H323 ELASTIX 2 AND ALCATEL 4400 CUT OFF -- RTP ?

Discussion in 'Elastix 2.x' started by scloeza, Jan 23, 2011.

  1. scloeza

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    Hello, Greetings.

    Infinitely grateful if someone can help me out of this problem.

    I'm connecting Asterisk <---> Alcatel 4400, when I make the call from Alcatel everything works beautifully and does not break, I made the connection until an hour and not cut and voice flows in two ways.

    Use with h323 trunk on Elastix 2.0 chan_ooh323.so

    My problem is when he called the Asterisk (xlite or bria) provides the call successfully, we present the flow of speech in both directions, but when you go for 3 seconds the call is interrupted, I've read countless forums and have helped me a lot reach the point where I am, and almost made it, if not for this detail. In some of the forums I found the recommendation to run the command cpl_online and got the following:

    4.3 BSD UNIX (xa000000) (ttyp0)

    mtch
    Password:

    # The role of the CPU is MAIN
    Application software identity

    R5.0Ux-d2.314-4-f-mx-c6s2

    Business identification: R5.0Ux

    Release:
    DELIVERY d2.314
    Patch identification: 4
    Dynamic patch identification: f

    Country: mx
    Cpu: c6s2

    ACD VERSION
    release : 4
    bug_fixing : 4
    protocol_id : 75
    version_dy_hr_stat : 11


    There is a dump on /dev/dsk/crashdump from 10/10/03 at 19:26:23

    (1)xa000000> cpl_online 0 4

    00000951-0F3DC8A4: Connected to Crystal 0 Coupler 4
    Escape character is '^D'
    00000952-0F3DCF9E: !!!!! h323 protocol version not supported=4
    00000953-0F3DCF9E: !!!!! h323 protocol version not supported=4
    00000954-0F3DDB80: !!!!! h323 protocol version not supported=4
    00000955-0F3DDB95: !!!!! h323 protocol version not supported=4
    00000956-0F3DDB97: !!!!! h323 protocol version not supported=4
    00000957-0F3DDB97: !!!!! h323 protocol version not supported=4
    00000958-0F3DDB97: !!!!! h323 protocol version not supported=4
    00000959-0F3DDCFF: H245 T101 expired: repetition without T38
    0000095A-0F3DDCFF: H245 TCS message: second time
    0000095B-0F3DDD13: !!!!! h323 protocol version not supported=4
    0000095C-0F3DDE90: ===>Send H245 Ind Message Terminal Capability Set Error T101
    expired
    0000095D-0F3DDE90: !!!!!! Terminal Capability Set ack not received !!!!!!
    0000095E-0F3DDE99: Assertion failed: (channelStatus == CHANNEL_INITIALIZED || ch
    annelStatus == CHANNEL_ALLOCATED) file /twdhs3/trinite1/ws/r_intip_4.xx/ca_voip/
    code/IP_JitterFax.cc - line 397


    Moreover H323 review several items and took me to use wireshark, so I got the following:

    Data.cap annexed

    Truly I can appreciate support me, and took almost a year. I'm traumatized,

    THANKS

    Sergio Cervera
    sergio.acervera @ gmail.com
     
  2. danardf

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    Hi..

    hmmm. maybe a compatibility issue or config issue.

    Try to make a file text to put your config trunk, h323 side 4400 and your config file ooh323.conf and more details than possible.

    If i've some free time, i'll try to make some test. But not now, and not in the same version (R9.x for me).
     
  3. scloeza

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    Hi, Franck, thank´s.

    This is my ooh323 http://forum.elastix.org/old_files/ooh323.rar

    From what I read, the team 172.16.67.253, contends the RTP parameter and asterisk does not send it. You think.

    I looked that way you can change that value and send that value from the Asterisk and I have not found anything

    Thanks for supporting me.

    Greetings
     
  4. danardf

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    At first, why put 2 same parameters in the config file?

    Code:
    [Alcatel]
    type=peer
    context=default
    ip=172.16.67.253   ; UPDATE with appropriate ip address
    port=1720    ; UPDATE with appropriate port
    rtptimeout=3 ;
    disallow=all
    ;allow=ulaw
    ;allow=alaw
    allow=g729
    ;allow=g723
    ;e164=10
    rtptimeout=60
    faststart=yes
    canreinvite=yes  ;requerido para que fluya la voz en ambos sentidos
    dtmfmode=h245signal
    rtptimeout=3
    rtptimeout=60

    Select only one line and not 2.

    canreinvite=yes.. Be careful, If the 4400 not use the RTP Direct in the network, then this parameter is wrong. If yes, also check that you haven't another parameter about "h323 RTP direct".
    You could still make some test why canerinvite=no

    You could select another context like for example from-internal too.
    If Elastix is a part of the 4400 network, then do it.
     
  5. scloeza

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    Hi,
    Mistakenly let the two in the last configuration, but I've only tested it with a line and I have changed to 3, 5, 60, 30, 180 and is the same.

    I've tried both options, conreinvite = no, the voice stream does not pass in both sense and yes the flow is two-way voice


    Now change the context to from-internal

    Thank´s
     
  6. scloeza

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    Hi,
    Mistakenly let the two in the last configuration, but I've only tested it with a line and I have changed to 3, 5, 60, 30, 180 and is the same.

    I've tried both options, conreinvite = no, the voice stream does not pass in both sense and yes the flow is two-way voice


    Now change the context to from-internal

    Thank´s
     
  7. danardf

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    Maybe try to find some log side Asteirsk. (ooh323 set debug)
    Check this file:
    /var/log/asterisk/h323_log

    normally, dtmfmode is RFC2833 no?
     
  8. scloeza

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    Hi good day.

    Hi good day.

    Make the change in DMTF = rfc2833, Annex I setup my ooh323 and full log of the asterisk, where the call is interrupted, I hope to serve, is in rtf format. http://forum.elastix.org/old_files/onepartofasteriskfulllog.rar

    Thanks



    Objective Open H.323 Channel Driver's Config:
    IP:port: 0.0.0.0:1720
    FastStart yes
    Tunneling no
    CallerId IP_Interna
    MediaWaitForConnect no
    Gatekeeper: No Gatekeeper
    H.323 LogFile: /var/log/asterisk/h323_log
    Context: default
    Capability: 0x100 (g729)
    DTMF Mode: rfc2833
    AccountCode: h323asterisk
    AMA flags: Unknown
    Aliases:
    100 Asterisk PBX
     
  9. danardf

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    Si tu regardes les log, tu vois bien çà:
    Code:
    [Jan 21 11:09:35] VERBOSE[5030] app_dial.c:     -- OOH323/172.16.67.253:1720-b7d14370 is ringing
    [Jan 21 11:09:39] VERBOSE[5030] app_dial.c:     -- OOH323/172.16.67.253:1720-b7d14370 answered SIP/8001-00000010
    [Jan 21 11:09:39] WARNING[5030] chan_ooh323.c: Don't know how to indicate condition 25 on ooh323c_o_10
    [Jan 21 11:09:39] VERBOSE[5030] rtp.c:     -- Packet2Packet bridging SIP/8001-00000010 and OOH323/172.16.67.253:1720-b7d14370
    Après il raccroche après 3", certainement à cause du warning chan_ooh323.c

    Maintenant il faudrait creuser pourquoi.
    Soit un problème de conf Asterisk, soit un probème de config 4400.

    Essayes de glaner quelques chose sur google.
    Il y a aussi ce post dans se forum avec le même problème
    http://www.elastix.org/es/component/kun ... -h323.html
    Essayes de faire un UP dessus pour savoir si la personne a résolu son problème.
     
  10. scloeza

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    Danardf Bonjour, bonjour.

    Le dernier lien je l'ai mis dans le forum, c'est à dire le dernier commentaire

    Je crois que je peux bouger les paramètres de la RTP d'Elastix et où je peux.

    Et notez que le Alcatel utilise h323 version 2 et la version 4 asteris poignées.

    Notez que j'ai une voix passerelle marque modèle Quintum Tenor AS400 et qu'il fonctionne correctement téléphone analogique <-> AS400 <-> Asterisk <-> Xlite ou Bria, avec G729 fonctionne correctement dans les deux sens et ne pas couper, Je vois l'AS400 a la version 3 du h323 et comme je le disais la version astérisque 4.

    C'est pourquoi je est traumatisé cette situation comme une équipe si cela fonctionne et l'autre pas, doit être quelque chose de très simple et je ne peux pas le prendre.

    Danardf Sérieusement, j'apprécie beaucoup votre soutien et intérêt, je pense qu'il serait m'aider à le résoudre

    Salutations
     
  11. danardf

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    Hi.
    This French is bad enough, sorry. But thanks for your effort :)
    Even if my english isn't perfect too. :laugh:

    Well,

    Alcatel 4400 use a range RPT port (from 32512 to 327xx or something like that).
    You can try to change every parameters within rtp.conf file.

    Else, I'll try to make some test this week, if i've some free time.
    But, continue to find a way.

    Regards
     
  12. scloeza

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    Thank's, I think if we understand better in English.

    Look, anybody do this morning all the changes in my equipment and nothing, the thing remains the same, I spent the whole day in google and nothing. H323 learned more thing but I find nothing more.

    He concluded that Alcatel is seeking the pámetros TerminalCapabilitySet (TCS) and the asterisk does not send. This I saw in a document which details the wireshark H245 flow. sessions with h323.
    http://www.voipforo.com/H323/H323ejemplo.php

    What I do notice is that the packet capture I made from Quintum (Tenor AS400) and asterisk all the steps perfectly fulfilled as indicated by the opening

    Now, I do some test disabling this option in Alcatel, think you can do something about it.

    Thanks for all your time.
     
  13. danardf

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    Like you use R5.0Ux version, I hope that if i can do it in R9.0, you will do it with your version.
    Lots of changes are there in R9, R9.1.
     
  14. scloeza

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    I hope that if you can, you have an idea or procedure to follow to disable this function.

    Thanks
     
  15. danardf

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    First test, but not completed, sorry because i've the communication from only one direction.
    OXE -> Elastix. But nothing Elastix -> OXE ...Grrrrr :S
    But the good news is, I can make a call up more one minute. But only in one sens. :lol:

    I hope that could make a call in both sens.
     
  16. scloeza

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    This is what I mentioned before, does not work elastix -> alcatel, only works 3 seconds and cut the call. The setup I have is that you paste into one of the above comments.

    I hope you make it, well I still do try, I am now preparing a new lab and I think down the oh323 not ooh323 and install it and compile it, I hope so.

    Greetings and luck,
     
  17. danardf

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    Maybe it's the reason.

    However, don't forget to put bindaddr=your_elastix_ip instead of 0.0.0.0
    It's very important.
     
  18. scloeza

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    Les comento, logre realizar la integración entre Elastix y Alcatel utilizando como artificio un Asterisk montado en Ubuntu 10.04 , ya que el h323 que trae integrado funciona a la perfección con Alcatel.

    Saludos espero les sirva.
     
  19. marzo

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    Podrias describir la configuración que hiciste en Elastix ?

    Que configuración fue necesario realizar en la planta Alcatel ?
     

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