【问题标题】:asterisk hangup call when hold保持时挂断电话
【发布时间】:2014-08-27 20:12:22
【问题描述】:

如果我拨打某人并将其置于保留状态,则 asterisk 将在几分钟后挂断。我在想有一个我找不到的地方。想法? 我想改变这个设置

freepbx => 工具 => 星号 sip 设置 => 媒体和 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'

【问题讨论】:

    标签: asterisk elastix


    【解决方案1】:

    您很可能使用 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)
    

    【讨论】:

    • 相同的设置可以在提供者端。您可以执行“sip set debug on”并检查哪一侧停止呼叫。
    【解决方案2】:

    正如@arheops 所说,您要查找的参数是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 -&gt; Unembedded FreePBX -&gt; Tools -&gt; Asterisk SIP settings -&gt; Media &amp; RTP settings,或/etc/asterisk/sip_general_custom.conf,因为sip.conf是FreePBX自动生成的,不应手动修改。

    【讨论】:

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