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Discussion: Appels se coupe aprés 10 min

  1. #1
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    Appels se coupe aprés 10 min

    Bonjour,

    nous avons un IPBX AVAYA relié à un serveur Asterisk 10.4.0, nous avons un problème avec les appels qui coupent tous seuls au bout de quelques minutes, 10 minutes la plupart du temps.

    j'ai lancé un Asterisk -r et j'ai remarqué que le message ci-dessous s'affiche lors de la coupure :

    [Jun 21 10:27:14] NOTICE[9480]: chan_sip.c:23340 handle_request_invite: Call from '' (196.168.9.250:18747) to extension '1011972597634083' rejected because extension not found in context 'default'.
    == Spawn extension (default, 501533146012884, 1) exited non-zero on 'H323/ip$192.168.9.12:17437/27316'
    [Jun 21 10:27:46] WARNING[9480]: chan_sip.c:3663 retrans_pkt: Retransmission timeout reached on transmission 846d7000234012a018784a0004bef70c for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/displ...etransmissions
    Packet timed out after 32000ms with no response

    prière de m'aider s'il vous plaît.

    Merci d'avance.

  2. #2
    Membre Senior
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    janvier 2011
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    Villejuif 94
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    Bonjour,



    Citation Envoyé par espacio Voir le message
    [Jun 21 10:27:14] NOTICE[9480]: chan_sip.c:23340 handle_request_invite: Call from '' (196.168.9.250:18747) to extension '1011972597634083' rejected because extension not found in context 'default'.
    Ton serveur est-il ouvert sur l'extérieur ? cela ressemble beaucoup à un scan ;whois indique une IP au Togo !

    == Spawn extension (default, 501533146012884, 1) exited non-zero on 'H323/ip$192.168.9.12:17437/27316'
    Ta liaison avec l'AVAYA est en H323 ? Il faudrait le code de retours "non-zero". Essayes un "core set verbose 5" dans la console.


    [Jun 21 10:27:46] WARNING[9480]: chan_sip.c:3663 retrans_pkt: Retransmission timeout reached on transmission 846d7000234012a018784a0004bef70c for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/displ...etransmissions
    Packet timed out after 32000ms with no response
    relatif au premier message (Jun 21 10:27:46 - Jun 21 10:27:14) = 32s ;confirme un scan ;les bots ne "hack" pas" !

  3. #3
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    Bonjour,


    Ton serveur est-il ouvert sur l'extérieur ? cela ressemble beaucoup à un scan ;whois indique une IP au Togo !

    Mon serveur est ouvert pour les ports 5060-5062 et 10000-20000 et seulement en provenance des adresses IP de mon opérateur SIP.


    Ta liaison avec l'AVAYA est en H323 ? Il faudrait le code de retours "non-zero". Essayes un "core set verbose 5" dans la console.

    Oui, ma liaison avec l'AVAYA est en H323, comment puis-je avoir le code de retours "non-zero", j'ai essayé le "core set verbose 5" voir les traces générés aprés :

    Code:
    h323gw*CLI> core set verbose 5
    Verbosity was 3 and is now 5
      == Using SIP RTP CoS mark 5
        -- Executing [33184888248@default:1] Dial("SIP/sbc.manivox.com-00001979", "H323/3501@avaya") in new stack
        -- Requested transfer capability: 0x00 - SPEECH
        -- Called H323/3501@avaya
        -- H323/avaya-4595 is making progress passing it to SIP/sbc.manivox.com-00001979
        -- H323/avaya-4595 is ringing
        -- H323/avaya-4595 answered SIP/sbc.manivox.com-00001979
      == Using SIP RTP CoS mark 5
    [Jun 21 17:08:32] NOTICE[9480]: chan_sip.c:23340 handle_request_invite: Call from '' (196.168.9.250:19585) to extension '900970567218447' rejected because extension not found in context 'default'.
    [Jun 21 17:09:04] WARNING[9480]: chan_sip.c:3663 retrans_pkt: Retransmission timeout reached on transmission 65b5a12531bccb6c85aff1d672ab43fa for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 32000ms with no response
      == Spawn extension (default, 33184888248, 1) exited non-zero on 'SIP/sbc.manivox.com-00001979'
      == Spawn extension (default, 33184888248, 1) exited non-zero on 'SIP/sbc.manivox.com-00001977'
      == Using SIP RTP CoS mark 5
        -- Executing [33184888245@default:1] Dial("SIP/sbc.manivox.com-0000197a", "H323/3503@avaya") in new stack
        -- Requested transfer capability: 0x00 - SPEECH
        -- Called H323/3503@avaya
        -- H323/avaya-4596 is making progress passing it to SIP/sbc.manivox.com-0000197a
        -- H323/avaya-4596 is ringing
        -- H323/avaya-4596 answered SIP/sbc.manivox.com-0000197a
      == Spawn extension (default, 33184888245, 1) exited non-zero on 'SIP/sbc.manivox.com-0000197a'
      == Using SIP RTP CoS mark 5
    [Jun 21 17:12:29] NOTICE[9480]: chan_sip.c:23340 handle_request_invite: Call from '' (196.168.9.250:19589) to extension '666011972597634083' rejected because extension not found in context 'default'.
      == Using SIP RTP CoS mark 5
        -- Executing [33184888245@default:1] Dial("SIP/sbc.manivox.com-0000197b", "H323/3503@avaya") in new stack
        -- Requested transfer capability: 0x00 - SPEECH
        -- Called H323/3503@avaya
        -- H323/avaya-4597 is making progress passing it to SIP/sbc.manivox.com-0000197b
        -- H323/avaya-4597 is ringing
      == Using SIP RTP CoS mark 5
        -- Executing [33184888245@default:1] Dial("SIP/sbc.manivox.com-0000197c", "H323/3503@avaya") in new stack
        -- Requested transfer capability: 0x00 - SPEECH
        -- Called H323/3503@avaya
        -- H323/avaya-4598 is making progress passing it to SIP/sbc.manivox.com-0000197c
        -- H323/avaya-4598 is ringing
      == Spawn extension (default, 33184888245, 1) exited non-zero on 'SIP/sbc.manivox.com-0000197b'
    [Jun 21 17:13:01] WARNING[9480]: chan_sip.c:3663 retrans_pkt: Retransmission timeout reached on transmission 17cf2b27557ac7c0190fbdd3669983c5 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 32001ms with no response
        -- H323/avaya-4598 answered SIP/sbc.manivox.com-0000197c
      == Spawn extension (default, 33184888245, 1) exited non-zero on 'SIP/sbc.manivox.com-0000197c'
      == Using SIP RTP CoS mark 5
        -- Executing [33184888245@default:1] Dial("SIP/sbc.manivox.com-0000197d", "H323/3503@avaya") in new stack
        -- Requested transfer capability: 0x00 - SPEECH
        -- Called H323/3503@avaya
        -- H323/avaya-4599 is making progress passing it to SIP/sbc.manivox.com-0000197d
        -- H323/avaya-4599 is ringing
      == Using SIP RTP CoS mark 5
    [Jun 21 17:14:12] NOTICE[9480]: chan_sip.c:23340 handle_request_invite: Call from '' (196.168.9.250:19597) to extension '0972598126387' rejected because extension not found in context 'default'.
        -- H323/avaya-4599 answered SIP/sbc.manivox.com-0000197d
    [Jun 21 17:14:44] WARNING[9480]: chan_sip.c:3663 retrans_pkt: Retransmission timeout reached on transmission e1ec36c4c7cd81b3f91171cc21eb7270 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 32000ms with no response
    h323gw*CLI>
    Merci beaucoup par avance.

  4. #4
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    Bonjour,
    Tout d'abord je m'excuse du retard à vous répondre et vous remercie de l’intérêt accordé à mon problème.

    Pour la première question, oui mon serveur est ouvert à l’extérieur pour les ports 5060-5062 et 10000-20000 en provenance de mon prestataire SIP seulement. ma localisation est le Maroc et non pas le Togo.

    Pour votre deuxième question, oui, ma liaison avec AVAYA est en H323, voir ci-dessous pour les traces demandées :

    Code:
    Connected to Asterisk 10.4.0 currently running on h323gw (pid = 2276)
    Verbosity is at least 3
    h323gw*CLI> core set verbose 5
    Verbosity was 3 and is now 5
        -- SIP/mani-000000bb is ringing
      == Spawn extension (default, 501433479424275, 1) exited non-zero on 'H323/ip$1                                                                                                                     92.168.9.12:25073/29140'
      == Spawn extension (default, 33184888249, 1) exited non-zero on 'SIP/sbc.manivox.com-000000b9'
        -- SIP/mani-000000bb answered H323/ip$192.168.9.12:25077/29243
        -- Executing [501133608276179@default:1] Dial("H323/ip$192.168.9.12:25078/29348", "SIP/mani/501133608276179") in new stack
      == Using SIP RTP CoS mark 5
        -- Called SIP/mani/501133608276179
        -- SIP/mani-000000bc is making progress passing it to H323/ip$192.168.9.12:25078/29348
        -- SIP/mani-000000bc is ringing
      == Spawn extension (default, 501133685545000, 1) exited non-zero on 'H323/ip$192.168.9.12:25077/29243'
        -- SIP/mani-000000bc answered H323/ip$192.168.9.12:25078/29348
      == Spawn extension (default, 501133608276179, 1) exited non-zero on 'H323/ip$192.168.9.12:25078/29348'
        -- Executing [501433479424049@default:1] Dial("H323/ip$192.168.9.12:25079/29455", "SIP/mani/501433479424049") in new stack
      == Using SIP RTP CoS mark 5
        -- Called SIP/mani/501433479424049
        -- SIP/mani-000000bd is making progress passing it to H323/ip$192.168.9.12:25079/29455
      == Spawn extension (default, 33184888249, 1) exited non-zero on 'SIP/sbc.manivox.com-000000b3'
      == Spawn extension (default, 33184888248, 1) exited non-zero on 'SIP/sbc.manivox.com-000000ae'
        -- Executing [501133149393714@default:1] Dial("H323/ip$192.168.9.12:25080/29564", "SIP/mani/501133149393714") in new stack
      == Using SIP RTP CoS mark 5
        -- Called SIP/mani/501133149393714
        -- SIP/mani-000000be is making progress passing it to H323/ip$192.168.9.12:25080/29564
        -- SIP/mani-000000be is ringing
      == Using SIP RTP CoS mark 5
        -- Executing [33184888245@default:1] Dial("SIP/sbc.manivox.com-000000bf", "H323/3503@avaya") in new stack
        -- Requested transfer capability: 0x00 - SPEECH
        -- Called H323/3503@avaya
        -- H323/avaya-136 is making progress passing it to SIP/sbc.manivox.com-000000bf
        -- H323/avaya-136 is ringing
        -- SIP/mani-000000be answered H323/ip$192.168.9.12:25080/29564
        -- SIP/mani-000000bd answered H323/ip$192.168.9.12:25079/29455
        -- H323/avaya-136 answered SIP/sbc.manivox.com-000000bf
      == Spawn extension (default, 501433479424049, 1) exited non-zero on 'H323/ip$192.168.9.12:25079/29455'
      == Spawn extension (default, 33184888245, 1) exited non-zero on 'SIP/sbc.manivox.com-000000b6'
      == Spawn extension (default, 33186260325, 1) exited non-zero on 'SIP/sbc.manivox.com-000000b8'
        -- Executing [501133629191395@default:1] Dial("H323/ip$192.168.9.12:25085/29675", "SIP/mani/501133629191395") in new stack
      == Using SIP RTP CoS mark 5
        -- Called SIP/mani/501133629191395
        -- SIP/mani-000000c0 is making progress passing it to H323/ip$192.168.9.12:25085/29675
        -- SIP/mani-000000c0 is ringing
      == Spawn extension (default, 501133149393714, 1) exited non-zero on 'H323/ip$192.168.9.12:25080/29564'
      == Spawn extension (default, 33184888248, 1) exited non-zero on 'SIP/sbc.manivox.com-000000b5'
        -- SIP/mani-000000c0 answered H323/ip$192.168.9.12:25085/29675
      == Using SIP RTP CoS mark 5
        -- Executing [33184888248@default:1] Dial("SIP/sbc.manivox.com-000000c1", "H323/3501@avaya") in new stack
        -- Requested transfer capability: 0x00 - SPEECH
        -- Executing [501433479426427@default:1] Dial("H323/ip$192.168.9.12:25086/29788", "SIP/mani/501433479426427") in new stack

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