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Asterisk chan_dongle Huawei E303 dźwięk tylko w jedna stronę

przemyslawuk 04 Maj 2014 18:44 1083 0
  • #1
    przemyslawuk
    Poziom 12  
    Witam, Mam na Linuxie Asteriska 1.8 z modemem E303. Problem jest taki, że jak dzwonię telefonGSM->dongle->asterisk->Twinkle(SIP) to słychać głos w Twinkle, ale w słuchawce telefonu GSM cisza. Odwrotnie Twinkle -> asterisk-> dongle-> telefonGSM dźwięk jest, ale okropnie zniekształcony.
    Natomiast telefonGSM->dongle->asterisk_echo_test słyszę swoje echo perfekcyjnie czysto.

    I jeszcze jedno. Jak dzwonie z telefonu GSM to rozmowa się nagrywa, jak dzwonię z Twinkle na telefon to pliku *wav nie ma w katalogu. O co chodzi?

    Moje asteriskCLI, sip i extension.conf :


    Code:
           -- Executing [+1234567890@dongle-incomming:1] NoOp("Dongle/dongle0-0100000001", "Call from dongle . Start call recording.") in new stack
    
            -- Executing [+1234567890@dongle-incomming:2] MixMonitor("Dongle/dongle0-0100000001", "1399136727.2.wav,ab") in new stack
            -- Executing [+1234567890@dongle-incomming:3] Dial("Dongle/dongle0-0100000001", "SIP/2101") in new stack
          == Begin MixMonitor Recording Dongle/dongle0-0100000001
          == Using SIP RTP TOS bits 184
          == Using SIP RTP CoS mark 5
            -- Called SIP/2101
            -- SIP/2101-00000001 is ringing
          == Spawn extension (dongle-incomming, +1234567890, 3) exited non-zero on 'Dongle/dongle0-0100000001'
          == End MixMonitor Recording Dongle/dongle0-0100000001
          == Using SIP RTP TOS bits 184
          == Using SIP RTP CoS mark 5
            -- Executing [796020110@sip-local:1] NoOp("SIP/2101-00000002", "Dongle dialing 796020110. Start call recording.") in new stack
            -- Executing [796020110@sip-local:2] MixMonitor("SIP/2101-00000002", "1399136778.4.wav,ab") in new stack
            -- Executing [796020110@sip-local:3] Dial("SIP/2101-00000002", "dongle/dongle0/796020110") in new stack
            -- Called dongle/dongle0/796020110
          == Begin MixMonitor Recording SIP/2101-00000002
            -- Dongle/dongle0-0100000002 is making progress passing it to SIP/2101-00000002
          == Spawn extension (sip-local, 796020110, 3) exited non-zero on 'SIP/2101-00000002'
          == End MixMonitor Recording SIP/2101-00000002
          == Using SIP RTP TOS bits 184
          == Using SIP RTP CoS mark 5
            -- Executing [796020110@sip-local:1] NoOp("SIP/2101-00000003", "Dongle dialing 796020110. Start call recording.") in new stack
            -- Executing [796020110@sip-local:2] MixMonitor("SIP/2101-00000003", "1399136831.6.wav,ab") in new stack
            -- Executing [796020110@sip-local:3] Dial("SIP/2101-00000003", "dongle/dongle0/796020110") in new stack
          == Begin MixMonitor Recording SIP/2101-00000003
            -- Called dongle/dongle0/796020110
            -- Dongle/dongle0-0100000003 is making progress passing it to SIP/2101-00000003
        [May  3 19:07:29] ERROR[13395]: chan_dongle.c:433 do_monitor_phone: [dongle0] timedout while waiting 'OK' in response to 'AT'
            -- [dongle0] Dongle has disconnected
          == Everyone is busy/congested at this time (1:0/0/1)
            -- Executing [796020110@sip-local:4] StopMixMonitor("SIP/2101-00000003", "") in new stack
            -- Executing [796020110@sip-local:5] NoOp("SIP/2101-00000003", "Terminating call to 796020110. Stop call recording.") in new stack
            -- Executing [796020110@sip-local:6] Hangup("SIP/2101-00000003", "") in new stack
          == Spawn extension (sip-local, 796020110, 6) exited non-zero on 'SIP/2101-00000003'
          == End MixMonitor Recording SIP/2101-00000003
            -- [dongle0] Trying to connect on /dev/ttyUSB2...
            -- [dongle0] Dongle has connected, initializing...
            -- [dongle0] Dongle initialized and ready

           [dongle-incomming]
        exten => _+X.,1,NoOp(Call from dongle ${CALLERID}. Start call recording.)
        same  => n,MixMonitor(${UNIQUEID}.wav,ab)
        same  => n,answer()
        same  => n,Dial(SIP/2101)
        same  => n,StopMixMonitor()
        same  => n,NoOp(Call from dongle ${CALLERID} ended. Stop call recording.)
        same  => n,hangup()

        [dongle-outcomming]
        exten => _[+0-9]XXXXX.,1,NoOp(Dongle dialing ${EXTEN}. Start call recording.)
        same  => n,MixMonitor(${UNIQUEID}.wav,ab)
        same  => n,answer()
        same  => n,Dial(dongle/dongle0/${EXTEN})
        same  => n,StopMixMonitor()
        same  => n,NoOp(Terminating call to ${EXTEN}. Stop call recording.)
        same  => n,hangup()

        ;sip.conf
        [soft-phone](!)                 ;(!) gives us a template
        type=friend
        context=sip-local               ;The context name is what links to the extensions
                                        ;and can be any name.
        host=dynamic                  ;device will register with asterisk
        nat=yes                         ;allow network address translation
        secret=1111                     ;Password needed to authenticate
        dtmfmode=auto                   ;Accept touch tones from device
        canreinvite=no
        disallow=all
        allow=ulaw,alaw,speex,gsm,h261,h263,h263p
        directmedia=no

        [lenovo](soft-phone)            ;lenovo is the name of the user which connects
        bindport=5066                   ;from the soft phone, it must match your SIP username
        callerid=2101