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Linux Intel
Asked: 2012-07-17 17:05:04 +0800 CST

用于来电和去电的 Skype Connect 和 Elastix

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我订购了 Skype Connect，我想将 Skype Connect 与我的 Elastix 服务器集成以处理来电和去电。

我使用以下信息通过 GUI 创建了新的 SIP 中继：

    Incoming Settings
[skype_in]
disallow=all
type=friend
username=sipusername
fromdomain=sip.skype.com
fromuser=sipusername
realm=sip.skype.com
host=sip.skype.com
dtmfmode=rfc2833
secret=sipuserpass
nat=yes
insecure=invite
qualify=yes
allow=alaw
allow=ulaw
amaflags=default
trustrpid=no
sendrpid=yes
context=from-trunk-sip-Skype_out

Outgoing Settings : 
[Skype_out]
context=from-trunk-sip-Skype_out

Register String:
SIPUSER:[email protected]

来电正常。

我试着拨打 00448717893642 和 448717893642 以获得伦敦语音时钟和许多其他号码，但拨出电话不起作用，它一直说（无法完成拨号）

拨号后的 Elastix 日志

[Jul 17 01:01:25] VERBOSE[3501] pbx.c:     -- Executing [448717893642@from-internal:1] ResetCDR("SIP/100-00000010", "") in new stack
[Jul 17 01:01:25] VERBOSE[3501] pbx.c:     -- Executing [448717893642@from-internal:2] NoCDR("SIP/100-00000010", "") in new stack
[Jul 17 01:01:25] VERBOSE[3501] pbx.c:     -- Executing [448717893642@from-internal:3] Progress("SIP/100-00000010", "") in new stack
[Jul 17 01:01:25] VERBOSE[3501] pbx.c:     -- Executing [448717893642@from-internal:4] Wait("SIP/100-00000010", "1") in new stack
[Jul 17 01:01:26] VERBOSE[3501] pbx.c:     -- Executing [448717893642@from-internal:5] Progress("SIP/100-00000010", "") in new stack
[Jul 17 01:01:26] VERBOSE[3501] pbx.c:     -- Executing [448717893642@from-internal:6] Playback("SIP/100-00000010", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[Jul 17 01:01:26] VERBOSE[3501] file.c:     -- <SIP/100-00000010> Playing 'silence/1.gsm' (language 'en')
[Jul 17 01:01:27] VERBOSE[3501] file.c:     -- <SIP/100-00000010> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
[Jul 17 01:01:29] VERBOSE[3501] file.c:     -- <SIP/100-00000010> Playing 'check-number-dial-again.gsm' (language 'en')
[Jul 17 01:01:32] VERBOSE[3501] pbx.c:     -- Executing [448717893642@from-internal:7] Wait("SIP/100-00000010", "1") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Executing [448717893642@from-internal:8] Congestion("SIP/100-00000010", "20") in new stack
[Jul 17 01:01:33] WARNING[3501] channel.c: Prodding channel 'SIP/100-00000010' failed
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:   == Spawn extension (from-internal, 448717893642, 8) exited non-zero on 'SIP/100-00000010'
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Executing [h@from-internal:1] Macro("SIP/100-00000010", "hangupcall") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Executing [s@macro-hangupcall:1] GotoIf("SIP/100-00000010", "1?endmixmoncheck") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Goto (macro-hangupcall,s,9)
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Executing [s@macro-hangupcall:9] NoOp("SIP/100-00000010", "End of MIXMON check") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Executing [s@macro-hangupcall:10] GotoIf("SIP/100-00000010", "1?nomeetmemon") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Goto (macro-hangupcall,s,28)
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Executing [s@macro-hangupcall:28] NoOp("SIP/100-00000010", "End of MEETME check") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Executing [s@macro-hangupcall:29] GotoIf("SIP/100-00000010", "1?noautomon") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Goto (macro-hangupcall,s,34)
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Executing [s@macro-hangupcall:34] NoOp("SIP/100-00000010", "TOUCH_MONITOR_OUTPUT=") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Executing [s@macro-hangupcall:35] GotoIf("SIP/100-00000010", "1?noautomon2") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Goto (macro-hangupcall,s,41)
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Executing [s@macro-hangupcall:41] NoOp("SIP/100-00000010", "MONITOR_FILENAME=") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Executing [s@macro-hangupcall:42] GotoIf("SIP/100-00000010", "1?skiprg") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Goto (macro-hangupcall,s,45)
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Executing [s@macro-hangupcall:45] GotoIf("SIP/100-00000010", "1?skipblkvm") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Goto (macro-hangupcall,s,48)
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Executing [s@macro-hangupcall:48] GotoIf("SIP/100-00000010", "1?theend") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Goto (macro-hangupcall,s,50)
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:     -- Executing [s@macro-hangupcall:50] Hangup("SIP/100-00000010", "") in new stack
[Jul 17 01:01:33] VERBOSE[3501] app_macro.c:   == Spawn extension (macro-hangupcall, s, 50) exited non-zero on 'SIP/100-00000010' in macro 'hangupcall'
[Jul 17 01:01:33] VERBOSE[3501] pbx.c:   == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-00000010'

我应该在传入和传出设置中做任何修改才能正常工作吗？

linux asterisk voip skype

2 个回答

  1. 你没有指定你是如何拨打电话的，所以我会推荐你​​参考Skype Connect 文档，其中说：

    E.164（国家代码和国内号码），所有呼叫的国际号码格式

    您很可能没有发送国家代码。这是所有 Skype Connect 拨出电话所必需的。

    所以如果你想拨打伦敦的时钟，你会发送：

    448717893642

    要拨打美国的号码，例如Pennsylvania 6-5000，您需要发送：

    12127365000

    可能您还需要+在数字的开头包含符号，例如：

    +44871893642

    快速的 Google 搜索表明您还需要在您的 Skype 帐户配置文件中明确允许拨出电话。由于我什至无法开始猜测的原因，默认情况下这似乎是禁用的。

    • 0
  2. Best Answer

    我通过插件解决了我的问题

    exten => _00X.,1, Answer
exten => _00X.,n, Set(CALLERID(num)=ID) ;
exten => _00X.,n, Dial(SIP/${EXTEN}@Skype_out)
exten => _00X.,n, Hangup

    进入 extensions.conf

    • 0

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