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спросил 2012-04-25 13:29:22 +0400

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Mixmonitor валит беседу с ошибкой Read factory .. and write factory .. both fail to provide 160 samples

Возникла необходимость записать диалог. После выполнения команды mixmonitor start SIP/channel /tmp/file.wav разговор завершается с нормальным статусом, в консоли ничего:

*CLI> mixmonitor start SIP/1-0000008b test.wav
  == Begin MixMonitor Recording SIP/1-0000008b
  == Spawn extension (default, NUM, 1) exited non-zero on 'SIP/1-0000008b'
  == End MixMonitor Recording SIP/1-0000008b

Включил дебаг режим и вот что там обнаружилось:

[Apr 25 12:43:29] VERBOSE[12029] app_mixmonitor.c:   == Begin MixMonitor Recording SIP/sipnet-00000086
[Apr 25 12:43:29] DEBUG[12029] audiohook.c: Read factory 0x8c09e58 and write factory 0x8c0a880 both fail to provide 160 samples
[Apr 25 12:43:29] DEBUG[12028] channel.c: Hanging up channel 'SIP/sipnet-00000086'
[Apr 25 12:43:29] DEBUG[12028] chan_sip.c: Hangup call SIP/sipnet-00000086, SIP callid 2343df1d02ae88a817686e142e9e92ca@sipnet.ru
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8b9e158'
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8ba1860'
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8a81600'
[Apr 25 12:43:29] DEBUG[12028] netsock2.c: Splitting '212.53.40.40:5060' gives...
[Apr 25 12:43:29] DEBUG[12028] netsock2.c: ...host '212.53.40.40' and port '5060'.
[Apr 25 12:43:29] DEBUG[12028] chan_sip.c: Trying to put 'BYE sip:pro' onto UDP socket destined for 212.53.40.40:5060
[Apr 25 12:43:29] DEBUG[12028] app_dial.c: Exiting with DIALSTATUS=ANSWER.
[Apr 25 12:43:29] DEBUG[12028] pbx.c: Spawn extension (default,89035218914,1) exited non-zero on 'SIP/1-00000085'
[Apr 25 12:43:29] VERBOSE[12028] pbx.c:   == Spawn extension (default, 89035218914, 1) exited non-zero on 'SIP/1-00000085'
[Apr 25 12:43:29] DEBUG[12028] channel.c: Soft-Hanging up channel 'SIP/1-00000085'
[Apr 25 12:43:29] DEBUG[12028] channel.c: Hanging up channel 'SIP/1-00000085'
[Apr 25 12:43:29] DEBUG[12028] chan_sip.c: Hangup call SIP/1-00000085, SIP callid 3024246226@192_168_10_2
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8aa7000'
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8bfb518'
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8728988'
[Apr 25 12:43:29] DEBUG[12028] netsock2.c: Splitting '10.10.10.10:8829' gives...
[Apr 25 12:43:29] DEBUG[12028] netsock2.c: ...host '10.10.10.10' and port '8829'.
[Apr 25 12:43:29] DEBUG[12028] chan_sip.c: Trying to put 'BYE sip:1@8' onto UDP socket destined for 89.169.180.214:8829
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: No provider found, checking channel drivers for SIP - 1
[Apr 25 12:43:29] DEBUG[1904] chan_sip.c: Checking device state for peer 1
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: Changing state for SIP/1 - state 1 (Not in use)
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: device 'SIP/1' state '1'
[Apr 25 12:43:29] DEBUG[1903] app_queue.c: Device 'SIP/1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Apr 25 12:43:29] DEBUG[12029] autochan.c: Removed autochan 0x8ba7318 from the list, about to free it
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: No provider found, checking channel drivers for SIP - sipnet
[Apr 25 12:43:29] DEBUG[1904] chan_sip.c: Checking device state for peer sipnet
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: Changing state for SIP/sipnet - state 1 (Not in use)
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: device 'SIP/sipnet' state '1'
[Apr 25 12:43:29] DEBUG[1903] app_queue.c: Device 'SIP/sipnet' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Apr 25 12:43:29] VERBOSE[12029] app_mixmonitor.c:   == End MixMonitor Recording SIP/sipnet-00000086

и канал закрывается после первых строк

[Apr 25 12:43:29] DEBUG[12029] audiohook.c: Read factory 0x8c09e58 and write factory 0x8c0a880 both fail to provide 160 samples
[Apr 25 12:43:29] DEBUG[12028] channel.c: Hanging up channel 'SIP/sipnet-00000086'

Как то понять в чем проблема так и не удалось.. почитал исходный код, ошибка возникает тут (music/audiohook.c):

static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audiohook, size_t samples)
{
        int i = 0, usable_read, usable_write;
        short buf1[samples], buf2[samples], *read_buf = NULL, *write_buf = NULL, *final_buf = NULL, *data1 = NULL, *data2 = NULL;
        struct ast_frame frame = {
                .frametype = AST_FRAME_VOICE,
                .subclass.codec = AST_FORMAT_SLINEAR,
                .data.ptr = NULL,
                .datalen = sizeof(buf1),
                .samples = samples,
        };

        /* Make sure both factories have the required samples */
        usable_read = (ast_slinfactory_available(&audiohook->read_factory) >= samples ? 1 : 0);
        usable_write = (ast_slinfactory_available(&audiohook->write_factory) >= samples ? 1 : 0);

        if (!usable_read && !usable_write) {
                /* If both factories are unusable bail out */
                ast_debug(1, "Read factory %p and write factory %p both fail to provide %zd samples\n", &audiohook->read_factory, &audiohook->write_factory, samples);
                return NULL;
        }

Mixmonitor валит беседу с ошибкой Read factory .. and write factory .. both fail to provide 160 samples

Возникла необходимость записать диалог. После выполнения команды mixmonitor start SIP/channel /tmp/file.wav разговор завершается с нормальным статусом, в консоли ничего:

*CLI> mixmonitor start SIP/1-0000008b test.wav
  == Begin MixMonitor Recording SIP/1-0000008b
  == Spawn extension (default, NUM, 1) exited non-zero on 'SIP/1-0000008b'
  == End MixMonitor Recording SIP/1-0000008b

Включил дебаг режим и вот что там обнаружилось:

[Apr 25 12:43:29] VERBOSE[12029] app_mixmonitor.c:   == Begin MixMonitor Recording SIP/sipnet-00000086
[Apr 25 12:43:29] DEBUG[12029] audiohook.c: Read factory 0x8c09e58 and write factory 0x8c0a880 both fail to provide 160 samples
[Apr 25 12:43:29] DEBUG[12028] channel.c: Hanging up channel 'SIP/sipnet-00000086'
[Apr 25 12:43:29] DEBUG[12028] chan_sip.c: Hangup call SIP/sipnet-00000086, SIP callid 2343df1d02ae88a817686e142e9e92ca@sipnet.ru
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8b9e158'
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8ba1860'
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8a81600'
[Apr 25 12:43:29] DEBUG[12028] netsock2.c: Splitting '212.53.40.40:5060' gives...
[Apr 25 12:43:29] DEBUG[12028] netsock2.c: ...host '212.53.40.40' and port '5060'.
[Apr 25 12:43:29] DEBUG[12028] chan_sip.c: Trying to put 'BYE sip:pro' onto UDP socket destined for 212.53.40.40:5060
[Apr 25 12:43:29] DEBUG[12028] app_dial.c: Exiting with DIALSTATUS=ANSWER.
[Apr 25 12:43:29] DEBUG[12028] pbx.c: Spawn extension (default,89035218914,1) exited non-zero on 'SIP/1-00000085'
[Apr 25 12:43:29] VERBOSE[12028] pbx.c:   == Spawn extension (default, 89035218914, 1) exited non-zero on 'SIP/1-00000085'
[Apr 25 12:43:29] DEBUG[12028] channel.c: Soft-Hanging up channel 'SIP/1-00000085'
[Apr 25 12:43:29] DEBUG[12028] channel.c: Hanging up channel 'SIP/1-00000085'
[Apr 25 12:43:29] DEBUG[12028] chan_sip.c: Hangup call SIP/1-00000085, SIP callid 3024246226@192_168_10_2
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8aa7000'
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8bfb518'
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8728988'
[Apr 25 12:43:29] DEBUG[12028] netsock2.c: Splitting '10.10.10.10:8829' gives...
[Apr 25 12:43:29] DEBUG[12028] netsock2.c: ...host '10.10.10.10' and port '8829'.
[Apr 25 12:43:29] DEBUG[12028] chan_sip.c: Trying to put 'BYE sip:1@8' onto UDP socket destined for 89.169.180.214:8829
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: No provider found, checking channel drivers for SIP - 1
[Apr 25 12:43:29] DEBUG[1904] chan_sip.c: Checking device state for peer 1
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: Changing state for SIP/1 - state 1 (Not in use)
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: device 'SIP/1' state '1'
[Apr 25 12:43:29] DEBUG[1903] app_queue.c: Device 'SIP/1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Apr 25 12:43:29] DEBUG[12029] autochan.c: Removed autochan 0x8ba7318 from the list, about to free it
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: No provider found, checking channel drivers for SIP - sipnet
[Apr 25 12:43:29] DEBUG[1904] chan_sip.c: Checking device state for peer sipnet
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: Changing state for SIP/sipnet - state 1 (Not in use)
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: device 'SIP/sipnet' state '1'
[Apr 25 12:43:29] DEBUG[1903] app_queue.c: Device 'SIP/sipnet' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Apr 25 12:43:29] VERBOSE[12029] app_mixmonitor.c:   == End MixMonitor Recording SIP/sipnet-00000086

и канал закрывается после первых строк

[Apr 25 12:43:29] DEBUG[12029] audiohook.c: Read factory 0x8c09e58 and write factory 0x8c0a880 both fail to provide 160 samples
[Apr 25 12:43:29] DEBUG[12028] channel.c: Hanging up channel 'SIP/sipnet-00000086'

Как то понять в чем проблема так и не удалось.. почитал исходный код, ошибка возникает тут (music/audiohook.c):

static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audiohook, size_t samples)
{
        int i = 0, usable_read, usable_write;
        short buf1[samples], buf2[samples], *read_buf = NULL, *write_buf = NULL, *final_buf = NULL, *data1 = NULL, *data2 = NULL;
        struct ast_frame frame = {
                .frametype = AST_FRAME_VOICE,
                .subclass.codec = AST_FORMAT_SLINEAR,
                .data.ptr = NULL,
                .datalen = sizeof(buf1),
                .samples = samples,
        };

        /* Make sure both factories have the required samples */
        usable_read = (ast_slinfactory_available(&audiohook->read_factory) >= samples ? 1 : 0);
        usable_write = (ast_slinfactory_available(&audiohook->write_factory) >= samples ? 1 : 0);

        if (!usable_read && !usable_write) {
                /* If both factories are unusable bail out */
                ast_debug(1, "Read factory %p and write factory %p both fail to provide %zd samples\n", &audiohook->read_factory, &audiohook->write_factory, samples);
                return NULL;
        }

inconn*CLI> core show translation
         Translation times between formats (in microseconds) for one second of data
          Source Format (Rows) Destination Format (Columns)

           g723   gsm  ulaw  alaw g726aal2 adpcm  slin lpc10  g729 speex  ilbc  g726  g722 siren7 siren14 slin16  g719 speex16 testlaw
     g723     -  2000  1001  1001     3000  1001  1000  4000  5000  7999  8999  3000  1001   5001    4000   2001     -    7000    1001
      gsm  5998     -  1000  1000     2999  1000   999  3999  4999  7998  8998  2999  1000   5000    3999   2000     -    6999    1000
     ulaw  5000  1001     -     1     2001     2     1  3001  4001  7000  8000  2001     2   4002    3001   1002     -    6001       2
     alaw  5000  1001     1     -     2001     2     1  3001  4001  7000  8000  2001     2   4002    3001   1002     -    6001       2
 g726aal2  5999  2000  1001  1001        -  1001  1000  4000  5000  7999  8999  3000  1001   5001    4000   2001     -    7000    1001
    adpcm  5000  1001     2     2     2001     -     1  3001  4001  7000  8000  2001     2   4002    3001   1002     -    6001       2
     slin  4999  1000     1     1     2000     1     -  3000  4000  6999  7999  2000     1   4001    3000   1001     -    6000       1
    lpc10  5999  2000  1001  1001     3000  1001  1000     -  5000  7999  8999  3000  1001   5001    4000   2001     -    7000    1001
     g729  5999  2000  1001  1001     3000  1001  1000  4000     -  7999  8999  3000  1001   5001    4000   2001     -    7000    1001
    speex  5999  2000  1001  1001     3000  1001  1000  4000  5000     -  8999  3000  1001   5001    4000   2001     -    7000    1001
     ilbc  5998  1999  1000  1000     2999  1000   999  3999  4999  7998     -  2999  1000   5000    3999   2000     -    6999    1000
     g726  5999  2000  1001  1001     3000  1001  1000  4000  5000  7999  8999     -  1001   5001    4000   2001     -    7000    1001
     g722  5000  1001     2     2     2001     2     1  3001  4001  7000  8000  2001     -   4000    3999   1000     -    5999       2
   siren7  9999  6000  5001  5001     7000  5001  5000  8000  9000 11999 12999  7000  4999      -    5998   2999     -    7998    5001
  siren14 15998 11999 11000 11000    12999 11000 10999 13999 14999 17998 18998 12999  7999   8999       -   5999     -   10998   11000
   slin16  7000  3001  2002  2002     4001  2002  2001  5001  6001  9000 10000  4001  2000   3000    2999      -     -    4999    2002
     g719     -     -     -     -        -     -     -     -     -     -     -     -     -      -       -      -     -       -       -
  speex16  8000  4001  3002  3002     5001  3002  3001  6001  7001 10000 11000  5001  3000   4000    3999   1000     -       -    3002
  testlaw  5000  1001     2     2     2001     2     1  3001  4001  7000  8000  2001     2   4002    3001   1002     -    6001       -

Mixmonitor валит беседу с ошибкой Read factory .. and write factory .. both fail to provide 160 samples

Возникла необходимость записать диалог. После выполнения команды mixmonitor start SIP/channel /tmp/file.wav разговор завершается с нормальным статусом, в консоли ничего:

*CLI> mixmonitor start SIP/1-0000008b test.wav
  == Begin MixMonitor Recording SIP/1-0000008b
  == Spawn extension (default, NUM, 1) exited non-zero on 'SIP/1-0000008b'
  == End MixMonitor Recording SIP/1-0000008b

Включил дебаг режим и вот что там обнаружилось:

[Apr 25 12:43:29] VERBOSE[12029] app_mixmonitor.c:   == Begin MixMonitor Recording SIP/sipnet-00000086
[Apr 25 12:43:29] DEBUG[12029] audiohook.c: Read factory 0x8c09e58 and write factory 0x8c0a880 both fail to provide 160 samples
[Apr 25 12:43:29] DEBUG[12028] channel.c: Hanging up channel 'SIP/sipnet-00000086'
[Apr 25 12:43:29] DEBUG[12028] chan_sip.c: Hangup call SIP/sipnet-00000086, SIP callid 2343df1d02ae88a817686e142e9e92ca@sipnet.ru
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8b9e158'
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8ba1860'
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8a81600'
[Apr 25 12:43:29] DEBUG[12028] netsock2.c: Splitting '212.53.40.40:5060' gives...
[Apr 25 12:43:29] DEBUG[12028] netsock2.c: ...host '212.53.40.40' and port '5060'.
[Apr 25 12:43:29] DEBUG[12028] chan_sip.c: Trying to put 'BYE sip:pro' onto UDP socket destined for 212.53.40.40:5060
[Apr 25 12:43:29] DEBUG[12028] app_dial.c: Exiting with DIALSTATUS=ANSWER.
[Apr 25 12:43:29] DEBUG[12028] pbx.c: Spawn extension (default,89035218914,1) exited non-zero on 'SIP/1-00000085'
[Apr 25 12:43:29] VERBOSE[12028] pbx.c:   == Spawn extension (default, 89035218914, 1) exited non-zero on 'SIP/1-00000085'
[Apr 25 12:43:29] DEBUG[12028] channel.c: Soft-Hanging up channel 'SIP/1-00000085'
[Apr 25 12:43:29] DEBUG[12028] channel.c: Hanging up channel 'SIP/1-00000085'
[Apr 25 12:43:29] DEBUG[12028] chan_sip.c: Hangup call SIP/1-00000085, SIP callid 3024246226@192_168_10_2
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8aa7000'
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8bfb518'
[Apr 25 12:43:29] DEBUG[12028] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8728988'
[Apr 25 12:43:29] DEBUG[12028] netsock2.c: Splitting '10.10.10.10:8829' gives...
[Apr 25 12:43:29] DEBUG[12028] netsock2.c: ...host '10.10.10.10' and port '8829'.
[Apr 25 12:43:29] DEBUG[12028] chan_sip.c: Trying to put 'BYE sip:1@8' onto UDP socket destined for 89.169.180.214:8829
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: No provider found, checking channel drivers for SIP - 1
[Apr 25 12:43:29] DEBUG[1904] chan_sip.c: Checking device state for peer 1
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: Changing state for SIP/1 - state 1 (Not in use)
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: device 'SIP/1' state '1'
[Apr 25 12:43:29] DEBUG[1903] app_queue.c: Device 'SIP/1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Apr 25 12:43:29] DEBUG[12029] autochan.c: Removed autochan 0x8ba7318 from the list, about to free it
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: No provider found, checking channel drivers for SIP - sipnet
[Apr 25 12:43:29] DEBUG[1904] chan_sip.c: Checking device state for peer sipnet
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: Changing state for SIP/sipnet - state 1 (Not in use)
[Apr 25 12:43:29] DEBUG[1904] devicestate.c: device 'SIP/sipnet' state '1'
[Apr 25 12:43:29] DEBUG[1903] app_queue.c: Device 'SIP/sipnet' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Apr 25 12:43:29] VERBOSE[12029] app_mixmonitor.c:   == End MixMonitor Recording SIP/sipnet-00000086

и канал закрывается после первых строк

[Apr 25 12:43:29] DEBUG[12029] audiohook.c: Read factory 0x8c09e58 and write factory 0x8c0a880 both fail to provide 160 samples
[Apr 25 12:43:29] DEBUG[12028] channel.c: Hanging up channel 'SIP/sipnet-00000086'

Как то понять в чем проблема так и не удалось.. почитал исходный код, ошибка возникает тут (music/audiohook.c):

static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audiohook, size_t samples)
{
        int i = 0, usable_read, usable_write;
        short buf1[samples], buf2[samples], *read_buf = NULL, *write_buf = NULL, *final_buf = NULL, *data1 = NULL, *data2 = NULL;
        struct ast_frame frame = {
                .frametype = AST_FRAME_VOICE,
                .subclass.codec = AST_FORMAT_SLINEAR,
                .data.ptr = NULL,
                .datalen = sizeof(buf1),
                .samples = samples,
        };

        /* Make sure both factories have the required samples */
        usable_read = (ast_slinfactory_available(&audiohook->read_factory) >= samples ? 1 : 0);
        usable_write = (ast_slinfactory_available(&audiohook->write_factory) >= samples ? 1 : 0);

        if (!usable_read && !usable_write) {
                /* If both factories are unusable bail out */
                ast_debug(1, "Read factory %p and write factory %p both fail to provide %zd samples\n", &audiohook->read_factory, &audiohook->write_factory, samples);
                return NULL;
        }

Если посмотреть список кодеков, то:

inconn*CLI> core show translation
         Translation times between formats (in microseconds) for one second of data
          Source Format (Rows) Destination Format (Columns)

           g723   gsm  ulaw  alaw g726aal2 adpcm  slin lpc10  g729 speex  ilbc  g726  g722 siren7 siren14 slin16  g719 speex16 testlaw
     g723     -  2000  1001  1001     3000  1001  1000  4000  5000  7999  8999  3000  1001   5001    4000   2001     -    7000    1001
      gsm  5998     -  1000  1000     2999  1000   999  3999  4999  7998  8998  2999  1000   5000    3999   2000     -    6999    1000
     ulaw  5000  1001     -     1     2001     2     1  3001  4001  7000  8000  2001     2   4002    3001   1002     -    6001       2
     alaw  5000  1001     1     -     2001     2     1  3001  4001  7000  8000  2001     2   4002    3001   1002     -    6001       2
 g726aal2  5999  2000  1001  1001        -  1001  1000  4000  5000  7999  8999  3000  1001   5001    4000   2001     -    7000    1001
    adpcm  5000  1001     2     2     2001     -     1  3001  4001  7000  8000  2001     2   4002    3001   1002     -    6001       2
     slin  4999  1000     1     1     2000     1     -  3000  4000  6999  7999  2000     1   4001    3000   1001     -    6000       1
    lpc10  5999  2000  1001  1001     3000  1001  1000     -  5000  7999  8999  3000  1001   5001    4000   2001     -    7000    1001
     g729  5999  2000  1001  1001     3000  1001  1000  4000     -  7999  8999  3000  1001   5001    4000   2001     -    7000    1001
    speex  5999  2000  1001  1001     3000  1001  1000  4000  5000     -  8999  3000  1001   5001    4000   2001     -    7000    1001
     ilbc  5998  1999  1000  1000     2999  1000   999  3999  4999  7998     -  2999  1000   5000    3999   2000     -    6999    1000
     g726  5999  2000  1001  1001     3000  1001  1000  4000  5000  7999  8999     -  1001   5001    4000   2001     -    7000    1001
     g722  5000  1001     2     2     2001     2     1  3001  4001  7000  8000  2001     -   4000    3999   1000     -    5999       2
   siren7  9999  6000  5001  5001     7000  5001  5000  8000  9000 11999 12999  7000  4999      -    5998   2999     -    7998    5001
  siren14 15998 11999 11000 11000    12999 11000 10999 13999 14999 17998 18998 12999  7999   8999       -   5999     -   10998   11000
   slin16  7000  3001  2002  2002     4001  2002  2001  5001  6001  9000 10000  4001  2000   3000    2999      -     -    4999    2002
     g719     -     -     -     -        -     -     -     -     -     -     -     -     -      -       -      -     -       -       -
  speex16  8000  4001  3002  3002     5001  3002  3001  6001  7001 10000 11000  5001  3000   4000    3999   1000     -       -    3002
  testlaw  5000  1001     2     2     2001     2     1  3001  4001  7000  8000  2001     2   4002    3001   1002     -    6001       -
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Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.