[asterisk-bugs] [JIRA] (ASTERISK-25674) Mixmonitor stop recording after atxfer
Leandro Dardini (JIRA)
noreply at issues.asterisk.org
Wed Jan 13 15:42:33 CST 2016
[ https://issues.asterisk.org/jira/browse/ASTERISK-25674?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=228997#comment-228997 ]
Leandro Dardini commented on ASTERISK-25674:
--------------------------------------------
This is the log
{noformat}
[2016-01-13 22:29:05] VERBOSE[3263][C-000005b7] pbx.c: Executing [55512345 at authenticated:1] Set("SIP/104-DEVEL-0000001e", "__TRANSFER_CONTEXT=authenticated") in new stack
[2016-01-13 22:29:05] VERBOSE[3263][C-000005b7] pbx.c: Executing [55512345 at authenticated:2] Set("SIP/104-DEVEL-0000001e", "RECORDINGFORMAT=wav") in new stack
[2016-01-13 22:29:05] VERBOSE[3263][C-000005b7] pbx.c: Executing [55512345 at authenticated:3] Set("SIP/104-DEVEL-0000001e", "__MIXMONITOR_FILENAME=srv01-1452720545.73.wav") in new stack
[2016-01-13 22:29:05] VERBOSE[3263][C-000005b7] pbx.c: Executing [55512345 at authenticated:4] MixMonitor("SIP/104-DEVEL-0000001e", "srv01-1452720545.73.wav,ab") in new stack
[2016-01-13 22:29:05] VERBOSE[3263][C-000005b7] pbx.c: Executing [55512345 at authenticated:5] Dial("SIP/104-DEVEL-0000001e", "SIP/onlytest/55512345") in new stack
[2016-01-13 22:29:05] VERBOSE[3264][C-000005b7] app_mixmonitor.c: Begin MixMonitor Recording SIP/104-DEVEL-0000001e
[2016-01-13 22:29:05] VERBOSE[3263][C-000005b7] netsock2.c: Using SIP RTP TOS bits 184
[2016-01-13 22:29:05] VERBOSE[3263][C-000005b7] netsock2.c: Using SIP RTP CoS mark 5
[2016-01-13 22:29:05] VERBOSE[3263][C-000005b7] app_dial.c: Called SIP/onlytest/55512345
[2016-01-13 22:29:05] VERBOSE[3263][C-000005b7] app_dial.c: SIP/onlytest-0000001f answered SIP/104-DEVEL-0000001e
[2016-01-13 22:29:05] VERBOSE[3270][C-000005b7] bridge_channel.c: Channel SIP/onlytest-0000001f joined 'simple_bridge' basic-bridge <d27615ad-8fbf-4514-8273-1aff120054df>
[2016-01-13 22:29:05] VERBOSE[24765] chan_sip.c: Extension Changed 104-DEVEL[authenticated] new state InUse for Notify User 100-DEVEL
[2016-01-13 22:29:05] VERBOSE[3263][C-000005b7] bridge_channel.c: Channel SIP/104-DEVEL-0000001e joined 'simple_bridge' basic-bridge <d27615ad-8fbf-4514-8273-1aff120054df>
[2016-01-13 22:29:21] VERBOSE[3270][C-000005b7] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/onlytest-0000001f'
[2016-01-13 22:29:25] VERBOSE[24969][C-000005b9] netsock2.c: Using SIP RTP TOS bits 184
[2016-01-13 22:29:25] VERBOSE[24969][C-000005b9] netsock2.c: Using SIP RTP CoS mark 5
[2016-01-13 22:29:25] VERBOSE[3807][C-000005b9] pbx.c: Executing [105 at authenticated:1] Set("SIP/104-DEVEL-00000020", "__TRANSFER_CONTEXT=authenticated") in new stack
[2016-01-13 22:29:25] VERBOSE[3807][C-000005b9] pbx.c: Executing [105 at authenticated:2] Set("SIP/104-DEVEL-00000020", "RECORDINGFORMAT=wav") in new stack
[2016-01-13 22:29:25] VERBOSE[3807][C-000005b9] pbx.c: Executing [105 at authenticated:3] Set("SIP/104-DEVEL-00000020", "__MIXMONITOR_FILENAME=srv01-1452720565.75.wav") in new stack
[2016-01-13 22:29:25] VERBOSE[3807][C-000005b9] pbx.c: Executing [105 at authenticated:4] MixMonitor("SIP/104-DEVEL-00000020", "srv01-1452720565.75.wav,ab") in new stack
[2016-01-13 22:29:25] VERBOSE[3807][C-000005b9] pbx.c: Executing [105 at authenticated:5] Dial("SIP/104-DEVEL-00000020", "SIP/105-DEVEL") in new stack
[2016-01-13 22:29:25] VERBOSE[3808][C-000005b9] app_mixmonitor.c: Begin MixMonitor Recording SIP/104-DEVEL-00000020
[2016-01-13 22:29:25] VERBOSE[3807][C-000005b9] netsock2.c: Using SIP RTP TOS bits 184
[2016-01-13 22:29:25] VERBOSE[3807][C-000005b9] netsock2.c: Using SIP RTP CoS mark 5
[2016-01-13 22:29:25] VERBOSE[3807][C-000005b9] app_dial.c: Called SIP/105-DEVEL
[2016-01-13 22:29:25] VERBOSE[3807][C-000005b9] app_dial.c: SIP/105-DEVEL-00000021 is ringing
[2016-01-13 22:29:25] WARNING[3807][C-000005b9] translate.c: no samples for ulawtolin
[2016-01-13 22:29:28] VERBOSE[3807][C-000005b9] app_dial.c: SIP/105-DEVEL-00000021 answered SIP/104-DEVEL-00000020
[2016-01-13 22:29:28] VERBOSE[4077][C-000005b9] bridge_channel.c: Channel SIP/105-DEVEL-00000021 joined 'simple_bridge' basic-bridge <ce4af82b-34a9-45b4-9cf3-68a081029ac6>
[2016-01-13 22:29:28] VERBOSE[3807][C-000005b9] bridge_channel.c: Channel SIP/104-DEVEL-00000020 joined 'simple_bridge' basic-bridge <ce4af82b-34a9-45b4-9cf3-68a081029ac6>
[2016-01-13 22:29:42] VERBOSE[24969][C-000005b7] bridge_channel.c: Channel SIP/onlytest-0000001f left 'simple_bridge' basic-bridge <d27615ad-8fbf-4514-8273-1aff120054df>
[2016-01-13 22:29:42] VERBOSE[24969][C-000005b7] bridge_channel.c: Channel SIP/104-DEVEL-00000020 left 'simple_bridge' basic-bridge <ce4af82b-34a9-45b4-9cf3-68a081029ac6>
[2016-01-13 22:29:42] VERBOSE[24969][C-000005b7] bridge_channel.c: Channel SIP/onlytest-0000001f swapped with SIP/104-DEVEL-00000020 into 'simple_bridge' basic-bridge <ce4af82b-34a9-45b4-9cf3-68a081029ac6>
[2016-01-13 22:29:42] VERBOSE[3263][C-000005b7] bridge_channel.c: Channel SIP/104-DEVEL-0000001e left 'simple_bridge' basic-bridge <d27615ad-8fbf-4514-8273-1aff120054df>
[2016-01-13 22:29:42] VERBOSE[3263][C-000005b7] pbx.c: Spawn extension (authenticated, 55512345, 5) exited non-zero on 'SIP/104-DEVEL-0000001e'
[2016-01-13 22:29:42] VERBOSE[3264][C-000005b7] app_mixmonitor.c: MixMonitor close filestream (mixed)
[2016-01-13 22:29:42] VERBOSE[3264][C-000005b7] app_mixmonitor.c: End MixMonitor Recording SIP/104-DEVEL-0000001e
[2016-01-13 22:29:42] VERBOSE[3807][C-000005b9] pbx.c: Spawn extension (authenticated, 105, 5) exited non-zero on 'SIP/104-DEVEL-00000020'
[2016-01-13 22:29:42] VERBOSE[3808][C-000005b9] app_mixmonitor.c: MixMonitor close filestream (mixed)
[2016-01-13 22:29:42] VERBOSE[3808][C-000005b9] app_mixmonitor.c: End MixMonitor Recording SIP/104-DEVEL-00000020
[2016-01-13 22:29:42] VERBOSE[3270][C-000005b7] res_musiconhold.c: Stopped music on hold on SIP/onlytest-0000001f
[2016-01-13 22:29:42] VERBOSE[24765] chan_sip.c: Extension Changed 104-DEVEL[authenticated] new state Idle for Notify User 100-DEVEL
[2016-01-13 22:30:00] VERBOSE[4077][C-000005b9] bridge_channel.c: Channel SIP/105-DEVEL-00000021 left 'native_rtp' basic-bridge <ce4af82b-34a9-45b4-9cf3-68a081029ac6>
[2016-01-13 22:30:00] VERBOSE[3270][C-000005b7] bridge_channel.c: Channel SIP/onlytest-0000001f left 'native_rtp' basic-bridge <ce4af82b-34a9-45b4-9cf3-68a081029ac6>
{noformat}
This is the dialplan:
{noformat}
context authenticated {
104 => {
Set(__TRANSFER_CONTEXT=authenticated);
Set(RECORDINGFORMAT=wav);
Set(__MIXMONITOR_FILENAME=${UNIQUEID}.${RECORDINGFORMAT});
MixMonitor(${MIXMONITOR_FILENAME},ab);
Dial(SIP/104-DEVEL);
}
105 => {
Set(__TRANSFER_CONTEXT=authenticated);
Set(RECORDINGFORMAT=wav);
Set(__MIXMONITOR_FILENAME=${UNIQUEID}.${RECORDINGFORMAT});
MixMonitor(${MIXMONITOR_FILENAME},ab);
Dial(SIP/105-DEVEL);
}
55512345 => {
Set(__TRANSFER_CONTEXT=authenticated);
Set(RECORDINGFORMAT=wav);
Set(__MIXMONITOR_FILENAME=${UNIQUEID}.${RECORDINGFORMAT});
MixMonitor(${MIXMONITOR_FILENAME},ab);
Dial(SIP/onlytest/55512345);
}
}
{noformat}
In /var/spool/asterisk/monitor I can find the following files:
{noformat}
# ls -lat /var/spool/asterisk/monitor/ | more
totale 1,3M
-rw-r--r-- 1 root root 441K 13 gen 22:29 srv01-1452720545.73.wav
-rw-r--r-- 1 root root 223K 13 gen 22:29 srv01-1452720565.75.wav
drwxr-xr-x 3 root root 16K 13 gen 22:29 .
{noformat}
> Mixmonitor stop recording after atxfer
> --------------------------------------
>
> Key: ASTERISK-25674
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-25674
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Applications/app_mixmonitor
> Affects Versions: 13.6.0
> Environment: Asterisk 13.6 compiled from source on CentOS 6.7 64 bit. I have also tried asterisk 13.7.0-rc2 finding the same problem
> Reporter: Leandro Dardini
> Assignee: Leandro Dardini
> Severity: Minor
>
> This seems the same bug affecting asterisk few years ago and that got fixed in asterisk 11. I tested asterisk 11.20 and the bug is not present. In short, when recording is active on a call, making an attended transfer will continue to record (in a different file, but that is correct) the "short talk", but does not record the transferred call. Let me picture better the case, if not enough clear.
> Extension 104 dials number 555-12345, recording is active and the call is recorded correctly.
> Extension 104 puts the call on hold and dials extension 105, recording is active and the call is recorded correctly.
> Extension 104 transfer the 555-12345 call leg to 105. Recording is NOT active and the call is NOT recorded.
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