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bazil112014-03-24 14:31:14
Asterisk
bazil11, 2014-03-24 14:31:14

Why are calls from internal to internal not recorded?

Asterisk 11.2.1. Configured via extensions.conf. I rummaged through everything I could, but apparently, for a complete understanding, I do not have enough practice.

localhost*CLI> dialplan show macro-stdexten
[ Context 'macro-stdexten' created by 'pbx_config' ]
  'a' =>            1. VoicemailMain(${ORIG_ARG1})                [pbx_config]
  's' =>            1. MixMonitor(/callsdata/${UNIQUEID}.wav49)   [pbx_config]
                    2. Set(__DYNAMIC_FEATURES=${FEATURES})        [pbx_config]
                    3. Set(ORIG_ARG1=${ARG1})                     [pbx_config]
                    4. GotoIf($["${FOLLOWME_${ARG1}}" = "1"]?6:4) [pbx_config]
                    5. Dial(${ARG2},${RINGTIME},${DIALOPTIONS})   [pbx_config]
                    6. Goto(s-${DIALSTATUS},1)                    [pbx_config]
                    7. Macro(stdexten-followme,${ARG1},${ARG2})   [pbx_config]
  's-BUSY' =>       1. Voicemail(${ORIG_ARG1},b)                  [pbx_config]
                    2. Goto(default,s,1)                          [pbx_config]
  's-NOANSWER' =>   1. Voicemail(${ORIG_ARG1},u)                  [pbx_config]
                    2. Goto(default,s,1)                          [pbx_config]
  '_s-.' =>         1. Goto(s-NOANSWER,1)                         [pbx_config]

-= 5 extensions (13 priorities) in 1 context. =-

Logically, the presence of a line in the dialplan clearly announces the call to the mixmonitor, but this does not happen (started Aster in -rvvvvvvvvvvvvvvvvvvvvvvvv, calls from outside and outside are written - in the logs there is a line mixmonitor started recording, finished recording, calls inside are not written).
Tell me, pliz, where to dig further? Perhaps there is some clever way to catch where the error is.

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2 answer(s)
R
Rsa97, 2014-03-25
@bazil11

And what does the console show when you call from internal to internal phones? Can you show the dialplan and console output?

R
Rsa97, 2014-03-24
@Rsa97

Check the canreinvite option in sip.conf or users.conf. If it is set to yes for both users, then after the initial connection, Asterisk offers ip-phones to connect directly, of course, the recording does not work.

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