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betaru

betaru

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  • Asterisk, рвутся звонки

    betaru
    @betaru Автор вопроса
    Вопрос закрыт, была сетевая проблема.
    Написано

  • Asterisk, рвутся звонки

    betaru
    @betaru Автор вопроса
    ну все логи у меня сохранены, могу любую информацию найти, [Nov 13 15:10:24] DEBUG[31363] channel.c: Got a FRAME_CONTROL (8) frame on channel DAHDI/i1/989*********-1010 [Nov 13 15:10:24] DEBUG[31363] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 13 15:10:24] DEBUG[31363] chan_dahdi.c: Requested indication 20 on channel DAHDI/i1/989*********-1010 [Nov 13 15:10:24] DEBUG[31363] channel.c: Bridge stops bridging channels SIP/117-00004715 and DAHDI/i1/989*********-1010 [Nov 13 15:10:24] DEBUG[31363] channel.c: Soft-Hanging up channel 'SIP/117-00004715' [Nov 13 15:10:24] DEBUG[31363] chan_dahdi.c: Disabled echo cancellation on channel 1 [Nov 13 15:10:24] DEBUG[31363] chan_dahdi.c: Set option TDD MODE, value: OFF(0) on DAHDI/i1/989*********-1010 [Nov 13 15:10:24] DEBUG[31363] chan_dahdi.c: Updated conferencing on 1, with 0 conference users [Nov 13 15:10:24] DEBUG[31363] chan_dahdi.c: Set option AUDIO MODE, value: OFF(0) on DAHDI/i1/989*********-1010 [Nov 13 15:10:24] VERBOSE[31363] chan_dahdi.c: -- Hungup 'DAHDI/i1/989*********-1010' [Nov 13 15:10:24] DEBUG[31363] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Nov 13 15:10:24] DEBUG[31363] pbx.c: Spawn extension (office,9989*********,3) exited non-zero on 'SIP/117-00004715' [Nov 13 15:10:24] VERBOSE[31363] pbx.c: == Spawn extension (office, 9989*********, 3) exited non-zero on 'SIP/117-00004715' [Nov 13 15:10:24] DEBUG[31363] channel.c: Soft-Hanging up channel 'SIP/117-00004715' [Nov 13 15:10:24] DEBUG[31363] channel.c: Hanging up channel 'SIP/117-00004715' [Nov 13 15:10:24] DEBUG[31363] chan_sip.c: Hanging up zombie call. Be scared. [Nov 13 15:10:24] DEBUG[31363] chan_sip.c: Updating call counter for incoming call [Nov 13 15:10:24] DEBUG[31363] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f27dc155f38' [Nov 13 15:10:24] DEBUG[31363] netsock2.c: Splitting '192.168.5.132:5060' into... [Nov 13 15:10:24] DEBUG[31363] netsock2.c: ...host '192.168.5.132' and port '5060'. [Nov 13 15:10:24] DEBUG[31363] chan_sip.c: Trying to put 'BYE sip:117' onto UDP socket destined for 192.168.5.132:5060
    Написано

  • Asterisk, рвутся звонки

    betaru
    @betaru Автор вопроса

    ссылка была битая, исправил :)

    Написано