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  • Почему не работает группа номеров для входящих вызовов (issabel 4, asterisk 11)?

    Nazariy
    @Nazariy Автор вопроса
    0.
    5c4847c817875443433094.jpeg
    1. сделал
    2. сделал
    3. сделал
    4.
    spoiler
    issabel*CLI> dialplan show ext-did-0002
    [ Context 'ext-did-0002' created by 'pbx_config' ]
      '134628' =>       1. Set(__FROM_DID=${EXTEN})                   [pbx_config]
                        2. Gosub(app-blacklist-check,s,1())           [pbx_config]
                        3. Set(CDR(did)=${FROM_DID})                  [pbx_config]
                        4. ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)})) [pbx_config]
                        5. Set(CHANNEL(musicclass)=default)           [pbx_config]
                        6. Set(__MOHCLASS=default)                    [pbx_config]
                        7. Set(__CALLINGPRES_SV=${CALLERPRES()})      [pbx_config]
                        8. Set(CALLERPRES()=allowed_not_screened)     [pbx_config]
         [dest-ext]     9. Goto(from-did-direct,102,1)                [pbx_config]
      'fax' =>          1. Goto(${CUT(FAX_DEST,^,1)},${CUT(FAX_DEST,^,2)},${CUT(FAX_DEST,^,3)}) [pbx_config]
      Include =>        'ext-did-0002-custom'                         [pbx_config]
    
    -= 2 extensions (10 priorities) in 1 context. =-

    5. сделал
    6.
    spoiler
    <--- SIP read from UDP:185.45.152.174:5060 --->
    <--- SIP read from UDP:185.45.152.174:5060 --->
    INVITE sip:134628@192.168.3.1:5060 SIP/2.0
    Record-Route: <sip:185.45.152.174;lr=on;ftag=as228d9596;nat=yes>
    Via: SIP/2.0/UDP 185.45.152.174;branch=z9hG4bKaed4.aacd7fc1cbaa307edc993b466321f392.0
    Via: SIP/2.0/UDP 185.45.152.131:5060;rport=5060;branch=z9hG4bK7fea108e
    Max-Forwards: 69
    From: "+380939333315" <sip:+380939333315@sip.задарма.com>;tag=as228d9596
    To: <sip:134628@185.45.152.174>
    Contact: <sip:+380939333315@185.45.152.131:5060>
    Call-ID: 27e390fc011aade23271ed911e86621b@185.45.152.131:5060
    CSeq: 102 INVITE
    User-Agent: задарма Voip
    Date: Wed, 23 Jan 2019 10:56:36 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces
    CALLED_DID: 380947117050
    Content-Type: application/sdp
    Content-Length: 411
    hostname: sipbalancer-2
    cc_num: 134628
    cc_counter: 1
    
    v=0
    o=root 80984630 80984630 IN IP4 185.45.152.176
    s=задарма Voip
    c=IN IP4 185.45.152.176
    t=0 0
    m=audio 18946 RTP/AVP 8 0 18 110 117 119 3 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:110 speex/8000
    a=rtpmap:117 speex/16000
    a=rtpmap:119 speex/32000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv

    в логе задарма переписал на русский чтоб система пропустила
  • Почему не работает группа номеров для входящих вызовов (issabel 4, asterisk 11)?

    Nazariy
    @Nazariy Автор вопроса
    dongle.conf
    spoiler
    [general]
    
    interval=15			; Number of seconds between trying to connect to devices
    
    ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
    ;jbenable = yes			; Enables the use of a jitterbuffer on the receiving side of a
    				; Dongle channel. Defaults to "no". An enabled jitterbuffer will
    				; be used only if the sending side can create and the receiving
    				; side can not accept jitter. The Dongle channel can't accept jitter,
    				; thus an enabled jitterbuffer on the receive Dongle side will always
    				; be used if the sending side can create jitter.
    
    ;jbforce = no			; Forces the use of a jitterbuffer on the receive side of a Dongle
    				; channel. Defaults to "no".
    
    ;jbmaxsize = 200		; Max length of the jitterbuffer in milliseconds.
    
    ;jbresyncthreshold = 1000	; Jump in the frame timestamps over which the jitterbuffer is
    				; resynchronized. Useful to improve the quality of the voice, with
    				; big jumps in/broken timestamps, usually sent from exotic devices
    				; and programs. Defaults to 1000.
    
    ;jbimpl = fixed			; Jitterbuffer implementation, used on the receiving side of a Dongle
    				; channel. Two implementations are currently available - "fixed"
    				; (with size always equals to jbmaxsize) and "adaptive" (with
    				; variable size, actually the new jb of IAX2). Defaults to fixed.
    
    ;jbtargetextra = 40		; This option only affects the jb when 'jbimpl = adaptive' is set.
    				; The option represents the number of milliseconds by which the new jitter buffer
    				; will pad its size. the default is 40, so without modification, the new
    				; jitter buffer will set its size to the jitter value plus 40 milliseconds.
    				; increasing this value may help if your network normally has low jitter,
    				; but occasionally has spikes.
    
    ;jblog = no			; Enables jitterbuffer frame logging. Defaults to "no".
    ;-----------------------------------------------------------------------------------
    
    [defaults]
    ; now you can set here any not required device settings as template
    ;   sure you can overwrite in any [device] section this default values
    
    ;context=default			; context for incoming calls
    ;context=from-gsm
    context=from-pstn
    group=0				; calling group
    rxgain=0			; increase the incoming volume; may be negative
    txgain=0			; increase the outgoint volume; may be negative
    autodeletesms=yes		; auto delete incoming sms
    resetdongle=yes			; reset dongle during initialization with ATZ command
    u2diag=-1			; set ^U2DIAG parameter on device (0 = disable everything except modem function) ; -1 not use ^U2DIAG command
    usecallingpres=yes		; use the caller ID presentation or not
    callingpres=allowed_passed_screen ; set caller ID presentation		by default use default network settings
    disablesms=no			; disable of SMS reading from device when received
    				;  chan_dongle has currently a bug with SMS reception. When a SMS gets in during a
    				;  call chan_dongle might crash. Enable this option to disable sms reception.
    				;  default = no
    
    language=ru			; set channel default language
    smsaspdu=yes			; if 'yes' send SMS in PDU mode, feature implementation incomplete and we strongly recommend say 'yes'
    mindtmfgap=45			; minimal interval from end of previews DTMF from begining of next in ms
    mindtmfduration=80		; minimal DTMF tone duration in ms
    mindtmfinterval=200		; minimal interval between ends of DTMF of same digits in ms
    
    callwaiting=auto		; if 'yes' allow incoming calls waiting; by default use network settings
    				; if 'no' waiting calls just ignored
    disable=no			; OBSOLETED by initstate: if 'yes' no load this device and just ignore this section
    
    initstate=start			; specified initial state of device, must be one of 'stop' 'start' 'remote'
    				;   'remove' same as 'disable=yes'
    
    ;exten=+1234567890		; exten for start incoming calls, only in case of Subscriber Number not available!, also set to CALLERID(ndid)
    
    dtmf=relax			; control of incoming DTMF detection, possible values:
    				;   off	   - off DTMF tones detection, voice data passed to asterisk unaltered
    				;              use this value for gateways or if not use DTMF for AVR or inside dialplan
    				;   inband - do DTMF tones detection
    				;   relax  - like inband but with relaxdtmf option
    				;  default is 'relax' by compatibility reason
    
    [dongle0] 
    audio=/dev/ttyUSB1
    data=/dev/ttyUSB2
    
    [dongle1]
    audio=/dev/ttyUSB4
    data=/dev/ttyUSB5

    задарма
    5c48346050a98298724302.jpeg
    dialplan show ext-did-0002
    spoiler
    issabel*CLI> dialplan show ext-did-0002
    [ Context 'ext-did-0002' created by 'pbx_config' ]
      '380935591582' => 1. Set(__FROM_DID=${EXTEN})                   [pbx_config]
                        2. Gosub(app-blacklist-check,s,1())           [pbx_config]
                        3. Set(CDR(did)=${FROM_DID})                  [pbx_config]
                        4. ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)})) [pbx_config]
                        5. Set(CHANNEL(musicclass)=default)           [pbx_config]
                        6. Set(__MOHCLASS=default)                    [pbx_config]
                        7. Set(__CALLINGPRES_SV=${CALLERPRES()})      [pbx_config]
                        8. Set(CALLERPRES()=allowed_not_screened)     [pbx_config]
         [dest-ext]     9. Goto(from-did-direct,102,1)                [pbx_config]
      'fax' =>          1. Goto(${CUT(FAX_DEST,^,1)},${CUT(FAX_DEST,^,2)},${CUT(FAX_DEST,^,3)}) [pbx_config]
      Include =>        'ext-did-0002-custom'                         [pbx_config]
    
    -= 2 extensions (10 priorities) in 1 context. =-

    так-же проверил функцию очереди, работает.
  • Почему не работает группа номеров для входящих вызовов (issabel 4, asterisk 11)?

    Nazariy
    @Nazariy Автор вопроса
    У меня еще есть trunk с купленным номером задарма.com, но я его отключил, чтоб логи не засорял. С ним логи не смотрел правда, но смысл идентичен: на внутренние он дозванивался сразу, на группу, отбивал сразу занято.
  • Почему не работает группа номеров для входящих вызовов (issabel 4, asterisk 11)?

    Nazariy
    @Nazariy Автор вопроса
    Спасибо за статью!
    INVITE это примерно это:
    spoiler
    Audio is at 15586
    Adding codec 100003 (ulaw) to SDP
    Adding codec 100002 (gsm) to SDP
    Adding codec 100004 (alaw) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (no NAT) to 192.168.3.27:1036:
    INVITE sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715 SIP/2.0
    Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK6afc17f5
    Max-Forwards: 70
    From: "dongle0" <sip:+380939333315@192.168.3.151>;tag=as74726125
    To: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
    Contact: <sip:+380939333315@192.168.3.151:5060>
    Call-ID: 2ff49f5d67e95131789afa962b474be1@192.168.3.151:5060
    CSeq: 102 INVITE
    User-Agent: IPBX-2.11.0(11.25.3)
    Date: Tue, 22 Jan 2019 17:32:10 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 285
    
    v=0
    o=root 1265205272 1265205272 IN IP4 192.168.3.151
    s=Asterisk PBX 11.25.3
    c=IN IP4 192.168.3.151
    t=0 0
    m=audio 15586 RTP/AVP 0 3 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv
    
    ---
        -- Called SIP/102
    [2019-01-22 19:32:10] WARNING[7523][C-00000036]: channel.c:1002 channel_indicate: [Dongle/dongle0-0100000021] Don't know how to indicate condition 22
    [2019-01-22 19:32:10] WARNING[7523][C-00000036]: channel.c:1002 channel_indicate: [Dongle/dongle0-0100000021] Don't know how to indicate condition 22
    
    <--- SIP read from UDP:192.168.3.27:1036 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK6afc17f5
    To: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
    From: "dongle0" <sip:+380939333315@192.168.3.151>;tag=as74726125
    Call-ID: 2ff49f5d67e95131789afa962b474be1@192.168.3.151:5060
    CSeq: 102 INVITE
    Content-Length: 0
    
    <------------->
    --- (7 headers 0 lines) ---
    [2019-01-22 19:32:11] WARNING[7523][C-00000036]: channel.c:1002 channel_indicate: [Dongle/dongle0-0100000021] Don't know how to indicate condition 33
    
    <--- SIP read from UDP:192.168.3.27:1036 --->
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK6afc17f5
    Contact: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
    To: "102"<sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>;tag=31158b0a
    From: "dongle0" <sip:+380939333315@192.168.3.151>;tag=as74726125
    Call-ID: 2ff49f5d67e95131789afa962b474be1@192.168.3.151:5060
    CSeq: 102 INVITE
    User-Agent: X-Lite release 5.4.0 stamp 94388
    Allow-Events: talk, hold
    Content-Length: 0
    
    <------------->
    --- (10 headers 0 lines) ---
    list_route: hop: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
    [2019-01-22 19:32:11] WARNING[7523][C-00000036]: channel.c:1002 channel_indicate: [Dongle/dongle0-0100000021] Don't know how to indicate condition 33
        -- SIP/102-00000014 is ringing
    
    <--- SIP read from UDP:192.168.3.27:1036 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK6afc17f5
    Contact: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
    To: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>;tag=31158b0a
    From: "dongle0" <sip:+380939333315@192.168.3.151>;tag=as74726125
    Call-ID: 2ff49f5d67e95131789afa962b474be1@192.168.3.151:5060
    CSeq: 102 INVITE
    Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    Supported: replaces
    User-Agent: X-Lite release 5.4.0 stamp 94388
    Content-Length: 204
    
    v=0
    o=- 2754198276 3 IN IP4 192.168.3.27
    s=X-Lite release 5.4.0 stamp 94388
    c=IN IP4 192.168.3.27
    t=0 0
    m=audio 61040 RTP/AVP 0 8 101
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=sendrecv
    
    <------------->
    --- (12 headers 9 lines) ---
    Found RTP audio format 0
    Found RTP audio format 8
    Found RTP audio format 101
    Found audio description format telephone-event for ID 101
    Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
    Peer audio RTP is at port 192.168.3.27:61040
    list_route: hop: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
    set_destination: Parsing <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715> for address/port to send to
    set_destination: set destination to 192.168.3.27:1036
    Transmitting (no NAT) to 192.168.3.27:1036:
    ACK sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715 SIP/2.0
    Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK0f593c5b
    Max-Forwards: 70
    From: "dongle0" <sip:+380939333315@192.168.3.151>;tag=as74726125
    To: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>;tag=31158b0a
    Contact: <sip:+380939333315@192.168.3.151:5060>
    Call-ID: 2ff49f5d67e95131789afa962b474be1@192.168.3.151:5060
    CSeq: 102 ACK
    User-Agent: IPBX-2.11.0(11.25.3)
    Content-Length: 0
    
    
    ---
    [2019-01-22 19:32:13] WARNING[7523][C-00000036]: channel.c:1002 channel_indicate: [Dongle/dongle0-0100000021] Don't know how to indicate condition 33
    [2019-01-22 19:32:13] WARNING[7523][C-00000036]: channel.c:1002 channel_indicate: [Dongle/dongle0-0100000021] Don't know how to indicate condition 22
        -- SIP/102-00000014 answered Dongle/dongle0-0100000021
    
    <--- SIP read from UDP:192.168.3.27:1036 --->
    BYE sip:+380939333315@192.168.3.151:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.3.27:1036;branch=z9hG4bK-524287-1---70976675f8e00f05;rport
    Max-Forwards: 70
    Contact: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
    To: "dongle0" <sip:+380939333315@192.168.3.151>;tag=as74726125
    From: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>;tag=31158b0a
    Call-ID: 2ff49f5d67e95131789afa962b474be1@192.168.3.151:5060
    CSeq: 2 BYE
    User-Agent: X-Lite release 5.4.0 stamp 94388
    Content-Length: 0

    примерно от invite до bye.
    Это я зачитал когда позвонил на внутренний номер, при звонке на группу invite нет вообще.
  • Почему не работает группа номеров для входящих вызовов (issabel 4, asterisk 11)?

    Nazariy
    @Nazariy Автор вопроса
    cli> core set verbose 3 при звонке
    spoiler
    == Spawn extension (app-blackhole, hangup, 2) exited non-zero on 'Dongle/dongle0-0100000019'
        -- Executing [380935591582@from-pstn:1] Set("Dongle/dongle0-010000001a", "__FROM_DID=380935591582") in new stack
        -- Executing [380935591582@from-pstn:2] Gosub("Dongle/dongle0-010000001a", "app-blacklist-check,s,1()") in new stack
        -- Executing [s@app-blacklist-check:1] GotoIf("Dongle/dongle0-010000001a", "0?blacklisted") in new stack
        -- Executing [s@app-blacklist-check:2] Set("Dongle/dongle0-010000001a", "CALLED_BLACKLIST=1") in new stack
        -- Executing [s@app-blacklist-check:3] Return("Dongle/dongle0-010000001a", "") in new stack
        -- Executing [380935591582@from-pstn:3] Set("Dongle/dongle0-010000001a", "CDR(did)=380935591582") in new stack
        -- Executing [380935591582@from-pstn:4] ExecIf("Dongle/dongle0-010000001a", "0 ?Set(CALLERID(name)=+380939333315)") in new stack
        -- Executing [380935591582@from-pstn:5] Set("Dongle/dongle0-010000001a", "CHANNEL(musicclass)=default") in new stack
        -- Executing [380935591582@from-pstn:6] Set("Dongle/dongle0-010000001a", "__MOHCLASS=default") in new stack
        -- Executing [380935591582@from-pstn:7] Set("Dongle/dongle0-010000001a", "__CALLINGPRES_SV=allowed_not_screened") in new stack
        -- Executing [380935591582@from-pstn:8] Set("Dongle/dongle0-010000001a", "CALLERPRES()=allowed_not_screened") in new stack
        -- Executing [380935591582@from-pstn:9] Goto("Dongle/dongle0-010000001a", "ext-group,600,1") in new stack
        -- Goto (ext-group,600,1)
        -- Executing [600@ext-group:1] Macro("Dongle/dongle0-010000001a", "user-callerid,") in new stack
        -- Executing [s@macro-user-callerid:1] Set("Dongle/dongle0-010000001a", "TOUCH_MONITOR=1548176187.52") in new stack
        -- Executing [s@macro-user-callerid:2] Set("Dongle/dongle0-010000001a", "AMPUSER=+380939333315") in new stack
        -- Executing [s@macro-user-callerid:3] GotoIf("Dongle/dongle0-010000001a", "0?report") in new stack
        -- Executing [s@macro-user-callerid:4] ExecIf("Dongle/dongle0-010000001a", "1?Set(REALCALLERIDNUM=+380939333315)") in new stack
        -- Executing [s@macro-user-callerid:5] Set("Dongle/dongle0-010000001a", "AMPUSER=") in new stack
        -- Executing [s@macro-user-callerid:6] GotoIf("Dongle/dongle0-010000001a", "0?limit") in new stack
        -- Executing [s@macro-user-callerid:7] Set("Dongle/dongle0-010000001a", "AMPUSERCIDNAME=") in new stack
        -- Executing [s@macro-user-callerid:8] GotoIf("Dongle/dongle0-010000001a", "1?report") in new stack
        -- Goto (macro-user-callerid,s,15)
        -- Executing [s@macro-user-callerid:15] GotoIf("Dongle/dongle0-010000001a", "0?continue") in new stack
        -- Executing [s@macro-user-callerid:16] Set("Dongle/dongle0-010000001a", "__TTL=64") in new stack
        -- Executing [s@macro-user-callerid:17] GotoIf("Dongle/dongle0-010000001a", "1?continue") in new stack
        -- Goto (macro-user-callerid,s,28)
        -- Executing [s@macro-user-callerid:28] Set("Dongle/dongle0-010000001a", "CALLERID(number)=+380939333315") in new stack
        -- Executing [s@macro-user-callerid:29] Set("Dongle/dongle0-010000001a", "CALLERID(name)=dongle0") in new stack
        -- Executing [s@macro-user-callerid:30] Set("Dongle/dongle0-010000001a", "CDR(cnum)=+380939333315") in new stack
        -- Executing [s@macro-user-callerid:31] Set("Dongle/dongle0-010000001a", "CDR(cnam)=dongle0") in new stack
        -- Executing [s@macro-user-callerid:32] Set("Dongle/dongle0-010000001a", "CHANNEL(language)=en") in new stack
        -- Executing [600@ext-group:2] Macro("Dongle/dongle0-010000001a", "blkvm-setifempty,") in new stack
        -- Executing [s@macro-blkvm-setifempty:1] GotoIf("Dongle/dongle0-010000001a", "1?init") in new stack
        -- Goto (macro-blkvm-setifempty,s,4)
        -- Executing [s@macro-blkvm-setifempty:4] Set("Dongle/dongle0-010000001a", "__BLKVM_CHANNEL=Dongle/dongle0-010000001a") in new stack
        -- Executing [s@macro-blkvm-setifempty:5] Set("Dongle/dongle0-010000001a", "SHARED(BLKVM,Dongle/dongle0-010000001a)=TRUE") in new stack
        -- Executing [s@macro-blkvm-setifempty:6] Set("Dongle/dongle0-010000001a", "GOSUB_RETVAL=TRUE") in new stack
        -- Executing [s@macro-blkvm-setifempty:7] MacroExit("Dongle/dongle0-010000001a", "") in new stack
        -- Executing [600@ext-group:3] GotoIf("Dongle/dongle0-010000001a", "1?skipov") in new stack
        -- Goto (ext-group,600,6)
        -- Executing [600@ext-group:6] Set("Dongle/dongle0-010000001a", "RRNODEST=") in new stack
        -- Executing [600@ext-group:7] Set("Dongle/dongle0-010000001a", "__NODEST=600") in new stack
        -- Executing [600@ext-group:8] GosubIf("Dongle/dongle0-010000001a", "0?sub-rgsetcid,s,1()") in new stack
        -- Executing [600@ext-group:9] Gosub("Dongle/dongle0-010000001a", "sub-record-check,s,1(rg,600,dontcare)") in new stack
        -- Executing [s@sub-record-check:1] Set("Dongle/dongle0-010000001a", "REC_POLICY_MODE_SAVE=") in new stack
        -- Executing [s@sub-record-check:2] GotoIf("Dongle/dongle0-010000001a", "1?check") in new stack
        -- Goto (sub-record-check,s,7)
        -- Executing [s@sub-record-check:7] Set("Dongle/dongle0-010000001a", "__MON_FMT=wav") in new stack
        -- Executing [s@sub-record-check:8] GotoIf("Dongle/dongle0-010000001a", "1?next") in new stack
        -- Goto (sub-record-check,s,11)
        -- Executing [s@sub-record-check:11] ExecIf("Dongle/dongle0-010000001a", "0?Return()") in new stack
        -- Executing [s@sub-record-check:12] ExecIf("Dongle/dongle0-010000001a", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack
        -- Executing [s@sub-record-check:13] GotoIf("Dongle/dongle0-010000001a", "0?rg,1") in new stack
        -- Executing [s@sub-record-check:14] Set("Dongle/dongle0-010000001a", "__REC_STATUS=INITIALIZED") in new stack
        -- Executing [s@sub-record-check:15] Set("Dongle/dongle0-010000001a", "NOW=1548176187") in new stack
        -- Executing [s@sub-record-check:16] Set("Dongle/dongle0-010000001a", "__DAY=22") in new stack
        -- Executing [s@sub-record-check:17] Set("Dongle/dongle0-010000001a", "__MONTH=01") in new stack
        -- Executing [s@sub-record-check:18] Set("Dongle/dongle0-010000001a", "__YEAR=2019") in new stack
        -- Executing [s@sub-record-check:19] Set("Dongle/dongle0-010000001a", "__TIMESTR=20190122-185627") in new stack
        -- Executing [s@sub-record-check:20] Set("Dongle/dongle0-010000001a", "__FROMEXTEN=+380939333315") in new stack
        -- Executing [s@sub-record-check:21] Set("Dongle/dongle0-010000001a", "__CALLFILENAME=rg-600-+380939333315-20190122-185627-1548176187.52") in new stack
        -- Executing [s@sub-record-check:22] Goto("Dongle/dongle0-010000001a", "rg,1") in new stack
        -- Goto (sub-record-check,rg,1)
        -- Executing [rg@sub-record-check:1] GosubIf("Dongle/dongle0-010000001a", "0?record,1(rg,dontcare,+380939333315)") in new stack
        -- Executing [rg@sub-record-check:2] Return("Dongle/dongle0-010000001a", "") in new stack
        -- Executing [600@ext-group:10] Set("Dongle/dongle0-010000001a", "RingGroupMethod=ringall") in new stack
        -- Executing [600@ext-group:11] Macro("Dongle/dongle0-010000001a", "dial,80,Ttr,101-102-100") in new stack
        -- Executing [s@macro-dial:1] GotoIf("Dongle/dongle0-010000001a", "0?dial") in new stack
        -- Executing [s@macro-dial:2] Set("Dongle/dongle0-010000001a", "CHANNEL(musicclass)=default") in new stack
        -- Executing [s@macro-dial:3] AGI("Dongle/dongle0-010000001a", "dialparties.agi") in new stack
        -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
     dialparties.agi: Failed to execute '/var/lib/asterisk/agi-bin/dialparties.agi': Permission denied
        -- Executing [s@macro-dial:4] NoOp("Dongle/dongle0-010000001a", "Returned from dialparties with no extensions to call and DIALSTATUS: ") in new stack
        -- Executing [600@ext-group:12] Gosub("Dongle/dongle0-010000001a", "sub-record-cancel,s,1()") in new stack
        -- Executing [s@sub-record-cancel:1] Set("Dongle/dongle0-010000001a", "__REC_POLICY_MODE=") in new stack
        -- Executing [s@sub-record-cancel:2] ExecIf("Dongle/dongle0-010000001a", "1?Return()") in new stack
        -- Executing [600@ext-group:13] Set("Dongle/dongle0-010000001a", "RingGroupMethod=") in new stack
        -- Executing [600@ext-group:14] GotoIf("Dongle/dongle0-010000001a", "0?nodest") in new stack
        -- Executing [600@ext-group:15] Set("Dongle/dongle0-010000001a", "__NODEST=") in new stack
        -- Executing [600@ext-group:16] Macro("Dongle/dongle0-010000001a", "blkvm-clr,") in new stack
        -- Executing [s@macro-blkvm-clr:1] Set("Dongle/dongle0-010000001a", "SHARED(BLKVM,Dongle/dongle0-010000001a)=") in new stack
        -- Executing [s@macro-blkvm-clr:2] Set("Dongle/dongle0-010000001a", "GOSUB_RETVAL=") in new stack
        -- Executing [s@macro-blkvm-clr:3] MacroExit("Dongle/dongle0-010000001a", "") in new stack
        -- Executing [600@ext-group:17] Goto("Dongle/dongle0-010000001a", "app-blackhole,hangup,1") in new stack
        -- Goto (app-blackhole,hangup,1)
        -- Executing [hangup@app-blackhole:1] NoOp("Dongle/dongle0-010000001a", "Blackhole Dest: Hangup") in new stack
        -- Executing [hangup@app-blackhole:2] Hangup("Dongle/dongle0-010000001a", "") in new stack
      == Spawn extension (app-blackhole, hangup, 2) exited non-zero on 'Dongle/dongle0-010000001a'
  • Почему не работает группа номеров для входящих вызовов (issabel 4, asterisk 11)?

    Nazariy
    @Nazariy Автор вопроса
    Кусок Debug во время звонка, не ожидал что там всего так много..
    spoiler
    <------------->
    Reliably Transmitting (no NAT) to 192.168.3.27:1042:
    OPTIONS sip:101@192.168.3.27:1042;rinstance=47fbe190a6e5ef79 SIP/2.0
    Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK4d61c5f8
    Max-Forwards: 70
    From: "Unknown" <sip:Unknown@192.168.3.151>;tag=as38365f1f
    To: <sip:101@192.168.3.27:1042;rinstance=47fbe190a6e5ef79>
    Contact: <sip:Unknown@192.168.3.151:5060>
    Call-ID: 3a098cb85f913a1419719e150f57df29@192.168.3.151:5060
    CSeq: 102 OPTIONS
    User-Agent: IPBX-2.11.0(11.25.3)
    Date: Tue, 22 Jan 2019 16:45:27 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Content-Length: 0
    
    
    ---
    
    <--- SIP read from UDP:192.168.3.27:1042 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK4d61c5f8
    Contact: <sip:192.168.3.27:1042>
    To: <sip:101@192.168.3.27:1042;rinstance=47fbe190a6e5ef79>;tag=1b78e377
    From: "Unknown"<sip:Unknown@192.168.3.151>;tag=as38365f1f
    Call-ID: 3a098cb85f913a1419719e150f57df29@192.168.3.151:5060
    CSeq: 102 OPTIONS
    Accept: application/sdp
    Accept-Language: en
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Supported: replaces
    Allow-Events: presence, message-summary, tunnel-info
    Content-Length: 0
    
    <------------->
    --- (13 headers 0 lines) ---
    Really destroying SIP dialog '3a098cb85f913a1419719e150f57df29@192.168.3.151:5060' Method: OPTIONS
    
    <--- SIP read from UDP:192.168.3.27:1036 --->
    REGISTER sip:192.168.3.151 SIP/2.0
    Via: SIP/2.0/UDP 192.168.3.27:1036;branch=z9hG4bK-524287-1---c9ec176b5a12f111;rport
    Max-Forwards: 70
    Contact: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
    To: "102"<sip:102@192.168.3.151>
    From: "102"<sip:102@192.168.3.151>;tag=604c8e4f
    Call-ID: 94388MWRlNzM2ZWY0NDBlNThlNzViNDY3ZTRkY2QzZjQwMDA
    CSeq: 23 REGISTER
    Expires: 3600
    Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
    User-Agent: X-Lite release 5.4.0 stamp 94388
    Authorization: Digest username="102",realm="asterisk",nonce="62337258",uri="sip:192.168.3.151",response="9791ea3712f4b35a1c03370ccfeead2a",algorithm=MD5
    Content-Length: 0
    
    <------------->
    --- (13 headers 0 lines) ---
    Sending to 192.168.3.27:1036 (NAT)
    Sending to 192.168.3.27:1036 (NAT)
    
    <--- Transmitting (no NAT) to 192.168.3.27:1036 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 192.168.3.27:1036;branch=z9hG4bK-524287-1---c9ec176b5a12f111;received=192.168.3.27;rport=1036
    From: "102"<sip:102@192.168.3.151>;tag=604c8e4f
    To: "102"<sip:102@192.168.3.151>;tag=as36cc7be4
    Call-ID: 94388MWRlNzM2ZWY0NDBlNThlNzViNDY3ZTRkY2QzZjQwMDA
    CSeq: 23 REGISTER
    Server: IPBX-2.11.0(11.25.3)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3f4a54a0"
    Content-Length: 0
    
    
    <------------>
    Scheduling destruction of SIP dialog '94388MWRlNzM2ZWY0NDBlNThlNzViNDY3ZTRkY2QzZjQwMDA' in 32000 ms (Method: REGISTER)
    
    <--- SIP read from UDP:192.168.3.27:1036 --->
    REGISTER sip:192.168.3.151 SIP/2.0
    Via: SIP/2.0/UDP 192.168.3.27:1036;branch=z9hG4bK-524287-1---e099ef307998b60c;rport
    Max-Forwards: 70
    Contact: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
    To: "102"<sip:102@192.168.3.151>
    From: "102"<sip:102@192.168.3.151>;tag=604c8e4f
    Call-ID: 94388MWRlNzM2ZWY0NDBlNThlNzViNDY3ZTRkY2QzZjQwMDA
    CSeq: 24 REGISTER
    Expires: 3600
    Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
    User-Agent: X-Lite release 5.4.0 stamp 94388
    Authorization: Digest username="102",realm="asterisk",nonce="3f4a54a0",uri="sip:192.168.3.151",response="77826891b1876db5eb1c1cf864ccb4d4",algorithm=MD5
    Content-Length: 0
    
    <------------->
    --- (13 headers 0 lines) ---
    Sending to 192.168.3.27:1036 (no NAT)
    Reliably Transmitting (no NAT) to 192.168.3.27:1036:
    OPTIONS sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715 SIP/2.0
    Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK3ac2a7f8
    Max-Forwards: 70
    From: "Unknown" <sip:Unknown@192.168.3.151>;tag=as28bdd887
    To: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
    Contact: <sip:Unknown@192.168.3.151:5060>
    Call-ID: 25b7d5f45e13bb056cd0d2a02fa54cbe@192.168.3.151:5060
    CSeq: 102 OPTIONS
    User-Agent: IPBX-2.11.0(11.25.3)
    Date: Tue, 22 Jan 2019 16:45:28 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Content-Length: 0
    
    
    ---
    
    <--- Transmitting (no NAT) to 192.168.3.27:1036 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.3.27:1036;branch=z9hG4bK-524287-1---e099ef307998b60c;received=192.168.3.27;rport=1036
    From: "102"<sip:102@192.168.3.151>;tag=604c8e4f
    To: "102"<sip:102@192.168.3.151>;tag=as36cc7be4
    Call-ID: 94388MWRlNzM2ZWY0NDBlNThlNzViNDY3ZTRkY2QzZjQwMDA
    CSeq: 24 REGISTER
    Server: IPBX-2.11.0(11.25.3)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Expires: 3600
    Contact: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>;expires=3600
    Date: Tue, 22 Jan 2019 16:45:28 GMT
    Content-Length: 0
    
    
    <------------>
    Scheduling destruction of SIP dialog '070a43506ee481e75c4c7bc27936efb2@192.168.3.151:5060' in 6400 ms (Method: NOTIFY)
    Reliably Transmitting (no NAT) to 192.168.3.27:1036:
    NOTIFY sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715 SIP/2.0
    Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK43f52759
    Max-Forwards: 70
    From: "Unknown" <sip:Unknown@192.168.3.151>;tag=as23446dda
    To: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
    Contact: <sip:Unknown@192.168.3.151:5060>
    Call-ID: 070a43506ee481e75c4c7bc27936efb2@192.168.3.151:5060
    CSeq: 102 NOTIFY
    User-Agent: IPBX-2.11.0(11.25.3)
    Event: message-summary
    Content-Type: application/simple-message-summary
    Content-Length: 88
    
    Messages-Waiting: no
    Message-Account: sip:*97@192.168.3.151
    Voice-Message: 0/0 (0/0)
    
    ---
    Scheduling destruction of SIP dialog '94388MWRlNzM2ZWY0NDBlNThlNzViNDY3ZTRkY2QzZjQwMDA' in 32000 ms (Method: REGISTER)
    
    <--- SIP read from UDP:192.168.3.27:1036 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK3ac2a7f8
    Contact: <sip:192.168.3.27:1036>
    To: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>;tag=6bdf1067
    From: "Unknown" <sip:Unknown@192.168.3.151>;tag=as28bdd887
    Call-ID: 25b7d5f45e13bb056cd0d2a02fa54cbe@192.168.3.151:5060
    CSeq: 102 OPTIONS
    Accept: application/sdp
    Accept-Language: en
    Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
    Supported: replaces
    User-Agent: X-Lite release 5.4.0 stamp 94388
    Allow-Events: talk, hold
    Content-Length: 0
    
    <------------->
    --- (14 headers 0 lines) ---
    Really destroying SIP dialog '25b7d5f45e13bb056cd0d2a02fa54cbe@192.168.3.151:5060' Method: OPTIONS
    
    <--- SIP read from UDP:192.168.3.27:1036 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK43f52759
    Contact: <sip:192.168.3.27:1036>
    To: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>;tag=1ed0cc08
    From: "Unknown" <sip:Unknown@192.168.3.151>;tag=as23446dda
    Call-ID: 070a43506ee481e75c4c7bc27936efb2@192.168.3.151:5060
    CSeq: 102 NOTIFY
    User-Agent: X-Lite release 5.4.0 stamp 94388
    Content-Length: 0
    
    <------------->
    --- (9 headers 0 lines) ---
    Really destroying SIP dialog '070a43506ee481e75c4c7bc27936efb2@192.168.3.151:5060' Method: NOTIFY
    
    <--- SIP read from UDP:192.168.3.27:1042 --->
  • Почему не работает группа номеров для входящих вызовов (issabel 4, asterisk 11)?

    Nazariy
    @Nazariy Автор вопроса
    "Попрорбуйте изменить стратегию дозвона на "ringall" " - пробовал все варианты.
    "снимите дамп во время звонка" - как это сделать? В CLI при входящем звонке ничего не выводится, просто со стороны АТС ложится трубка.
  • Настройка gulp-rsync. Как исправить gulp-rsync: bash: rsync: command not found?

    Nazariy
    @Nazariy Автор вопроса
    а на хосте 195.191.24.138 rsync есть? Должен быть на обоих хостах

    Проблема оказалась на хостинге, уже все поправили.
    Спасибо большое за помощь!
  • Настройка gulp-rsync. Как исправить gulp-rsync: bash: rsync: command not found?

    Nazariy
    @Nazariy Автор вопроса
    rsync у вас из командной строки выполняется? Есть вообще такая команда?

    nazaru@DESKTOP-1U2HBA5:/mnt/c/OSPanel/domains/gsmrepair.loc$ rsync
    rsync  version 3.1.2  protocol version 31
    Copyright (C) 1996-2015 by Andrew Tridgell, Wayne Davison, and others.
    Web site: http://rsync.samba.org/
    Capabilities:
        64-bit files, 64-bit inums, 64-bit timestamps, 64-bit long ints,
        socketpairs, hardlinks, symlinks, IPv6, batchfiles, inplace,
        append, ACLs, xattrs, iconv, symtimes, prealloc
    
    rsync comes with ABSOLUTELY NO WARRANTY.  This is free software, and you
    are welcome to redistribute it under certain conditions.  See the GNU
    General Public Licence for details.
    
    rsync is a file transfer program capable of efficient remote update
    via a fast differencing algorithm.
    
    Usage: rsync [OPTION]... SRC [SRC]... DEST
      or   rsync [OPTION]... SRC [SRC]... [USER@]HOST:DEST
      or   rsync [OPTION]... SRC [SRC]... [USER@]HOST::DEST
      or   rsync [OPTION]... SRC [SRC]... rsync://[USER@]HOST[:PORT]/DEST
      or   rsync [OPTION]... [USER@]HOST:SRC [DEST]
      or   rsync [OPTION]... [USER@]HOST::SRC [DEST]
      or   rsync [OPTION]... rsync://[USER@]HOST[:PORT]/SRC [DEST]
    The ':' usages connect via remote shell, while '::' & 'rsync://' usages connect
    to an rsync daemon, and require SRC or DEST to start with a module name.
    
    Use "rsync --daemon --help" to see the daemon-mode command-line options.
    Please see the rsync(1) and rsyncd.conf(5) man pages for full documentation.
    See http://rsync.samba.org/ for updates, bug reports, and answers
    rsync error: syntax or usage error (code 1) at main.c(1569) [client=3.1.2]

    я с лога Options удалил то сильно длинный код получился
  • Настройка gulp-rsync. Как исправить gulp-rsync: bash: rsync: command not found?

    Nazariy
    @Nazariy Автор вопроса
    из командной строки (терминал phpstorm).
    Выходит так:
    C:\OSPanel\domains\gsmrepair.loc> bash (перехожу в linux)
    nazaru@DESKTOP-1U2HBA5:/mnt/c/OSPanel/domains/gsmrepair.loc$ gulp deploy
    этим привожу в действие этот код:
    let gulp = require('gulp'),    
        rsync = require('gulp-rsync');
    
    gulp.task('deploy', function() {
      return gulp.src('test_rsync/**')
        .pipe(rsync({
          //root: 'test_rsync/',
          hostname: 'gsmrepairco@195.191.24.138',
          destination: 'public_html/temp',
          archive: true,
          silent: false,
          compress: true,
        }));
    });