Inalp Patton Gateway -> t38modem -> Hylafax?

mhamann · 25 Okt 2007

Hallo werte Smortnode Profis,

ich versuche seit einiger Zeit die t38 Funktionen der Smartnode Gateways von Patton (hier im test ein SN4638 mit R4.2 2007-09-19 H323 SIP BRI) zu nutzen.

Ich möchte dabei Fax über T38 zusammen mit t38modem und Hylafax realisieren.

Dafür habe ich auf einem Testsystem t38modem Version 0.8.0 eingerichtet dazu Hylafax mit dem im t38modem Package enthaltenen Config gefüttert. Auf dem Patton ein entsprechendes Destination Pattern eingerichtet (hier die Zielrufnummer 4477).

Generell geht der Anruf auch soweit durch. T38modem zeigt mir die eingehenden Anrufe und dabei auch die Ziel und Quellrufnummer. Dann ist die Leitung jedoch tot und im Patton Debugging zeigt "connection" broken an. Auf gateway Seite schiebe ich die entsprechenden Calls einfach auf ein H323 Interface mit einem entsprechenden Voip Profil mit T38 im Faxmodus.

Ich habe mal die entsprechenden Logmeldungen unten angehängt.

Ich fände es sehr nützlich, wenn man in dieser Kombination die T38 Funktion der Smartnodes verwenden könnte.

Hat das schon mal jemand getestet oder sogar direkt im Einsatz?

Falls ja würde mich über jedweden Tipp freuen...

Gruß
Michael

Code:

bit146:~# t38modem -n -o trace.log -p ttyx0,ttyx1 --route [email protected]                                                                                                   T38Modem Version 0.8.0
 by OpenH323 Project on Unix Linux (2.6.18-i686)

Route O/G calls:
  all --> 10.240.30.17
Codecs (in preference order):
 Table:
   G.711-uLaw-64k <1>
   G.711-ALaw-64k <2>
   T.38-UDP <3>
 Set:
   0:
     0:
       G.711-uLaw-64k <1>
       G.711-ALaw-64k <2>
       T.38-UDP <3>

Waiting for incoming calls for "OpenH323 Answering Machine v0.8.0"
I/C connection
To:   4477
Started logical channel: T-101 G.711-uLaw-64k <2> IsTransmitter
Started logical channel: R-3 G.711-ALaw-64k <2> IsReceiver
Closing connection

auf dem gateway sieht das Ganze folgendermassen aus:

Code:

Patton_01#18:47:25  H323  > [EP h323if-0029] Created
18:47:25  H323  > [EP h323if-0029] > Stack: Allocated new call: 0x010ed060
18:47:25  H323  > [EP h323if-0029] > LocCp: Build Local Capabilities merging Configured/Peer Capabilities
18:47:25  H323  > [EP h323if-0029]          Configured Codecs:       Voice: G.711 A-law[20/20], G.711 u-law[20/20] / Fax: G.711 A-law[20/20][vbd]
18:47:25  H323  > [EP h323if-0029]          Peer Codecs:             N/A
18:47:25  H323  > [EP h323if-0029]          Resulting Local Codecs:  Voice: G.711 A-law[20/20], G.711 u-law[20/20] / Fax: G.711 A-law[20/20][vbd]
18:47:25  H323  > [EP h323if-0029] > RemCp: Build Remote Capabilities merging Configured/Remote Capabilities
18:47:25  H323  > [EP h323if-0029]          Configured Codecs:       Voice: G.711 A-law[20/20], G.711 u-law[20/20] / Fax: G.711 A-law[20/20][vbd]
18:47:25  H323  > [EP h323if-0029]          Remote Codecs:           N/A
18:47:25  H323  > [EP h323if-0029]          Resulting Remote Codecs: Voice: G.711 A-law[20/20], G.711 u-law[20/20] / Fax: G.711 A-law[20/20][vbd]
18:47:25  H323  > [PR h323if] Added endpoint h323if-0029
18:47:25  H323  > [EP h323if-0029] > LocCp: Build Local Capabilities merging Configured/Peer Capabilities
18:47:25  H323  > [EP h323if-0029]          Configured Codecs:       Voice: G.711 A-law[20/20], G.711 u-law[20/20] / Fax: G.711 A-law[20/20][vbd]
18:47:25  H323  > [EP h323if-0029]          Peer Codecs:             N/A
18:47:25  H323  > [EP h323if-0029]          Resulting Local Codecs:  Voice: G.711 A-law[20/20], G.711 u-law[20/20] / Fax: G.711 A-law[20/20][vbd]
18:47:25  H323  > [EP h323if-0029] > RemCp: Build Remote Capabilities merging Configured/Remote Capabilities
18:47:25  H323  > [EP h323if-0029]          Configured Codecs:       Voice: G.711 A-law[20/20], G.711 u-law[20/20] / Fax: G.711 A-law[20/20][vbd]
18:47:25  H323  > [EP h323if-0029]          Remote Codecs:           N/A
18:47:25  H323  > [EP h323if-0029]          Resulting Remote Codecs: Voice: G.711 A-law[20/20], G.711 u-law[20/20] / Fax: G.711 A-law[20/20][vbd]
18:47:25  H323  > [EP h323if-0029] > Stack: Dial to remote terminal...
18:47:25  H323  > [EP h323if-0029]          Destination Address:             TA:10.240.30.10,TEL:4477,4477 (complete)
18:47:25  H323  > [EP h323if-0029]          Source Address:                  TEL:,
18:47:25  H323  > [EP h323if-0029]          Presentation Indicator:          Presentation restricted
18:47:25  H323  > [EP h323if-0029]          Screening Indicator:             Network provided
18:47:25  H323  > [EP h323if-0029]          Information Transfer Capability: Speech
18:47:25  H323  > [EP h323if-0029]          Display:
18:47:25  H323  > [EP h323if-0029]          User-User:
18:47:25  H323  > [EP h323if-0029]          Via-Address Supprt:              disabled
18:47:25  H323  > [EP h323if-0029]          Overlap-Sending Support:         enabled
18:47:25  H323  > [EP h323if-0029] > SM   : New state TERMINAL TRYING (RAS)
18:47:25  H323  > [EP h323if-0029] < SM   : Event PEER CONNECTED (State TERMINAL TRYING (RAS))
18:47:25  H323  > [EP h323if-0029] < SM   : Event PROGRESS (State TERMINAL TRYING (RAS))
18:47:25  H323  > [EP h323if-0029] < Stack: State: DIALTONE
18:47:25  H323  > [EP h323if-0029] < Stack: Call-ID: 0228a9cbbeb213a41af800a0ba01c610
18:47:25  H323  > [EP h323if-0029 FS] > Stack: Build Outgoing Faststart Request...
18:47:25  H323  > [EP h323if-0029 FS]          Add Outgoing Logical Channel: G.711 A-law 64k (Audio)
18:47:25  H323  > [EP h323if-0029 FS]          Add Incoming Logical Channel: G.711 A-law 64k (Audio)
18:47:25  H323  > [EP h323if-0029 FS]          Add Outgoing Logical Channel: G.711 u-law 64k (Audio)
18:47:25  H323  > [EP h323if-0029 FS]          Add Incoming Logical Channel: G.711 u-law 64k (Audio)
18:47:25  H323  > [EP h323if-0029 CH] < SM   : Event FASTSTART (State IDLE)
18:47:25  H323  > [EP h323if-0029] Create mode
18:47:25  H323  > [EP h323if-0029 CH] > SM   : New state OPENING (FASTSTART)
18:47:25  H323  > [EP h323if-0029] < SM   : Event PROGRESS (State TERMINAL TRYING (RAS))
18:47:25  H323  > [EP h323if-0029] < SM   : Event TERMINAL DIALTONE (State TERMINAL TRYING (RAS))
18:47:25  H323  > [EP h323if-0029] > SM   : New state TERMINAL TRYING
18:47:26  H323  > [EP h323if-0029] < Stack: State: PROCEEDING
18:47:26  H323  > [EP h323if-0029] < Stack: Received Q.931 CALL PROCEEDING
18:47:26  H323  > [EP h323if-0029]          Progress Indicator: (none)
18:47:26  H323  > [EP h323if-0029] < SM   : Event TERMINAL PROCEEDING (State TERMINAL TRYING)
18:47:26  H323  > [EP h323if-0029] > SM   : New state TERMINAL PROCEEDING
18:47:26  H323  > [EP h323if-0029] < Stack: State: RINGBACK
18:47:26  H323  > [EP h323if-0029] < Stack: Received Q.931 ALERTING
18:47:26  H323  > [EP h323if-0029]          Progress Indicator: (none)
18:47:26  H323  > [EP h323if-0029] < SM   : Event TERMINAL ALERTING (State TERMINAL PROCEEDING)
18:47:26  H323  > [EP h323if-0029] > SM   : New state TERMINAL ALERTING
18:47:26  H323  > [EP h323if-0029] < Stack: State: CONNECTED (CALL-SETUP)
18:47:26  H323  > [EP h323if-0029] < SM   : Event TERMINAL CONNECTED (State TERMINAL ALERTING)
18:47:26  H323  > [EP h323if-0029] > SM   : New state CONNECTED
18:47:26  H323  > [EP h323if-0029] > SM   : Action Send-Status-Inquiry
18:47:26  H323  > [EP h323if-0029] > Stack: Send Q.931 STATUS INQUIRY
18:47:26  H323  > [EP h323if-0029] > SM   : Action Start-Status-Inquiry-Timer
18:47:26  H323  > [EP h323if-0029] > SM   : Action Open-Control-Channel
18:47:26  H323  > [EP h323if-0029] > [EP h323if-0029 CC] Opening...
18:47:26  H323  > [EP h323if-0029] < Stack: Received Q.931 STATUS; audit successful
18:47:26  H323  > [EP h323if-0029] > RemCp: Build Remote Capabilities merging Configured/Remote Capabilities
18:47:26  H323  > [EP h323if-0029]          Configured Codecs:       Voice: G.711 A-law[20/20], G.711 u-law[20/20] / Fax: G.711 A-law[20/20][vbd]
18:47:26  H323  > [EP h323if-0029]          Remote Codecs:           Voice: G.711 u-law[any/240][ss][vuf], G.711 A-law[any/240][ss][vuf] / Fax: G.711 u-law[any/240][vbd], G.711 A-law[any/240][vbd], T.38 UDP[rel] / Modem: G.711 u-law[any/240][!ec][vbd], G.711 A-law[any/240][!ec][vbd]
18:47:26  H323  > [EP h323if-0029]          Resulting Remote Codecs: Voice: G.711 u-law[20/20], G.711 A-law[20/20] / Fax: G.711 A-law[20/20][vbd]
18:47:26  H323  > [EP h323if-0029] < SM   : Event PROGRESS (State CONNECTED)
18:47:26  H323  > [EP h323if-0029 CH] < SM   : Event CONTROL-UP (State OPENING (FASTSTART))
18:47:26  H323  > [EP h323if-0029 FS] < Stack: Remote terminal rejected faststart procedure
18:47:26  H323  > [EP h323if-0029] Cancel mode
18:47:26  H323  > [EP h323if-0029 CH] < SM   : Event MODE-FINISHED (State OPENING (FASTSTART))
18:47:26  H323  > [EP h323if-0029 CH] > SM   : New state FAILED
18:47:26  H323  > [EP h323if-0029] < SM   : Event PROGRESS (State CONNECTED)
18:47:26  H323  > [EP h323if-0029] Create mode
18:47:26  H323  > [EP h323if-0029 MODE 01] > Create Outgoing Logical Channel [EP h323if-0029 OLC 00]
18:47:26  H323  > [EP h323if-0029 CH] > SM   : New state OPENING
18:47:26  H323  > [EP h323if-0029] < SM   : Event PROGRESS (State CONNECTED)
18:47:26  H323  > [EP h323if-0029] < Stack: State: CONNECTED (CALL)
18:47:26  H323  > [EP h323if-0029] < Stack: New Incoming Logical Channel [EP h323if-0029 ILC 00]
18:47:26  H323  > [EP h323if-0029 CH] < SM   : Event MODE-UP (State OPENING)
18:47:26  H323  > [EP h323if-0029 CH] > SM   : New state UP
18:47:26  H323  > [EP h323if-0029] < SM   : Event PROGRESS (State CONNECTED)
18:47:27  H323  > [EP h323if-0029] < DP   : Connection Broken
18:47:37  H323  > [EP h323if-0029 CH] < SM   : Event MODE-FINISHED (State UP)
18:47:37  H323  > [EP h323if-0029 CH] < SM   : Event CONTROL-DOWN (State UP)
18:47:37  H323  > [EP h323if-0029 CH] < SM   : Event MODE-FINISHED (State UP)
18:47:37  H323  > [EP h323if-0029 CH] > SM   : New state SUSPENDED
18:47:37  H323  > [EP h323if-0029] < SM   : Event PROGRESS (State CONNECTED)
18:47:37  H323  > [EP h323if-0029] < Stack: State: DISCONNECTED (NORMAL)
18:47:37  H323  > [EP h323if-0029] < Stack: Received Q.931 RELEASE-COMPLETE (Cause: Normal call clearing)
18:47:37  H323  > [EP h323if-0029] < SM   : Event TERMINAL RELEASE (State CONNECTED)
18:47:37  H323  > [EP h323if-0029] > SM   : New state RELEASED
18:47:37  H323  > [EP h323if-0029] > SM   : Action Terminal-Drop
18:47:37  H323  > [EP h323if-0029] > Stack: Sending Q.931 RELEASE-COMPLETE
18:47:37  H323  > [EP h323if-0029] > SM   : Action Stop-Status-Inquiry-Timer
18:47:37  H323  > [EP h323if-0029] < Stack: State: DISCONNECTED (LOCAL)
18:47:37  H323  > [EP h323if-0029] < Stack: Received Q.931 RELEASE-COMPLETE (Cause: Normal call clearing)
18:47:37  H323  > [EP h323if-0029] < SM   : Event TERMINAL RELEASE (State RELEASED)
18:47:37  H323  > [EP h323if-0029] < Stack: State: IDLE
18:47:37  H323  > [EP h323if-0029] < SM   : Event TERMINAL RELEASE (State RELEASED)
18:47:37  H323  > [EP h323if-0029] > Stack: Closing call 0x010ed060...
18:47:37  H323  > [PR h323if] Removed endpoint h323if-0029
18:47:37  H323  > [PR h323if] Destroyed endpoint h323if-0029
18:47:37  H323  > [EP h323if-0029] Destroy mode
18:47:37  H323  > [EP h323if-0029] Destroy mode
18:47:37  H323  > [EP h323if-0029] Destroyed

Hylafax meldet dazu dann:

Code:

Oct 25 18:42:01 bit146 HylaFAX[2741]: HylaFAX INET Protocol Server: restarted.
Oct 25 18:42:02 bit146 FaxGetty[2562]: MODEM VYACHESLAV FROLOV T38FAX/0.8.0
Oct 25 18:42:02 bit146 FaxGetty[2562]: HELLO
Oct 25 18:43:49 bit146 FaxGetty[2562]: ANSWER: Call ID 1 ""
Oct 25 18:43:49 bit146 FaxGetty[2562]: ANSWER: Call ID 2 "4477"
Oct 25 18:44:00 bit146 FaxGetty[2562]: ANSWER: Ring detected without successful handshake
Oct 25 18:44:08 bit146 FaxGetty[2562]: MODEM VYACHESLAV FROLOV T38FAX/0.8.0
Oct 25 18:45:57 bit146 FaxGetty[2561]: ANSWER: Call ID 1 ""
Oct 25 18:45:57 bit146 FaxGetty[2561]: ANSWER: Call ID 2 "4477"
Oct 25 18:46:09 bit146 FaxGetty[2561]: ANSWER: Ring detected without successful handshake
Oct 25 18:46:16 bit146 FaxGetty[2561]: MODEM VYACHESLAV FROLOV T38FAX/0.8.0
Oct 25 18:47:26 bit146 FaxGetty[2562]: ANSWER: Call ID 1 ""
Oct 25 18:47:26 bit146 FaxGetty[2562]: ANSWER: Call ID 2 "4477"
Oct 25 18:47:37 bit146 FaxGetty[2562]: ANSWER: Ring detected without successful handshake
Oct 25 18:47:45 bit146 FaxGetty[2562]: MODEM VYACHESLAV FROLOV T38FAX/0.8.0

foschi · 26 Okt 2007

[Edit foschi: nicht richtig gelesen...]

Es gibt zwischen den T.38-Implementierungen der verschiedenen Hersteller immer Unterschiede, die den gemeinsamen Betrieb verhindern. Wenn Du Patton <-> Patton verwendest, dann sollte es gehen.

geistio · 20 Jun 2008

Hat man da vielleicht schon eine Lösung gefunden?

mhamann · 26 Jun 2008

nein, habe das Thema damals erfolglos aufgegeben. Könnte aber demnächst nochmal schauen ob sich was verändert hat.

Gerade bei t38modem scheint sich aber schon länger nichts mehr getan zu haben...

MfG
Michael

Seddi75 · 12 Mrz 2009

Der Thread ist zwar schon ein bisschen älter, aber ich grab ihn trotzdem mal aus.

Hab das soeben via SIP zum laufen bekommen:

Patton SN 4552 <-> T38modem <-> Hylafax

Das Smartnode 4552 hat die aktuelle Firmware "R5.3 2009-01-15 SIP" und folgendes unspektakuläres VOIP Profil:

Code:

VoIP Profile: VOIP
==================

  Used:                                 by 1 module(s)

  Voice Codecs
  ------------

    G.711 u-law:                        Rx=20ms Tx=20ms 

  Fax Codecs
  ----------

    T.38 UDP:                           Rx=1ms Tx=1ms Relay 

  Modem Codecs
  Codec Negotiation:                    enabled

  Dejitter
  --------

    Mode:                               Static
    Max. Delay:                         120ms
    Max. Packet Loss:                   4/1000
    Shrink Speed:                       1
    Grow Step:                          1
    Grow Attenuation:                   1
    High Pass Filter:                   enabled
    Post Filter:                        enabled
    CED Net Detection:                  enabled
    CED Net Observe Time:               3000ms
    CED Net Detect Time:                300ms

  Fax
  ---

    Detection:                          CED Tone
    T.38 High Speed Redundant Packets:  0
    T.38 Low Speed Redundant Packets:   0
    Max. Bit Rate:                      14400bps
    Volume:                             -9.500dB
    Error Correction:                   enabled
    HDLC:                               enabled
    Dejitter Max Delay:                 200ms
    Max Datagram Size:                  244 Byte

  Modem
  -----

    Max. Bit Rate:                      14400
    Volume:                             -9.500dB
    HDLC:                               enabled
    Dejitter Max Delay:                 200ms

  DTMF
  ----

    Relay:                              enabled
    Relay-method:                       Default
    Signaling-vendor:                   Default
    Mute Encoder:                       enabled

  FLASH-HOOK
  ----------

    Relay-method:                       DTMF
    Signaling-vendor:                   Default

  RTP
  ---

    Payload Type NTE:                   101
    Payload Type NSE:                   100
    Payload Type Clearmode:             97
    Payload Type Redundancy:            255
    Payload Type G.726-32:              2
    Payload Type G.726-32-AAL2:         2

  BCD
  ---

    Avtivation:                         on-rx-activation
    Timeout:                            1000
    Traffic Class:                      local-default

Das T38modem ist in der Version 1.0 bzw. genauer gesagt CVS Stand vom 11.03.2009. PTlib ist ebenfalls CVS Stand 11.03.09 und Opal hat den CVS Stand vom 21.05.2007.

Code:

cvs -z9 -d :pserver:[email protected]:/cvsroot/openh323 co ptlib_unix
cvs -z9 -d :pserver:[email protected]:/cvsroot/openh323 co -D "5/21/2007 23:59:59" opal
cvs -z9 -d :pserver:[email protected]:/cvsroot/openh323 co t38modem

Das T38modem starte ich mit folgenden Parametern:

Code:

t38modem --no-h323 -n --ptty +/dev/ttyx0,+/dev/ttyx1 --route "modem:.*=sip:<dn>@10.0.0.77" --route "sip:.*=modem:<dn>" --sip-redundancy 3 --sip-old-asn

"--sip-old-asn" war in meinem Fall der Knackpunkt, ohne diese Option geht es nicht. Der Rest des Test-Systems ist ein normales Debian Stable (Hylafax, libavcodec, etc. ) ohne weiteren SchnickSchnack.

Grüße
Sven

Edit Guard-X: Bitte nächste mal Code statt Quote Tags verwenden!

geistio · 16 Mrz 2009

Wow!
Vielen Dank!
Dieses Setup funktioniert bei mir (SN4552 und Hylafax) auch tadellos.
Da scheint sich wirklich was getan zu haben bei t38modem... das letzte mal, als ich dieses Setup ausprobiert habe, hat nichts funktioniert...
Der eigene Hylafax-Server ist jetzt in "greifbare Nähe" gerutscht.

TheEagle · 17 Mrz 2009

Hallo Seddi75 und/oder geistio,

könntet Ihr evtl. noch mehr Details über eure Konfiguration liefern? Versuche seit Tagen das Ganze mit einem SPA2102 zum Laufen zu bekommen, aber momentan scheint es bei mir an der Übergabe vom T38Modem an Hylafax zu scheitern. (siehe OP: "Ring detected without successful handshake") Bin leider noch zu neu in der ganzen Materie, und hab mir das meiste zusammengegoogelt. Aber irgendwo hakt es noch, dehalb wäre ich euch sehr verbunden.

Mein Setup noch kurz: Fax ---[analog]--> SPA2102 (T38 enabled)---[SIP]---->Asterisk---[SIP]--->T38Modem---->HylaFAX.

Seddi75 · 18 Mrz 2009

@geistio
Schön das es nicht nur bei mir klappt

Rennt inzwischen hier seit Tagen ohne Fehler.

@TheEagle
Puh .. ich glaube nicht das du ein problem T38Modem->Hylafax hast ... das ist eigentlich problemlos. Denke mal eher das irgendwo die T38 Umschaltung nicht klappt. Es wird ja während der stehenden Verbindung auf T38 umgeschaltet, wenn erkannt wird das es sich um ein fax handelt, das macht wohl bei vielen Kombinationen Probleme, daran scheiterte es auch immer mit dem SN.
Da solltest du mal eher die Logs von Asterisk und dem SPA anschauen.

ichego1 · 1 Okt 2009

Habe es mit Asterisk und t38 probiert

-- Executing [1607845@voip-in:1] Dial("SIP/1607845-09312fb8", "SIP/T38 modem") in new stack
-- Called T38modem
-- SIP/T38modem-09318180 answered SIP/1607845-09312fb8
-- Packet2Packet bridging SIP/1607845-09312fb8 and SIP/T38modem-09318180
== Spawn extension (voip-in, 1607845, 1) exited non-zero on 'SIP/1607845-09312fb8'

und das beim modem

Call[1] from sip:[email protected]:5063 to 127.0.0.1:6060>;tag=6ef7c271-c9ac-de11-9c4f-000c291dbd15, route to modem:127
Open AudioModemMediaStream-Source-PCM-16 for Call[1]
Open OpalRTPMediaStream-Sink-G.711-uLaw-64k for Call[1]
Open OpalRTPMediaStream-Source-G.711-uLaw-64k for Call[1]
Open AudioModemMediaStream-Sink-PCM-16 for Call[1]
Close OpalRTPMediaStream-Sink-G.711-uLaw-64k
Close OpalRTPMediaStream-Source-G.711-uLaw-64k
Close AudioModemMediaStream-Source-PCM-16
Close AudioModemMediaStream-Sink-PCM-16
Call[1] cleared

bekomme kein piepston wie fax

hat jemand eine lösung

the · 12 Jan 2010

hi,

sind zwar schon etwas länger her die postings...

aber auch ich bekomme keine signalgebung <no carrier detected>

hat jemand mal eine grundsätzlich anleitung?

mfg
-the-

ichego1 · 18 Aug 2010

T38MODEM UNTER DEBIAN Lenny beim hochfahren starten

Code:

t38modem --no-h323 -n --ptty +/dev/ttyx0,+/dev/ttyx1 --route "modem:.*=sip:<dn>@10.0.0.77" --route "sip:.*=modem:<dn>" --sip-redundancy 3 --sip-old-asn

wo muss ich das unter debian reinfriemeln

hat jemand eine Idee

danke

the · 1 Dez 2010

t38modem never ending story

hi,

nach dem ich etwas eingespannt war, will ich es mit patton(192.168.100.71) <-> asterisk (192.168.100.70 bzw. 127.0.0.1) <t38modem> (127.0.0.1) noch mal wagen.

mein testsystem: debian squeeze
die versionen:
- asterisk 1.6.2x
- spandsp 0.0.6x
- iaxmodem 1.2x (als referenz!)
- t38modem 1.2x

ich habe das ganze soweit installiert und funktionsgeprüft:
- verbindung via sipgate (voice) funktioniert
- faxempfang via iax2 hebt ab und fiept.
- faxversand via iax2 wählt raus.

aber das liebe t38modem will und will nicht!
- beim versand bekomme ich "no dialtone"
- beim empfang bricht das ganze auch zusammen (kein signalton)

da hier ja leute behaupten es laufen zu haben, bitte ich um den notwendigen geistesblitz!

anbei meine konfigurationen:

meine patton

Code:

#----------------------------------------------------------------#
#                                                                #
# SN4552/2BIS/EUI                                                #
# R5.3 2009-11-17 SIP                                            #
# 2010-12-01T11:55:53                                            #
# SN/00A0BA033983                                                #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
gui type basic
no terminal telnet
clock local offset +01:00
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary 192.168.100.70 port 123 version 4

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
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 voip mitFAX
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 g711ulaw64k rx-length 20 tx-length 20
  dtmf-relay rtp
  rtp traffic-class local-default
  ced net-side-detection
  fax transmission 1 relay t38-udp
  fax transmission 2 bypass g711alaw64k
  fax redundancy low-speed 0 high-speed 1
  fax volume -10.5
  modem dejitter-max-delay 300
  no modem detection on-remote-fax-request

profile voip t38
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 g711ulaw64k rx-length 20 tx-length 20
  ced net-side-detection
  fax transmission 1 relay t38-udp

profile pstn default

profile sip default

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 192.168.100.71 255.255.255.0
    icmp router-discovery

subscriber ppp SUB_PPPOE
  dial out
  no multilink
  authentication chap
  authentication pap
  bind interface IF_IP_WAN router

context cs switch
  national-prefix 0
  international-prefix 00

  routing-table called-e164 ROUTING_ALL_IN
  routing-table called-e164 ROUTING_ALL_OUT
    route .T dest-service HUNT_ALL_IN

  interface isdn ISDN0
    route call dest-interface asterisk
    dtmf-dialing
    use profile tone-set Europe

  interface isdn IF_S0_01

  interface sip asterisk
    bind context sip-gateway asterisk
    route call dest-interface ISDN0
    remote 192.168.100.70 5060
    use profile voip t38
    use profile tone-set Europe

  interface sip IF_SIP_SERVICE

  service hunt-group HUNT_ALL_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

  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

authentication-service AUTH_SVC

location-service LOCATION_SVC
  domain 1 192.168.100.70

  identity-group default

    registration outbound
      proxy 1 192.168.100.70
      register auto

    call outbound
      proxy 1 192.168.100.70

context sip-gateway asterisk

  interface 192.168.100.70
    bind interface IF_IP_LAN context router port 5060

context sip-gateway asterisk
  no shutdown

port ethernet 0 0
  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
  bind interface IF_IP_LAN router
  no shutdown

port bri 0 0
  clock auto
  encapsulation q921

  q921
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface ISDN0 switch

port bri 0 0
  no shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    uni-side auto
    encapsulation q931

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

port bri 0 1
  no shutdown

t38modem

Code:

#!/bin/bash
 
T38_PROCESSES=2
START_PORT=6060
ASTERISK_IP=127.0.0.1
BIND_IP=127.0.0.1
 
case "$1" in
          start)
                    COUNTER=0
                    CURRENT_PORT=${START_PORT}
 
                    while [  $COUNTER -lt $T38_PROCESSES ]; do
 			      COMMAND="t38modem -tttt -o /var/log/t38modem.log --no-h323 -u T38modem${COUNTER} --sip-listen udp\$${BIND_IP}:${CURRENT_PORT} --ptty +/dev/ttyT38-${COUNTER}  --route \"modem:.*=sip:<dn>@${ASTERISK_IP}\" --route \"sip:.*=modem:<dn>\""	
                              exec $COMMAND > /dev/null 2>&1 &
                              PID=$!
                              echo "Starting t38modem with pid $PID (pidfile /var/run/t38modem/$COUNTER)"
                              echo $PID > /var/run/t38modem/$COUNTER
                              COUNTER=$[ COUNTER + 1 ]
                              CURRENT_PORT=$[ CURRENT_PORT + 1 ]
                    done
          ;;
          stop)
                    echo -n "Stopping t38modem pids: "
 
                    for pidfile in `ls /var/run/t38modem`
                    do
                              _PID=`cat /var/run/t38modem/$pidfile`
                              kill -9 $_PID
                              rm /var/run/t38modem/$pidfile
                              echo -n $_PID" "
                    done
                    echo " Done"
          ;;
          *)
                    echo "Usage: $0 {start|stop}" >&2
                    exit 1
          ;;
esac
 
exit 0

sip.conf

Code:

[general]
t38pt_udptl = yes
context=default
port = 5060
bindaddr = 0.0.0.0
srvlookup=yes
language=de
register => XXX:[email protected]/XXX

[XXX]
username=XXX
secret=XXX
type=friend
fromuser=XXX
host=sipgate.de
fromdomain=sipgate.de
insecure=invite
canreinvite=no
nat=yes
disallow=all
allow=alaw
allow=ulaw
localdiaplan=4
dtmfmode=rfc2833

[sipgate_de_in]
type=peer
fromdomain=sipgate.de
host=sipgate.de
disallow=all
allow=alaw
allow=ulaw
allow=gsm
dtmfmode=rfc2833
context=office

[isdnGateway01]
t38pt_udptl = yes 
type=peer
insecure=invite
host=192.168.100.71
context=office
localdialplan=4

[t38modem-options](!)
type = friend
host = 127.0.0.1
permit=127.0.0.1
context=fax-out
canreinvite = yes
allow =all
t38pt_udptl = yes
dtmfmode = rfc2833
nat = no
 
[T38modem0](t38modem-options)
port = 6060
 
[T38modem1](t38modem-options)
port = 6061

extensions.conf

Code:

;!
[default]
exten => 1001,1,Answer()
exten => 1001,2,Playback(hello-world)
exten => 1001,3,Hangup()

; ansage bei anrufen über sipgate
exten => XXX,1,Answer()
exten => XXX,n,Playback(zeiten)
exten => XXX,n,Hangup()

[sonstige]

[office]
include = default
plancomment = office
parked = no

; IAXMODEM
exten => _711,1,NoOp("Incoming fax to ${EXTEN}")
exten => _711,n,Set(AGAINCOUNTER=0)                          
exten => _711,n,Wait(3)
exten => _711,n(again),Set(AGAINCOUNTER=$[${AGAINCOUNTER} + 1])
exten => _711,n,GotoIf($[${AGAINCOUNTER} <= 20]?call:hangup)
exten => _711,n(call),Dial(IAX2/${EXTEN}@iaxmodem-${RAND(0,1)},20,gH)             
exten => _71,n,Wait(2)
exten => _711,n,GotoIf($["${DIALSTATUS}" != "ANSWER"]?again:checktime)
exten => _711,n(checktime),GotoIf($[${ANSWEREDTIME} < 10]?again:hangup)
exten => _711,n(hangup),Hangup()                                                


; T38 Modem
exten => _710,1,NoOp("Incoming fax to ${EXTEN}")
exten => _710,n,Set(AGAINCOUNTER=0)                          
exten => _710,n,Wait(6)
exten => _710,n(again),Set(AGAINCOUNTER=$[${AGAINCOUNTER} + 1])
exten => _710,n,GotoIf($[${AGAINCOUNTER} <= 20]?call:hangup)
exten => _710,n(call),Dial(SIP/${EXTEN}@T38modem${RAND(0,1)},20,gH)
exten => _710,n,Wait(2)
exten => _710,n,GotoIf($["${DIALSTATUS}" != "ANSWER"]?again:checktime)
exten => _710,n(checktime),GotoIf($[${ANSWEREDTIME} < 10]?again:hangup)
exten => _710,n(hangup),Hangup()                                                


; spandsp
exten => _712,1,Verbose(1,### Eingehendes Fax ${CDR(uniqueid)})
exten => _712,n,Set(LOCALSTATIONID=Meine Firma)
exten => _712,n,Answer()
exten => _712,n,Wait(6)
exten => _712,n,ReceiveFAX(/tmp/fax-${CDR(uniqueid)}.tif)
exten => _712,n,Verbose(1,###       FAXSTATUS: ${FAXSTATUS})
exten => _712,n,Verbose(1,###        FAXERROR: ${FAXERROR})
exten => _712,n,Verbose(1,###         FAXMODE: ${FAXMODE})
exten => _712,n,Verbose(1,###        FAXPAGES: ${FAXPAGES})
exten => _712,n,Verbose(1,###      FAXBITRATE: ${FAXBITRATE})
exten => _712,n,Verbose(1,###   FAXRESOLUTION: ${FAXRESOLUTION})
exten => _712,n,Verbose(1,### REMOTESTATIONID: ${REMOTESTATIONID})
exten => _712,n,Hangup()

;iaxmodem
[fax-out-0]
exten => _X.,1,NoOp("IAXModem an ${EXTEN}")
exten => _X.,n,Answer()
exten => _X.,n,Dial(SIP/${EXTEN}@isdnGateway01)
exten => _X.,n,Hangup()

; t38modem
[fax-out]
exten => _X.,1,NoOp("T38Modem an ${EXTEN}")
exten => _X.,n,Dial(SIP/${EXTEN}@isdnGateway01)
exten => _X.,n,Hangup()

abschliesend mein log vom t38modem

Code:

INVITE sip:[email protected]:6060 SIP/2.0
Date: Wed, 01 Dec 2010 10:36:53 GMT^M
CSeq: 102 INVITE^M
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK69d20473;rport^M
User-Agent: Asterisk PBX 1.6.2.9-2^M
From: "YYY" <sip:[email protected]>;tag=as452e7adb^M
Call-ID: [email protected]^M
Supported: replaces, timer^M
To: <sip:[email protected]:6060>^M
Contact: <sip:[email protected]>^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO^M
Content-Type: application/sdp^M
Content-Length: 415^M
Max-Forwards: 70^M
^M
v=0^M
o=root 1886933360 1886933360 IN IP4 127.0.0.1^M
s=Asterisk PBX 1.6.2.9-2^M
c=IN IP4 127.0.0.1^M
t=0 0^M
m=audio 11988 RTP/AVP 0 3 8 112 5 10 7 110 111 9^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:3 GSM/8000^M
a=rtpmap:8 PCMA/8000^M
a=rtpmap:112 AAL2-G726-32/8000^M
a=rtpmap:5 DVI4/8000^M
a=rtpmap:10 L16/8000^M
a=rtpmap:7 LPC/8000^M
a=rtpmap:110 speex/8000^M
a=rtpmap:111 G726-32/8000^M
a=rtpmap:9 G722/8000^M
a=ptime:20^M
a=sendrecv^M2010/12/01 11:36:53.514 Opal Liste...0xf5f10b70 SIP     Sending PDU (301 bytes) to: rem=udp$127.0.0.1:5060,local=udp$127.0.0.1:6060,if=127.0.0.1%lo
SIP/2.0 100 Trying^M
CSeq: 102 INVITE^M
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK69d20473;rport^M
From: "YYY" <sip:[email protected]>;tag=as452e7adb^M
Call-ID: [email protected]^M
To: <sip:[email protected]:6060>^M
Contact: <sip:[email protected]:6060>^M
Content-Length: 0^M
^M

2010/12/01 11:36:53.514 Opal Liste...0xf5f10b70 OpalUDP Setting interface to 127.0.0.1%lo
2010/12/01 11:36:53.515 Opal Liste...0xf5f10b70 MonSock Created monitored socket for interfaces 127.0.0.1
2010/12/01 11:36:53.515 Opal Liste...0xf5f10b70 SIP     Created transport udp$127.0.0.1:5060<if=udp$127.0.0.1>
2010/12/01 11:36:53.515 Opal Liste...0xf5f10b70 OpalUDP Started connect to 127.0.0.1:5060
2010/12/01 11:36:53.515 Opal Liste...0xf5f10b70 PWLib   File handle high water mark set: 28 PUDPSocket
2010/12/01 11:36:53.515 Opal Liste...0xf5f10b70 MonSock Created bundled UDP socket 127.0.0.1:34510
2010/12/01 11:36:53.515 Opal Liste...0xf5f10b70 Call    Created Call[jc61510341]
2010/12/01 11:36:53.516 Opal Liste...0xf5f10b70 PWLib   File handle high water mark set: 29 PUDPSocket
2010/12/01 11:36:53.516 Opal Liste...0xf5f10b70 MySIPEndPoint::CreateConnection for Call[jc61510341]
2010/12/01 11:36:53.516 Opal Liste...0xf5f10b70 OpalCon Created connection Call[jc61510341]-EP<sip>[988cdd94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.516 Opal Liste...0xf5f10b70 RFC2833 Handler created
2010/12/01 11:36:53.516 Opal Liste...0xf5f10b70 RFC2833 Handler created
2010/12/01 11:36:53.517 Opal Liste...0xf5f10b70 SIP     Created connection.
2010/12/01 11:36:53.517 Opal Liste...0xf5f10b70 SIP     Queueing PDU "102 INVITE sip:[email protected]:6060", transaction=z9hG4bK69d20473, token=988cdd94-a4fb-df11-930f-00163e710f0b
2010/12/01 11:36:53.517 Opal Liste...0xf5f10b70 PWLib   File handle high water mark set: 30 Thread unblock pipe
2010/12/01 11:36:53.517 Opal Liste...0xf5f10b70 PTLib   Thread high water mark set: 11
2010/12/01 11:36:53.517 Opal Liste...0xf5f10b70 Opal    Transport clean up on termination
2010/12/01 11:36:53.517 Opal Liste...0xf5f10b70 Opal    Transport Close
2010/12/01 11:36:53.517 Opal Liste...0xf5f10b70 Opal    Deleted transport udp$127.0.0.1:5060<if=udp$127.0.0.1:6060>
2010/12/01 11:36:53.518         Pool:0xf5e10b70 SIP     Handling PDU "102 INVITE sip:[email protected]:6060" for token=988cdd94-a4fb-df11-930f-00163e710f0b
2010/12/01 11:36:53.518         Pool:0xf5e10b70 SIP     Initial INVITE from sip:[email protected]:6060
2010/12/01 11:36:53.518         Pool:0xf5e10b70 SIP     Set Request URI to sip:[email protected]
2010/12/01 11:36:53.518         Pool:0xf5e10b70 SIP     Updating dialog tag from "" to "as452e7adb"
2010/12/01 11:36:53.518         Pool:0xf5e10b70 SIP     Product Info: name="Asterisk", version="", vendor="", comments="PBX 1.6.2.9-2"
2010/12/01 11:36:53.519         Pool:0xf5e10b70 OPAL    Checking incoming call for NAT: local=127.0.0.1, peer=127.0.0.1, sig=127.0.0.1
2010/12/01 11:36:53.519         Pool:0xf5e10b70 OpalCon SetPhase from UninitialisedPhase to SetUpPhase for Call[jc61510341]-EP<sip>[988cdd94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.519         Pool:0xf5e10b70 OpalMan OnIncoming connection Call[jc61510341]-EP<sip>[988cdd94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.519         Pool:0xf5e10b70 OpalCon Applying string options:
2010/12/01 11:36:53.519         Pool:0xf5e10b70 Call    GetOtherPartyConnection Call[jc61510341]-EP<sip>[988cdd94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.519         Pool:0xf5e10b70 OpalMan Searching for route "sip:[email protected]:6060     sip:[email protected]:6060"
2010/12/01 11:36:53.519         Pool:0xf5e10b70 OpalMan Set up connection to "sip:[email protected]:6060"
2010/12/01 11:36:53.519         Pool:0xf5e10b70 MySIPEndPoint::CreateConnection for Call[jc61510341]
2010/12/01 11:36:53.520         Pool:0xf5e10b70 OpalCon Created connection Call[jc61510341]-EP<sip>[2e28de94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.520         Pool:0xf5e10b70 RFC2833 Handler created
2010/12/01 11:36:53.520         Pool:0xf5e10b70 RFC2833 Handler created
2010/12/01 11:36:53.520         Pool:0xf5e10b70 SIP     Created connection.
2010/12/01 11:36:53.520         Pool:0xf5e10b70 Call[jc61510341] from sip:[email protected]:6060 to sip:[email protected]:6060, route to sip:udp$127.0.0.1:6060
2010/12/01 11:36:53.520         Pool:0xf5e10b70 SIP     OnIncomingConnection succeeded for INVITE from sip:[email protected]:6060 for Call[jc61510341]-EP<sip>[988cdd94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.520         Pool:0xf5e10b70 Call    GetOtherPartyConnection Call[jc61510341]-EP<sip>[988cdd94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.520         Pool:0xf5e10b70 Call    OnSetUp Call[jc61510341]-EP<sip>[988cdd94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.520         Pool:0xf5e10b70 MySIPConnection::SetUpConnection Call[jc61510341]-EP<sip>[2e28de94-a4fb-df11-930f-00163e710f0b] name=T38modem0
2010/12/01 11:36:53.520         Pool:0xf5e10b70 MySIPConnection::SetUpConnection new name=YYY
2010/12/01 11:36:53.520         Pool:0xf5e10b70 SIP     SetUpConnection: sip:[email protected]:6060
2010/12/01 11:36:53.520         Pool:0xf5e10b70 OpalCon SetPhase from UninitialisedPhase to SetUpPhase for Call[jc61510341]-EP<sip>[2e28de94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.521         Pool:0xf5e10b70 OpalCon Applying string options:


2010/12/01 11:36:53.521         Pool:0xf5e10b70 SIP     No SRV lookup as has explicit port number.
2010/12/01 11:36:53.521         Pool:0xf5e10b70 SIP     Connecting to sip:[email protected]:6060 via sip:[email protected]:6060
2010/12/01 11:36:53.521         Pool:0xf5e10b70 PWLib   File handle high water mark set: 31 PUDPSocket
2010/12/01 11:36:53.521         Pool:0xf5e10b70 MonSock Created monitored socket for interfaces 127.0.0.1
2010/12/01 11:36:53.521         Pool:0xf5e10b70 SIP     Created transport udp$127.0.0.1:6060<if=udp$127.0.0.1>
2010/12/01 11:36:53.521         Pool:0xf5e10b70 OpalUDP Started connect to 127.0.0.1:6060
2010/12/01 11:36:53.522         Pool:0xf5e10b70 PWLib   File handle high water mark set: 32 PUDPSocket
2010/12/01 11:36:53.522         Pool:0xf5e10b70 MonSock Created bundled UDP socket 127.0.0.1:41658
2010/12/01 11:36:53.522         Pool:0xf5e10b70 OpalUDP Writing to interface 0 - "127.0.0.1%lo"
2010/12/01 11:36:53.522         Pool:0xf5e10b70 OpalMan Listener interfaces: associated transport=None
    udp$127.0.0.1:6060
2010/12/01 11:36:53.523         Pool:0xf5e10b70 SIP     Updating dialog tag from "" to "2e28de94-a4fb-df11-930f-00163e710f0b"
2010/12/01 11:36:53.526         Pool:0xf5e10b70 SIP     INVITE transaction id=z9hG4bKec22df94-a4fb-df11-930f-00163e710f0b created.
2010/12/01 11:36:53.526         Pool:0xf5e10b70 SIP     Creating INVITE request
2010/12/01 11:36:53.526         Pool:0xf5e10b70 MediaFormat     Removing codecs
2010/12/01 11:36:53.526         Pool:0xf5e10b70 MediaFormat     Removing codecs
2010/12/01 11:36:53.527         Pool:0xf5e10b70 Call    GetMediaFormats for Call[jc61510341]-EP<sip>[2e28de94-a4fb-df11-930f-00163e710f0b]
T.38

2010/12/01 11:36:53.527         Pool:0xf5e10b70 SIP     No media formats of type audio, not adding SDP
2010/12/01 11:36:53.527         Pool:0xf5e10b70 MediaFormat     Removing codecs
2010/12/01 11:36:53.527         Pool:0xf5e10b70 MediaFormat     Removing codecs
2010/12/01 11:36:53.527         Pool:0xf5e10b70 Call    GetMediaFormats for Call[jc61510341]-EP<sip>[2e28de94-a4fb-df11-930f-00163e710f0b]
T.38
2010/12/01 11:36:53.527         Pool:0xf5e10b70 SIP     No media formats of type video, not adding SDP
2010/12/01 11:36:53.527         Pool:0xf5e10b70 OpalCon SetPhase from SetUpPhase to ReleasingPhase for Call[jc61510341]-EP<sip>[2e28de94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.527         Pool:0xf5e10b70 OpalCon Releasing Call[jc61510341]-EP<sip>[2e28de94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.527         Pool:0xf5e10b70 OpalCon Call end reason for Call[jc61510341]-EP<sip>[2e28de94-a4fb-df11-930f-00163e710f0b] set to EndedByCapabilityExchange
2010/12/01 11:36:53.527         Pool:0xf5e10b70 PWLib   File handle high water mark set: 34 Thread unblock pipe
2010/12/01 11:36:53.527         Pool:0xf5e10b70 PTLib   Thread high water mark set: 12
2010/12/01 11:36:53.528         Pool:0xf5e10b70 SIP     Aborting INVITE transaction since connection is in releasing phase
2010/12/01 11:36:53.528         Pool:0xf5e10b70 SIP     Destroying transaction id=z9hG4bKec22df94-a4fb-df11-930f-00163e710f0b which is not yet terminated.
2010/12/01 11:36:53.528         Pool:0xf5e10b70 SIP     Transaction id=z9hG4bKec22df94-a4fb-df11-930f-00163e710f0b destroyed.
2010/12/01 11:36:53.528    OnRelease:0xf5dd0b70 SIP     OnReleased: Call[jc61510341]-EP<sip>[2e28de94-a4fb-df11-930f-00163e710f0b], phase = ReleasingPhase
2010/12/01 11:36:53.528    OnRelease:0xf5dd0b70 OpalCon SetPhase from ReleasingPhase to ReleasingPhase for Call[jc61510341]-EP<sip>[2e28de94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.528    OnRelease:0xf5dd0b70 OpalCon Media streams closed.
2010/12/01 11:36:53.528    OnRelease:0xf5dd0b70 OpalCon SetPhase from ReleasingPhase to ReleasedPhase for Call[jc61510341]-EP<sip>[2e28de94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.528    OnRelease:0xf5dd0b70 OpalCon OnReleased Call[jc61510341]-EP<sip>[2e28de94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.528    OnRelease:0xf5dd0b70 OpalEP  OnReleased Call[jc61510341]-EP<sip>[2e28de94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.528    OnRelease:0xf5dd0b70 OpalMan OnReleased Call[jc61510341]-EP<sip>[2e28de94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.528    OnRelease:0xf5dd0b70 Call    OnReleased Call[jc61510341]-EP<sip>[2e28de94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.528    OnRelease:0xf5dd0b70 OpalCon SetPhase from SetUpPhase to ReleasingPhase for Call[jc61510341]-EP<sip>[988cdd94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.529         Pool:0xf5e10b70 SIP     Could not write to sip:[email protected]:6060 -
2010/12/01 11:36:53.529         Pool:0xf5e10b70 OpalCon Already released Call[jc61510341]-EP<sip>[2e28de94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.529         Pool:0xf5e10b70 SIP     OnSetUp failed for INVITE from sip:[email protected]:6060 for Call[jc61510341]-EP<sip>[988cdd94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.529         Pool:0xf5e10b70 OpalCon Already released Call[jc61510341]-EP<sip>[988cdd94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.529         Pool:0xf5e10b70 SIP     Handled PDU "102 INVITE sip:[email protected]:6060"
2010/12/01 11:36:53.529    OnRelease:0xf5dd0b70 OpalCon Releasing Call[jc61510341]-EP<sip>[988cdd94-a4fb-df11-930f-00163e710f0b]
2010/12/01 11:36:53.529    OnRelease:0xf5dd0b70 OpalCon Call end reason for Call[jc61510341]-EP<sip>[988cdd94-a4fb-df11-930f-00163e710f0b] set to EndedByCapabilityExchange
...

danke
-the-

domi2001 · 16 Feb 2012

Der Thread ist nun zwar schon ein bisschen in die Jahre gekommen, aber das Thema ist für mich dennoch aktuell.

Meine Konstellation besteht ebenso aus einem Patton SN4552 -> T.38-Modem -> Hylafax.
Primär sollen nur Faxe mittels Hylafax empfangen werden können.

Hylafax rufe ich folgendermaßen auf:

t38modem --no-h323 -n --ptty +/dev/ttyx0,+/dev/ttyx1 --route "modem:.*=sip:<dn>@192.168.7.103" --route "sip:.*=modem:<dn>" --sip-redundancy 3 --sip-old-asn

Eingehende Anrufe sehe ich und diese werden auch angenommen - allerdings passiert nicht mehr als das und die Verbindung wird nach einigen Sekuden wieder getrennt.

Das Logfile sagt:

Call[u0c08a33a10] from sip:[email protected] to sip:[email protected], route to modem:911123
Open OpalRTPMediaStream-Source-G.711-ALaw-64k for Call[u0c08a33a10]
Open AudioModemMediaStream-Sink-PCM-16 for Call[u0c08a33a10]
Open AudioModemMediaStream-Source-PCM-16 for Call[u0c08a33a10]
Open OpalRTPMediaStream-Sink-G.711-ALaw-64k for Call[u0c08a33a10]
Close OpalRTPMediaStream-Sink-G.711-ALaw-64k for Call[u0c08a33a10]
Close AudioModemMediaStream-Source-PCM-16 for Call[u0c08a33a10]
Close AudioModemMediaStream-Sink-PCM-16 for Call[u0c08a33a10]
Close OpalRTPMediaStream-Source-G.711-ALaw-64k for Call[u0c08a33a10]
Open OpalRTPMediaStream-Source-T.38 for Call[u0c08a33a10]
Open T38ModemMediaStream-Source-T.38 for Call[u0c08a33a10]
Open OpalRTPMediaStream-Sink-T.38 for Call[u0c08a33a10]
Open T38ModemMediaStream-Sink-T.38 for Call[u0c08a33a10]
Close T38ModemMediaStream-Sink-T.38 for Call[u0c08a33a10]
Close OpalRTPMediaStream-Source-T.38 for Call[u0c08a33a10]
Close OpalRTPMediaStream-Sink-T.38 for Call[u0c08a33a10]
Close T38ModemMediaStream-Source-T.38 for Call[u0c08a33a10]
Call[u0c08a33a10] cleared

Woran liegt das?
Hat jemand eine lauffähige Konfiguration für die genannte Konstellation?

Vielen Dank im Voraus!

Schöne Grüße,
domi2001

TheEagle · 16 Feb 2012

Ich hatte damals aufgegeben und per Sangoma Analog Karte erledigt. Asterisk 1.8 bzw Version 10 hat aber auch einiges an Neuerungen dabei, mit Version 1.8 kann ich von "guten" T.38 Gegenstellen Faxe direkt annehmen, ohne Hylafax. Interessanter ist aber die T.38 Gatewayfunktion von Asterisk 10, damit solltest Du evtl. in der Lage sein eingehende T.38 Faxe über das gute alte iaxmodem an Hylafax abzugeben. Oder nimmst die direkt in Asterisk an, aber Hylafax kann natürlich mehr. Getestet habe ich Version 10 aber selbst noch nicht, also alles nur Theorie von mir nach Lesen der Release-Notes.

HobbyStern · 18 Feb 2012

WIr halten fest - t.38 Implementierungen begeistern wegen der großen Flexi- und Interoperabilität - aber ganz klar, so "richtig" produktiv sind Sie im softwaregeschehen noch nicht und schon gar nicht in heterogenen Aufbauten...

Ich schiebe t.38 mit mehreren Endgeräten auch schon seit Jahren vor mir her..

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