Question

我刚开始使用Asterisk并按照O'Reilly Guide设置测试Asterisk服务器。我已经用CentOS 6.4建立了一个VM，它有1GB RAM和50GB硬盘。

安装完成后，我在2台可以互相通话的电脑上成功安装了软电话。我现在必须录制这些电话 - 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的通话工作正常。但是/var/spool/asterisk/monitor下没有录音。而且，在调用期间，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呼叫启用呼叫记录？

Answer 1

当您使用mixmonitor时，您已经检查过您的SIP设备

directmedia=no

您也可以通过启用调试来获得更多信息

Answer 2

如果您按原样关注O'Reilly的书，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用户的写入权限。您需要为实际写入文件夹的用户/组授予写权限（即将.wav文件写入该位置）。

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

这应该为你解决。希望这有帮助。

使用MixMonitor在Asterisk中拨打录音

2 个答案: