我有兩個星號服務器,其中一個帶有PBX inflash,另外一個只有Asterisk安裝在CentOS上。我需要從PBXINFLASH遷移到Asterisk 11.9.0。 Flash中的PbX正在運行Asterisk 10.12.1。 我有一個撥號方案,它在Asterisk 10.12.1上工作得非常好,但在帶有Asterisk 11.9.0的新方框中,DTMF或用戶鍵輸入不是撥號方案的一部分。我試圖做dtmf調試兩個服務器是相同的調試結果沒有區別,也奇怪這是我的一個機器dialprn工作正常,其他工作部分。撥號方案是呼叫方呼叫屏幕,呼叫方按1繼續,接收方獲得呼叫,系統要求按1接受呼叫或掛斷現在一個Asterisk 11.9.0呼叫方按1輸入工作正常,但第二個用戶/接收方按1不做任何事情。帶有read()函數的星號11.9問題和dtmf
我正在使用sip帳戶來測試我的dtmf。我交換了我的SIP帳戶並啜飲軟電話以測試相同的問題。以下是兩個部分相同的宏半工作和下半年沒有采取用戶輸入
上半年工作,並採取用戶輸入。
exten => _X.,n,GotoIf($[${GROUP_COUNT(${CallerNum})} > 1]?Exceeded) ;Exceeded?
exten => _X.,n,Set(HngupCount=1);Hangup
exten => _X.,n,Flite(Please press 1 to speak with ${destUID})
exten => _X.,n,Read(yesno,sip-silence,1,,2,5)
exten => _X.,n,GotoIf($[${yesno} = 1]?continue:hangup)
它不能正常工作或接受用戶的輸入下半場: -
[macro-Dial2]
exten => s,1,Wait(1);ResetCDR
exten => s,n,Set(_StartTime=${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten => s,n,ResetCDR
exten => s,n,Set(_RCount=1)
exten => s,n(Repeat),Flite(Hi there)
exten => s,n,Flite(${ARG1} wants to speak to you. Please press 1 to accept the call. 2 to forward the call to voicemail or 3 to reject the call.)
exten => s,n,Flite(we are connecting you)
exten => s,n,Read(ACCEPT,sip-silence,1,,1,5)
exten => s,n,Set(_RCount=$[${RCount} + 1])
exten => s,n,NoOp(Counter is ${RCount} -- the user selected: ${ACCEPT});
exten => s,n,Gotoif($[${ACCEPT} = 1]?accept:vm) ;Accept the call
exten => s,n(vm),Gotoif($[${ACCEPT} = 2]?voicemail:rej) ;forward the call to dummy voicemail (Actually just record the callers message)
exten => s,n(rej),Gotoif($[${ACCEPT} = 3]?reject) ;Reject the call and hangup
exten => s,n,Gotoif($[${RCount} > 2]?reject:Repeat) ; If no key pressed, just hangup the call and inform the User.
exten => s,n(accept),set(SecLeg=${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
在下半年能正常工作,直到下面幾行:
exten => s,n,Flite(${ARG1} wants to speak to you. Please press 1 to accept the call. 2 to forward the call to voicemail or 3 to reject the call.)
exten => s,n,Flite(we are connecting you)
您可以發佈來自AST CLI失敗調用的文本嗎?我需要看到「AST認爲」能夠在這裏幫助你。 – MichelV69
[0K <--- SIP從UDP讀:122.173.207.156:5060 ---> SIP/2.0 200 OK 途經:SIP/2.0/UDP 173.230.137.73:5060;branch=z9hG4bK3d82f078;rport=5060 聯繫人: To:; tag = 6b7c1722 From: ;標記= as292ce1cc 的Call-ID:[email protected]:5060 的Cseq:103 BYE 的User-Agent:Zoiper rev.11137 的Content-Length:0 –
user1492502
請設置「詳細「設置爲9,以便撥號計劃處理可見,並在」R「之前和之後發佈來自五行的相關輸出ead()「聲明。除非您僅通過SIP-INFO傳輸DTMF,否則SIP數據包調試在這裏毫無價值。 – MichelV69