• Определение звонка?

    @daler86 Автор вопроса
    Andrey Barbolin,
    Зарегистрирован внешний номер,
    Оператор компании,
    У меня модуль CDR выключен, случайно нету мануала или ссылки по настройке CDR????
    Написано
  • Не работает оповещение очереди?

    @daler86 Автор вопроса
    Andrey Barbolin, Repeat Frequency = 30 seconds ставил не работает голосовая очередь
    Написано
  • Нет входящего звука с аппаратов Yealink w60p?

    @daler86 Автор вопроса
    Andrey Barbolin,
    Огромное спасибо! Заработало!
    Написано
  • Нет входящего звука с аппаратов Yealink w60p?

    @daler86 Автор вопроса
    Andrey Barbolin,
    ParameterName : ParameterValue
    ====================================================
    100rel : yes
    accountcode :
    aggregate_mwi : true
    allow : (ulaw|alaw)
    allow_subscribe : true
    allow_transfer : true
    aors : 777
    auth : 777-auth
    call_group :
    callerid : "device" <777>
    callerid_privacy : allowed_not_screened
    callerid_tag :
    connected_line_method : invite
    context : from-internal
    cos_audio : 0
    cos_video : 0
    device_state_busy_at : 0
    direct_media : false
    direct_media_glare_mitigation : none
    direct_media_method : invite
    disable_direct_media_on_nat : false
    dtls_ca_file :
    dtls_ca_path :
    dtls_cert_file :
    dtls_cipher :
    dtls_fingerprint : SHA-256
    dtls_private_key :
    dtls_rekey : 0
    dtls_setup : active
    dtls_verify : No
    dtmf_mode : rfc4733
    fax_detect : false
    force_avp : false
    force_rport : true
    from_domain :
    from_user :
    g726_non_standard : false
    ice_support : false
    identify_by : username
    inband_progress : false
    language : ru
    mailboxes :
    media_address :
    media_encryption : no
    media_encryption_optimistic : false
    media_use_received_transport : false
    message_context :
    moh_suggest : default
    mwi_from_user :
    named_call_group :
    named_pickup_group :
    one_touch_recording : false
    outbound_auth :
    outbound_proxy :
    pickup_group :
    record_off_feature : automixmon
    record_on_feature : automixmon
    rewrite_contact : true
    rpid_immediate : false
    rtp_engine : asterisk
    rtp_ipv6 : false
    rtp_keepalive : 0
    rtp_symmetric : true
    rtp_timeout : 0
    rtp_timeout_hold : 0
    sdp_owner : -
    sdp_session : Asterisk
    send_diversion : true
    send_pai : true
    send_rpid : false
    set_var :
    srtp_tag_32 : false
    sub_min_expiry : 0
    t38_udptl : false
    t38_udptl_ec : none
    t38_udptl_ipv6 : false
    t38_udptl_maxdatagram : 0
    t38_udptl_nat : false
    timers : yes
    timers_min_se : 90
    timers_sess_expires : 1800
    tone_zone :
    tos_audio : 0
    tos_video : 0
    transport :
    trust_id_inbound : true
    trust_id_outbound : false
    use_avpf : false
    use_ptime : false
    user_eq_phone : false

    Не помогло
    Написано
  • Нет входящего звука с аппаратов Yealink w60p?

    @daler86 Автор вопроса
    Andrey Barbolin,
    asterisk -rx "pjsip show settings"

    Global Settings:

    ParameterName : ParameterValue
    ====================================================
    debug : no
    default_from_user : asterisk
    default_outbound_endpoint : dpma_endpoint
    endpoint_identifier_order : ip,username,anonymous
    keep_alive_interval : 0
    max_forwards : 70
    max_initial_qualify_time : 0
    user_agent : FPBX-13.0.192.16(13.7.1)

    System Settings:

    ParameterName : ParameterValue
    ==========================================
    compact_headers : false
    disable_tcp_switch : true
    threadpool_auto_increment : 5
    threadpool_idle_timeout : 60
    threadpool_initial_size : 0
    threadpool_max_size : 50
    timer_b : 32000
    timer_t1 : 500
    ----------------------------------

    asterisk -rx "pjsip show endpoint 777"

    Endpoint:
    I/OAuth:
    Aor:
    Contact:
    Transport:
    Identify:
    Match:
    Channel:
    Exten: CLCID:
    =========================================================================================

    Endpoint: 777/777 Not in use 0 of inf
    InAuth: 777-auth/777
    Aor: 777 1
    Contact: 777/sip:777@192.168.40.30:5060 3e409db323 Avail 5.890
    Identify: 777-identify/777

    ParameterName : ParameterValue
    ====================================================
    100rel : yes
    accountcode :
    aggregate_mwi : true
    allow : (ulaw|alaw)
    allow_subscribe : true
    allow_transfer : true
    aors : 777
    auth : 777-auth
    call_group :
    callerid : "device" <777>
    callerid_privacy : allowed_not_screened
    callerid_tag :
    connected_line_method : invite
    context : from-internal
    cos_audio : 0
    cos_video : 0
    device_state_busy_at : 0
    direct_media : true
    direct_media_glare_mitigation : none
    direct_media_method : invite
    disable_direct_media_on_nat : false
    dtls_ca_file :
    dtls_ca_path :
    dtls_cert_file :
    dtls_cipher :
    dtls_fingerprint : SHA-256
    dtls_private_key :
    dtls_rekey : 0
    dtls_setup : active
    dtls_verify : No
    dtmf_mode : rfc4733
    fax_detect : false
    force_avp : false
    force_rport : true
    from_domain :
    from_user :
    g726_non_standard : false
    ice_support : false
    identify_by : username
    inband_progress : false
    language : ru
    mailboxes :
    media_address :
    media_encryption : no
    media_encryption_optimistic : false
    media_use_received_transport : false
    message_context :
    moh_suggest : default
    mwi_from_user :
    named_call_group :
    named_pickup_group :
    one_touch_recording : false
    outbound_auth :
    outbound_proxy :
    pickup_group :
    record_off_feature : automixmon
    record_on_feature : automixmon
    rewrite_contact : true
    rpid_immediate : false
    rtp_engine : asterisk
    rtp_ipv6 : false
    rtp_keepalive : 0
    rtp_symmetric : true
    rtp_timeout : 0
    rtp_timeout_hold : 0
    sdp_owner : -
    sdp_session : Asterisk
    send_diversion : true
    send_pai : true
    send_rpid : false
    set_var :
    srtp_tag_32 : false
    sub_min_expiry : 0
    t38_udptl : false
    t38_udptl_ec : none
    t38_udptl_ipv6 : false
    t38_udptl_maxdatagram : 0
    t38_udptl_nat : false
    timers : yes
    timers_min_se : 90
    timers_sess_expires : 1800
    tone_zone :
    tos_audio : 0
    tos_video : 0
    transport :
    trust_id_inbound : true
    trust_id_outbound : false
    use_avpf : false
    use_ptime : false
    user_eq_phone : false
    Написано
  • Нет входящего звука с аппаратов Yealink w60p?

    @daler86 Автор вопроса
    roswell,
    Enable Outbound Proxy Server поставиль ip сервер FreePBX не помогло,
    и включиль 'NAT' такая же проблема .
    Написано
  • Нет входящего звука с аппаратов Yealink w60p?

    @daler86 Автор вопроса
    Andrey Barbolin,
    asterisk -rx "sip show settings"

    Global Settings:
    ----------------
    UDP Bindaddress: 0.0.0.0:5068
    TCP SIP Bindaddress: 0.0.0.0:5068
    TLS SIP Bindaddress: Disabled
    Videosupport: Yes
    Textsupport: No
    Ignore SDP sess. ver.: No
    AutoCreate Peer: Off
    Match Auth Username: No
    Allow unknown access: Yes
    Allow subscriptions: Yes
    Allow overlap dialing: Yes
    Allow promisc. redir: No
    Enable call counters: No
    SIP domain support: No
    Path support : No
    Realm. auth: No
    Our auth realm asterisk
    Use domains as realms: No
    Call to non-local dom.: Yes
    URI user is phone no: No
    Always auth rejects: Yes
    Direct RTP setup: No
    User Agent: FPBX-13.0.192.16(13.7.1)
    SDP Session Name: Asterisk PBX 13.7.1
    SDP Owner Name: root
    Reg. context: (not set)
    Regexten on Qualify: No
    Trust RPID: No
    Send RPID: No
    Legacy userfield parse: No
    Send Diversion: Yes
    Caller ID: Unknown
    From: Domain:
    Record SIP history: Off
    Auth. Failure Events: Off
    T.38 support: No
    T.38 EC mode: Unknown
    T.38 MaxDtgrm: 4294967295
    SIP realtime: Disabled
    Qualify Freq : 60000 ms
    Q.850 Reason header: No
    Store SIP_CAUSE: No

    Network QoS Settings:
    ---------------------------
    IP ToS SIP: CS3
    IP ToS RTP audio: EF
    IP ToS RTP video: AF41
    IP ToS RTP text: CS0
    802.1p CoS SIP: 4
    802.1p CoS RTP audio: 5
    802.1p CoS RTP video: 6
    802.1p CoS RTP text: 5
    Jitterbuffer enabled: No

    Network Settings:
    ---------------------------
    SIP address remapping: Enabled using externaddr
    Externhost:
    Externaddr: 15.12.55.5:0
    Externrefresh: 10
    Localnet: 192.168.40.0/255.255.255.0


    Global Signalling Settings:
    ---------------------------
    Codecs: (ulaw|alaw|h264|mpeg4)
    Relax DTMF: No
    RFC2833 Compensation: No
    Symmetric RTP: Yes
    Compact SIP headers: No
    RTP Keepalive: 0 (Disabled)
    RTP Timeout: 30
    RTP Hold Timeout: 300
    MWI NOTIFY mime type: application/simple-message-summary
    DNS SRV lookup: No
    Pedantic SIP support: Yes
    Reg. min duration 60 secs
    Reg. max duration: 3600 secs
    Reg. default duration: 120 secs
    Sub. min duration 60 secs
    Sub. max duration: 3600 secs
    Outbound reg. timeout: 20 secs
    Outbound reg. attempts: 0
    Outbound reg. retry 403:0
    Notify ringing state: Yes
    Include CID: No
    Notify hold state: Yes
    SIP Transfer mode: open
    Max Call Bitrate: 384 kbps
    Auto-Framing: No
    Outb. proxy:
    Session Timers: Accept
    Session Refresher: uas
    Session Expires: 1800 secs
    Session Min-SE: 90 secs
    Timer T1: 500
    Timer T1 minimum: 100
    Timer B: 32000
    No premature media: Yes
    Max forwards: 70

    Default Settings:
    -----------------
    Allowed transports: TCP
    Outbound transport: TCP
    Context: from-sip-external
    Record on feature: automon
    Record off feature: automon
    Force rport: Yes
    DTMF: rfc2833
    Qualify: 0
    Keepalive: 0
    Use ClientCode: No
    Progress inband: No
    Language: ru
    Tone zone:
    MOH Interpret: default
    MOH Suggest:
    Voice Mail Extension: *97
    ----
    asterisk -rx "sip show peer 777"
    Peer 777 not found.
    Написано