如果我拨打某人并暂停,星号会在几分钟后挂断。我在想某个地方有一个我找不到的地方。想法? 我想改变这个设置
freepbx => tools => asterisk sip setting =>媒体与媒体RTP设置
日志摘录:
[Mar 9 09:49:16] VERBOSE[19807] pbx.c: -- Executing [788787636@Local-route:1] Macro("SIP/100-000804aa", "user-callerid,SKIPTTL,") in new stack
[Mar 9 09:49:16] VERBOSE[19807] pbx.c: -- Executing [788787636@Local-route:2] NoOp("SIP/100-000804aa", "Calling Out Route: to-outside") in new stack
[Mar 9 09:49:16] VERBOSE[19807] pbx.c: -- Executing [788787636@Local-route:3] Set("SIP/100-000804aa", "MOHCLASS=ros-moh") in new stack
[Mar 9 09:49:16] VERBOSE[19807] pbx.c: -- Executing [788787636@Local-route:4] Set("SIP/100-000804aa", "_NODEST=") in new stack
[Mar 9 09:49:16] VERBOSE[19807] pbx.c: -- Executing [788787636@Local-route:5] Macro("SIP/100-000804aa", "record-enable,100,OUT,") in new stack
[Mar 9 09:49:16] VERBOSE[19807] pbx.c: -- Executing [788787636@Local-route:6] Macro("SIP/100-000804aa", "dialout-trunk,1,88787636,") in new stack
[Mar 9 09:50:11] VERBOSE[19807] res_agi.c: <SIP/100-000804aa>AGI Tx >> agi_dnid: 788787636
[Mar 9 09:50:11] VERBOSE[19807] res_agi.c: <SIP/100-000804aa>AGI Tx >> 200 result=1 (788787636)
[Mar 9 09:50:11] VERBOSE[19807] pbx.c: == Spawn extension (Local-route, 788787636, 6) exited non-zero on 'SIP/100-000804aa'
答案 0 :(得分:0)
很可能你会使用sip。它在sip.conf中有参数
rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
; on the audio channel
; when we're on hold (must be > rtptimeout)
答案 1 :(得分:0)
您正在寻找的参数是@arheops所说的rtpholdtimeout
。
默认情况下,它在/etc/asterisk/sip.conf
上配置。但是您不应该在该文件上设置值,而应该通过Elastix Web GUI(实际上是FreePBX Web GUI)进行设置。 PBX -> Unembedded FreePBX -> Tools -> Asterisk SIP settings -> Media & RTP settings
或/etc/asterisk/sip_general_custom.conf
,因为sip.conf
由FreePBX自动生成,不应手动修改。