使用 MixMonitor 在 Asterisk 中通话录音

Question

我刚开始使用 Asterisk 并按照 O'Reilly Guide 设置测试 Asterisk 服务器。我用 CentOS 6.4 设置了一个虚拟机，它有 1GB 内存和 50GB 硬盘。

安装后，我在两台电脑上成功设置了软电话，可以互相通话。我现在必须记录这些电话 - 在搜索时，我找到了 this site 并相应地编辑了我的 extensions.conf。

这是我的extensions.conf

[globals]

EXT_TESTTWO=SIP/0000FFFF0002
EXT_TESTONE=SIP/0000FFFF0001

[default]
exten => 0000FFFF0001,hint,SIP/0000FFFF0001
exten => 0000FFFF0002,hint,SIP/0000FFFF0002

[Queues]
exten => 7001,1,Verbose(2,${CALLERID(all)} entering the support queue)
        same => n,Queue(support)
        same => n,Hangup()
exten => 7002,1,Verbose(2,${CALLERID(all)} entering the sales queue)
        same => n,Queue(sales)
        same => n,Hangup()

[macro-automon]
exten => s,1,Set(MONITOR_FILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${CALLERID(num)})
        same => n,Playback(beep) ; optional - hear when recording starts
        same => n,MixMonitor(${MONITOR_FILENAME}.wav,b)

[LocalSets]
include => Queues                              ; allow phones to call queues    
exten => 101,1,Dial(${EXT_TESTONE},20,m)       ; Replace 0000FFFF0001 with your device name
        same => n,Playback(vm-nobodyavail)     ; Play "no one's available"
        same => n,Hangup()

exten => 102,1,Noop(Dialing 102);
exten => 102,n,Macro(automon) ; start monitor
exten => 102,n,Dial(SIP/0000FFFF0002,30) ; 30 secs
exten => 102,n,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => 102-NOANSWER,1,Voicemail(20,us) ; If unavailable, send to voicemail w/ unavail announce
exten => 102-NOANSWER,n,Playback(vm-goodbye)
exten => 102-NOANSWER,n,Hangup
exten => 102-BUSY,1,Voicemail(${MACRO_EXTEN},bs) ; If busy, send to voicemail w/ busy announce
exten => 102-BUSY,n,Playback(vm-goodbye)
exten => 102-BUSY,n,Hangup
exten => _102-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer

;exten => 102,1,Dial(${EXT_TESTTWO},20,m)  ;Replace 0000FFFF0002 with your device name
        ;same => n,Playback(vm-nobodyavail)     ; Play "no one's available"
        ;same => n,Hangup()

从 101 到 102 和 102 到 101 的呼叫工作正常。但是/var/spool/asterisk/monitor 下没有录音。此外，在通话期间，我得到以下调试输出：

Read factory 0x7f971001f428 was pretty quick last time, waiting for them.
Read factory 0x7f971001f428 and write factory 0x7f9710020068 both fail to provide 160 samples
No remote address on RTP instance '0x7f9710009258' so dropping frame
Read factory 0x7f971001f428 was pretty quick last time, waiting for them.
Read factory 0x7f971001f428 and write factory 0x7f9710020068 both fail to provide 160 samples
No remote address on RTP instance '0x7f9710009258' so dropping frame
Read factory 0x7f971001f428 was pretty quick last time, waiting for them.
Read factory 0x7f971001f428 and write factory 0x7f9710020068 both fail to provide 160 samples
No remote address on RTP instance '0x7f9710009258' so dropping frame
Read factory 0x7f971001f428 was pretty quick last time, waiting for them.
Read factory 0x7f971001f428 and write factory 0x7f9710020068 both fail to provide 160 samples
No remote address on RTP instance '0x7f9710009258' so dropping frame
Read factory 0x7f971001f428 was pretty quick last time, waiting for them.
Read factory 0x7f971001f428 and write factory 0x7f9710020068 both fail to provide 160 samples
No remote address on RTP instance '0x7f9710009258' so dropping frame
Read factory 0x7f971001f428 was pretty quick last time, waiting for them.

我在这里做错了什么？如何为特定分机上的来电和去电启用通话录音？

Answer 1

当您使用 mixmonitor 时，您已检查您的 sip 设备是否有

directmedia=no

您还可以通过启用调试获得更多信息

Answer 2

如果您按原样遵循 O'Reilly 手册，它会要求您创建一个名为 asteriskpbx 的用户并以该用户身份运行其余配置。因此，在开始的安装过程中，/var/spool/asterisk/monitor 文件夹可能只有 root 的 write 权限。您需要为实际写入文件夹的用户/组授予 write 权限（即将.wav 文件写入该位置）。

chmod -R 775 /var/spool/asterisk/monitor

这应该会为您解决问题。希望这会有所帮助。