. ID: 0
14:09:44  DSP   > Scheduler: removed source processor 0xe5a77c
14:09:44  DSP   > Scheduler: added source processor 0xe5a77c
14:09:44  DP    > RTP-00/0022: removeConnector(00000000)
14:09:44  ICC   > [EP IF_ISDN_S00-009fe800] State: ACTIVE, Event: PEER RELEASED
14:09:44  ICC   > [EP IF_ISDN_S00-009fe800] Set state to DISCONNECT INDICATION
14:09:44  ICC   > IsdnEndpointDss1Net(0x9fe800)::UpdateDatapath peerProvidesData: disabled
14:09:44  IDP   > [EP IF_ISDN_S00-009fe800] Change dtmf detection to 'disabled'
14:09:44  DP    > TDM-00/00/00: The following config package could not be converted:
SEQUENCE {
  packageId UUIDpackageId UUID = CfgToneDetection
  argument SEQUENCE {
    profile GeneralString IMPLICIT [CONTEXT 1] = 'default' OPTIONAL
    detectDtmf BOOLEAN IMPLICIT [CONTEXT 2] = false OPTIONAL
  }
}
14:09:44  IDP   > [EP IF_ISDN_S00-009fe800] Play tone: release-tone
14:09:44  DP    > TDM-00/00/00: signal(Tones/Cadence-Signal)
14:09:44  ICC   > [EP IF_ISDN_S00-009fe800] AOC-S Net << Call Clearing
14:09:44  ICC   > [EP IF_ISDN_S00-009fe800] AOC-S Net: Set state to AOC Idle
14:09:44  ICC   > [EP IF_ISDN_S00-009fe800] AOC-D Net << Call Clearing
14:09:44  ICC   > [EP IF_ISDN_S00-009fe800] AOC-E Net << Call Clearing
14:09:44  ICC   > [EP IF_ISDN_S00-009fe800] AOC-E Net >> AOCE Invoke
14:09:44  ICC   > [EP IF_ISDN_S00-009fe800] AOC-E Net: Set state to AOC Idle
14:09:44  ICC   > [EP IF_ISDN_S00-009fe800] >> [080045]
DISCONNECT (DSS1 Ntwk)
  [08028190]
  Cause : normal call clearing
    private network serving local user - CCITT - Q.931
  [1E028188]
  Progress indicator : inband information available
    private network serving local user - CCITT

14:09:44  ICC   > IsdnEndpointDss1Net(0x9fe800)::UpdateDatapath peerProvidesData: disabled
14:09:44  DP    > RTP-00/0022: unregisterEventCallback(0xa1165c)
14:09:44  DP    > RTP-00/0022: setOperationMode(inactive)
14:09:44  RTP   > [02000022] Set mode: TX/RX -> INACTIVE
14:09:44  DPMUX > Changed next processor of port 0x0022face: 0x0
14:09:44  RTP   > [02000022] (BCD) Event=disable | New State=disabled
14:09:44  DPMUX > Deactivating directpath port: 0x0022face
14:09:44  NTE   > [00e1f638] Tx reset request.
14:09:44  NTE   > Scheduler: Unregister element immediately 0x00e1f638
14:09:44  NTE   > Scheduler: Unregistered element: 00e1f638
14:09:44  RTP   > [TERM 2000022] Tx-State=idle | Tx-Event=reset
14:09:44  RTP   > [TERM 2000022] Rx-State=idle | Rx-Event=reset
14:09:44  RTP   > [TERM 2000022] Stop Rx Fiter-Timeout
14:09:44  ICC   > [IF_ISDN_S00] << Message: primitive=50
14:09:44  ICC   > [EP IF_ISDN_S00-009fe800] << [08004D]
RELEASE (DSS1 Ntwk)

14:09:44  IDP   > [EP IF_ISDN_S00-009fe800] Releasing b-Channel 0 (Datapath termination 1000000)
14:09:44  DP    > TDM-00/00/00: signal(ToneStop-Signal)
14:09:44  DP    > TDM-00/00/00: setOperationMode(inactive)
14:09:44  TDM   > [DSP 0xe5a328] Admin operation mode send/receive -> inactive
14:09:44  TDM   > [DSP 0xe5a328] Dsp operation mode send/receive -> inactive
14:09:44  TDM   > [DSP 0xe5a328] Deactivating TX.
14:09:44  DSP   > Scheduler: removed source processor 0xe5a77c
14:09:44  TDM   > [DSP 0xe5a328] Deactivating RX.
14:09:44  TDM   > DATA_BUFF_TX: Reset drops pending packet
14:09:44  DSP   > [Port 3] State=voice: Event=close
14:09:44  DSP   > [Port 3] Set tx plugin to none.
14:09:44  DSP   > [Port 3] Set tx media type 2.
14:09:44  DSP   > [Port 3] Set rx media type 2.
14:09:44  DSP   > [Port 3] Closing (dsp 0, channel 0)
14:09:44  DSP   > [Channel 0 0] SENDING IDLE PACKET
14:09:44  DSP   > [Port 3] New state=closed
14:09:44  TDM   > [DSP 0xe5a328] Disconnected port 0 bChannel 0 from DSP timeslot 0
14:09:44  TDM   > TDM-00/00/00 Releasing DSP resource.
14:09:44  DP    > TDM-00/00/00: subtractFromContext(00000021)
14:09:44  DP    > TDM-00/00/00: unregisterEventCallback(0xc92254)
14:09:44  ICC   > [EP IF_ISDN_S00-009fe800] State: DISCONNECT INDICATION, Event: TERMINAL RELEASE IND
14:09:44  ICC   > [EP IF_ISDN_S00-009fe800] Set state to NULL
14:09:45  ICC   > IsdnEndpointDss1Net(0x9fe800)::UpdateDatapath peerProvidesData: disabled
14:09:45  IDP   > [EP IF_ISDN_S00-009fe800] Stop tone
14:09:45  IDP   > [EP IF_ISDN_S00-009fe800] Change datapath direction to Inactive.
14:09:45  ICC   > [IF_ISDN_S00] CLEARING CALL IF_ISDN_S00-009fe800
14:09:45  ICC   > [IF_ISDN_S00] Removed endpoint IF_ISDN_S00-009fe800
14:09:45  ICC   > [IF_ISDN_S00] Destroying finished calls.
14:09:45  ICC   > [IF_ISDN_S00] Destroyed endpoint IF_ISDN_S00-009fe800
14:09:45  DSP   > [Channel 0 0] IDLE_PACKET_RECEIVED
14:10:02  SIP_TR> [STACK] > Stack: to 172.16.100.10

Wichtig sind die zeitlichen Verzögerungen zwischen der SIP-OK Nachricht, dem öffnen des Datenpfades, der SIP-ACK Nachricht sowie dem Connect in Richtung ISDN

Code: 
[B]OK Message:[/B]

14:09:40  SIP_TR> [STACK] < Stack: from 172.16.100.10
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.100.5:5060;branch=z9hG4bK89edcaeb6d0af5f61;received=172.16.100.10
From: <sip:[email protected]:5060>;tag=ec4a7644a2
To: <sip:[email protected]>;tag=as0cf81c40
Call-ID: b4ce047c7b58faaf
CSeq: 31694 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 4042 4042 IN IP4 172.16.100.10
s=session
c=IN IP4 172.16.100.10
t=0 0
m=audio 10794 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

[B]Öffnen Datenpfad[/B]

14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO] Configuring datapath termination: RTP
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Local Address:  172.16.100.5/4864
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Remote Address: 172.16.100.10/10794
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Codec:          G.711 u-law (20 ms)
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Media Type:     audio
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   Payload:        Voice=0, SID=13, NTE-Local=101, NTE-Remote=101, NSE-Local=100, NSE-Remote=100
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   SSRC:           10514680
14:09:40  SIP_DP> [EP IF_SIP_OUTGOING-00c987a0/0 AUDIO]   TOS:            0
14:09:40  DP    > RTP-00/0022: configure(RTP-Config)

[B]SIP-ACK Message: [/B]

14:09:40  SIP_TR> [STACK] > Stack: to 172.16.100.10
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.16.100.5:5060;branch=z9hG4bKcf408eb5344f20d6d
Proxy-Authorization: Digest username="1",realm="inalp.com",nonce="01cad78a",uri="sip:[email protected]",response="3823dd888f8b75fa2ca139c3fc5126e7",algorithm=MD5
Max-Forwards: 70
From: <sip:[email protected]:5060>;tag=ec4a7644a2
To: <sip:[email protected]>;tag=as0cf81c40
Call-ID: b4ce047c7b58faaf
CSeq: 31694 ACK
User-Agent: Patton SN4634 3BIS UI 00A0BA021563 R5.3 2009-03-18 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0

[B]ISDN Connect Message:[/B]

14:09:40  ICC   > [EP IF_ISDN_S00-009fe800] >> [080007]
CONNECT (DSS1 Ntwk)
  [4C050180353438]
  Connected number : 2
    unknown number - E.164 numbering plan
    presentation allowed - user provided not screened

In diesem Fall ist ersichtlich, dass:
1. alles innerhalb einer Sekunde Stadtfindet
2. Die SIP-Ack sowie die ISDN Connect Message erst nach dem öffnen des Datenpfades gesendet werden.

Beim Server handelt es sich um einen Asterisk, beim SmartNode um einen SN4634. Das Sip interface sowie der Gateway hat keine besondere Einstellung. (kein early connect oder disconnect.)

Code: 
context cs
 
 interface sip IF_SIP_OUTGOING
    bind context sip-gateway SIP_GW
    route call dest-interface IF_ISDN_S00
    remote 172.16.100.10
    address-translation outgoing-call from-header user-part fix 1 host-part call

context cs
    no shutdown

location-service SER_LOC_ASTERISK
  domain 1 172.16.100.10

  identity 1

    authentication outbound
      authenticate 1 authentication-service SER_AUTH_ASTERISK username 1

    registration outbound
      registrar 172.16.100.10
      register auto

context sip-gateway SIP_GW

  interface IF_SIP_GW
    bind interface LAN context router port 5060

context sip-gateway SIP_GW
  bind location-service SER_LOC_ASTERISK
  no shutdown

Hoffe diese Informationien helfen weiter.
Viele Grüsse
Kusi
 
Lösung

Hallo zusammen,

in der Zwischenzeit konnte eine Lösung gefunden werden. jwm war mit dem Parameter "inband-info accept force call-setup call-proceeding" schon recht nah dran. Allerdings habe ich dann im Forum diesen Artikel gefunden: http://www.ip-phone-forum.de/showthread.php?p=1252278 . Der Parameter "inband-info accept force call-setup setup" brachte dann den Erfolg.
Aber Vielen Dank für die Unterstützung !!
Thore
 
Hallo Thore,

tb87729 schrieb:
Der Parameter "inband-info accept force call-setup setup" brachte dann den Erfolg.
Zum Vergrößern anklicken....
ich kann auf meiner SmartNode 4638 keinen Unterschied mit und ohne 'inband-info accept force call-setup setup' feststellen. In beiden Fällen fehlt etwas weniger als die erste Sekunde des Gesprächs. Meine Test-Nebenstelle auf dem Asterisk-Server sieht so aus (sollte auch für Nicht-Asteriskbenutzer verständlich sein ;) ):
Code: 
15 => {
        Ringing();
        Wait(3);
        Answer();
        SayDigits(123456);
        Hangup();
}
Wenn ich Ringing() und Wait(3) entferne, so daß kein Call Progress im ISDN entsteht und wenn ich zweimal sehr kurz hintereinander anrufe, dann fehlt der Gesprächsanfang nicht. Ich interpretiere das so, daß es irgendeine Latenz beim Aufbau des Medienpfads im DSP gibt und der DSP beim zweiten Anruf den Kanal noch nicht wieder abgebaut hat. Ich habe bei Patton ein Support-Ticket dazu aufgemacht und warte im Moment noch auf Antwort.

Gruß
Henning
 
Hallo zusammen,

ich habe das Problem nochmal genauer untersucht und konnte es mit SmartWare 5.5 und 5.6 nachstellen. Der Fehler trat nur bei Anrufen von SIP zu ISDN auf, nicht umgekehrt bei ISDN zu SIP. Mit SmartWare 5.7 tritt der Fehler nicht mehr auf.

Gruß
Henning
 
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