Ответы пользователя по тегу SIP
  • Asterisk 13 + Addpac. Почему не работает один из GSM каналов?

    gadzhi15
    @gadzhi15 Автор вопроса
    Дебаг на Addpac

    Received SIP PDU from ( 192.168.5.100:5182 )
    OPTIONS sip:79298835533@192.168.5.110 SIP/2.0
    Via: SIP/2.0/UDP 192.168.5.100:5182;branch=z9hG4bK38acdae6
    Max-Forwards: 70
    From: "asterisk" <sip:asterisk@192.168.5.100:5182>;tag=as73d493c2
    To: <sip:79298835533@192.168.5.110>
    Contact: <sip:asterisk@192.168.5.100:5182>
    Call-ID: 0cab9ef20d2b01d46b1eeae901b25c94@192.168.5.100:5182
    CSeq: 102 OPTIONS
    User-Agent: D-link
    Date: Wed, 09 Nov 2016 22:39:18 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Content-Length: 0
    
    
    
            Sending SIP PDU to ( 192.168.5.100:5182 ) from 5060
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.5.100:5182;branch=z9hG4bK38acdae6
    From: "asterisk" <sip:asterisk@192.168.5.100:5182>;tag=as73d493c2
    To: <sip:79298835533@192.168.5.110>
    Call-ID: 0cab9ef20d2b01d46b1eeae901b25c94@192.168.5.100:5182
    CSeq: 102 OPTIONS
    User-Agent: AddPac SIP Gateway
    Accept: application/sdp
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY
    Content-Length: 0
    
    
            Received SIP PDU from ( 192.168.5.100:5182 )
    OPTIONS sip:79640095533@192.168.5.110 SIP/2.0
    Via: SIP/2.0/UDP 192.168.5.100:5182;branch=z9hG4bK39ee3f91
    Max-Forwards: 70
    From: "asterisk" <sip:asterisk@192.168.5.100:5182>;tag=as61c3b219
    To: <sip:79640095533@192.168.5.110>
    Contact: <sip:asterisk@192.168.5.100:5182>
    Call-ID: 5fcfbaef466b8f0c55781c4f095366c7@192.168.5.100:5182
    CSeq: 102 OPTIONS
    User-Agent: D-link
    Date: Wed, 09 Nov 2016 22:39:18 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Content-Length: 0
    
    
    
            Sending SIP PDU to ( 192.168.5.100:5182 ) from 5060
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.5.100:5182;branch=z9hG4bK39ee3f91
    From: "asterisk" <sip:asterisk@192.168.5.100:5182>;tag=as61c3b219
    To: <sip:79640095533@192.168.5.110>
    Call-ID: 5fcfbaef466b8f0c55781c4f095366c7@192.168.5.100:5182
    CSeq: 102 OPTIONS
    User-Agent: AddPac SIP Gateway
    Accept: application/sdp
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY
    Content-Length: 0
    �


    Вот дебаг со второй симки которая работает
    135     <CEP    000100> : Call Received
    136     <CEP    000100> : Call Received
    137     <CEP    000100> : Call Initiated : calledNumber() crv(0) total(0)
    138     <Call   179>    : ******  Call Created status(InitiatedByMobile) ver(8.5
    139     <CEP    000100> : Decode CID : FFFFFF80  E 10  C 2B 37 39 36 37 39 33 32
    140     <CEP    000100> : Mobile CID : time() callingNumber(79679328252) calling
    141     <CEP    000100> : Calling number(79679328252)
    142     <CEP    000100> : Call id(ecb02358-e738-ab8a-8143-0002a409fd2e) callNum(
    143     <Call   179>    : MatchAllProcess After Sorted
                              <0>  id(2000) dest(T) prefer(0) selected(71)
    144     <Call   179>    : Initiate callee with dial-peer(T) status(CalleeDetermi
    145     <NetEP  179>    : InitiateOutCall: calledNum(79640095533) callingNum(796
    146     <NetEP  179>    : DoCall: calledAddr(sip:79640095533@192.168.5.100:5182)
    147     <SIP    179>    : SetLocalAudioFormats : outbound(TRUE) hqaEnable(FALSE)
    148     <SIP    179>    : SetLocalAudioFormats : myVoipPeer(2000) is not NULL, v
    149     <SIP    179>    : SetLocalAudioFormats : outbound(TRUE) hqaEnable(FALSE)
    150     <SIP    179>    : SetLocalAudioFormats : myVoipPeer(2000) is not NULL, v
    151     <SIP    0>      : No authentication information available
    152     <SIP    179>    : Send INVITE Request
    
            Sending SIP PDU to ( 192.168.5.100:5182 ) from 5060
    INVITE sip:79640095533@192.168.5.100:5182 SIP/2.0
    Via: SIP/2.0/UDP 192.168.5.110:5060;branch=z9hG4bKec58d244a4112
    From: <sip:79640095533@192.168.5.100>;tag=ec58d244a4
    To: <sip:79640095533@192.168.5.100:5182>
    Call-ID: ecb02358-9066-d215-8144-0002a409fd2e@192.168.5.110
    CSeq: 112 INVITE
    Supported: replaces, timer, 100rel, early-session
    Min-SE: 1800
    Date: Wed, 09 Nov 2016 23:27:40 GMT
    Session-Expires: 1800
    User-Agent: AddPac SIP Gateway
    Contact: <sip:79640095533@192.168.5.110>
    Accept: application/sdp
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO
    Content-Type: application/sdp
    Content-Length: 290
    Max-Forwards: 70
    Remote-Party-ID: <sip:79679328252@192.168.5.100>;screen=yes;party=calling
    
    v=0
    o=79640095533 1478734060 1478734060 IN IP4 192.168.5.110
    s=AddPac Gateway SDP
    c=IN IP4 192.168.5.110
    t=1478734060 0
    m=audio 23358 RTP/AVP 8 0 18 101
    a=ptime:20
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    
            Received SIP PDU from ( 192.168.5.100:5182 )
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.5.110:5060;branch=z9hG4bKec58d244a4112;received=192.168
    From: <sip:79640095533@192.168.5.100>;tag=ec58d244a4
    To: <sip:79640095533@192.168.5.100:5182>
    Call-ID: ecb02358-9066-d215-8144-0002a409fd2e@192.168.5.110
    CSeq: 112 INVITE
    Server: D-link
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
    Supported: replaces, timer
    Session-Expires: 1800;refresher=uas
    Contact: <sip:79640095533@192.168.5.100:5182>
    Content-Length: 0
    
    
    153     <SIP    179>    : Receive 100 Trying
    154     <SIP    179>    : Transaction (112 INVITE) proceeding
    
            Received SIP PDU from ( 192.168.5.100:5182 )
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 192.168.5.110:5060;branch=z9hG4bKec58d244a4112;received=192.168
    From: <sip:79640095533@192.168.5.100>;tag=ec58d244a4
    To: <sip:79640095533@192.168.5.100:5182>;tag=as14eba75c
    Call-ID: ecb02358-9066-d215-8144-0002a409fd2e@192.168.5.110
    CSeq: 112 INVITE
    Server: D-link
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
    Supported: replaces, timer
    Session-Expires: 1800;refresher=uas
    Contact: <sip:79640095533@192.168.5.100:5182>
    Content-Type: application/sdp
    Require: timer
    Content-Length: 275
    
    v=0
    o=root 984719863 984719863 IN IP4 192.168.5.100
    s=Asterisk PBX 13.11.2
    c=IN IP4 192.168.5.100
    t=0 0
    m=audio 6240 RTP/AVP 8 0 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=maxptime:150
    a=sendrecv
    
    155     <SIP    179>    : Receive 183 Session Progress
    156     <SIP    179>    : Transaction (112 INVITE) proceeding
    157     <SIP    179>    : Received Session Progress response
    158     <SIP    179>    : SetLocalAudioFormats : outbound(TRUE) hqaEnable(FALSE)
    159     <SIP    179>    : SetLocalAudioFormats : myVoipPeer(2000) is not NULL, v
    160     <SIP    179>    : Get SIP Audio MediaFormat : 8
    161     <Call   179>    : PreConnected from(fffffffe)
    162     <NetCon 179>    : Alert received (inband tone explicitly).
    163     <Call   179>    : Alert from(fffffffe) pseudo(0) inband(1) status(Callee
    164     <SIP    178>    : Set Terminated Success for 111 INVITE
    
            Received SIP PDU from ( 192.168.5.100:5182 )
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.5.110:5060;branch=z9hG4bKec58d244a4112;received=192.168
    From: <sip:79640095533@192.168.5.100>;tag=ec58d244a4
    To: <sip:79640095533@192.168.5.100:5182>;tag=as14eba75c
    Call-ID: ecb02358-9066-d215-8144-0002a409fd2e@192.168.5.110
    CSeq: 112 INVITE
    Server: D-link
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
    Supported: replaces, timer
    Session-Expires: 1800;refresher=uas
    Contact: <sip:79640095533@192.168.5.100:5182>
    Content-Type: application/sdp
    Require: timer
    Content-Length: 275
    
    v=0
    o=root 984719863 984719863 IN IP4 192.168.5.100
    s=Asterisk PBX 13.11.2
    c=IN IP4 192.168.5.100
    t=0 0
    m=audio 6240 RTP/AVP 8 0 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=maxptime:150
    a=sendrecv
    
    165     <SIP    179>    : Receive 200 OK
    166     <SIP    179>    : Received INVITE OK response
    167     <SIP    179>    : Send ACK Request
    
            Sending SIP PDU to ( 192.168.5.100:5182 ) from 5060
    ACK sip:79640095533@192.168.5.100:5182 SIP/2.0
    Via: SIP/2.0/UDP 192.168.5.110:5060;branch=z9hG4bKec58d244a4112
    From: <sip:79640095533@192.168.5.100>;tag=ec58d244a4
    To: <sip:79640095533@192.168.5.100:5182>;tag=as14eba75c
    Call-ID: ecb02358-9066-d215-8144-0002a409fd2e@192.168.5.110
    CSeq: 112 ACK
    Content-Length: 0
    Max-Forwards: 70
    
    168     <SIP    179>    : SetLocalAudioFormats : outbound(TRUE) hqaEnable(FALSE)
    169     <SIP    179>    : SetLocalAudioFormats : myVoipPeer(2000) is not NULL, v
    170     <SIP    179>    : Get SIP Audio MediaFormat : 8
    171     <Call   179>    : Connected from(fffffffe)
    172     <NetEP  179>    : Call with sip:79640095533@192.168.5.100:5182 establish
    173     <SIP    179>    : Check Event Relation code(200)
    174     <SIP    179>    : Set Terminated Success for 112 INVITE
    175     <CEP    000100> : Disconnected(16) at Busy
    176     <Call   179>    : Terminated from(100) this(Local:CallClear) before(NULL
    177     <CEP    000100> : DisconnectCall at Idle
    178     <SIP    179>    : ReleaseWithBYE
    179     <SIP    179>    : Send BYE Request
    Ответ написан
  • Asterisk 13 и Addpac.Почему не работают исходящие?

    gadzhi15
    @gadzhi15 Автор вопроса
    Разобрался. В диалплане Dial(SIP/7903481****/017${EXTEN:1:10},30,m)
    Указываем 017, так как в настройках addpac в translation rules указал 017T
    Ответ написан
  • Мульифон и Asterisk. Как решить проблему прерывистого голоса?

    gadzhi15
    @gadzhi15 Автор вопроса
    Так же периодически отваливается сам мультифон. sip show registry выводит Request sent и time out
    Ответ написан
  • Установка Asterisk+FreePBX, руки крюки?

    gadzhi15
    @gadzhi15
    По опыту, набил много шишек с этим FreePBX. Поставьте чистый Asterisk. Я вас уверяю, вы его полюбите. И желательно на CentOS.
    Ответ написан