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Discussion: Afficher flux video d'une caméra dans Xlite

  1. #1
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    août 2011
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    Afficher flux video d'une caméra dans Xlite

    Bonjour,

    Je dispose d'une camera IP Mobotix, elle offre la particularité de gérer un compte sip et se comporte comme un téléphone. Je souhaiterais afficher la vidéo avec Xilte (d’après la doc c'est possible), j'ai tout bien paramétré dans sip.conf:

    videosupport=yes
    allow=h263

    Dans xlite j'ai gardé que le codec H263 (le seule dispo dans la camera).
    Lors d'un appel, la camera répond, j'ai la phonie mais de vidéo et au niveau CLI j'ai ca:


    -- Executing [5520@phones:1] Set("SIP/5501-0000000a", "CHANNEL(language)=fr") in new stack
    -- Executing [5520@phones:2] Dial("SIP/5501-0000000a", "SIP/5520,20") in new stack
    == Extension Changed 5501[subscriptions] new state InUse for Notify User 5503
    == Using SIP RTP CoS mark 5
    -- Called SIP/5520
    -- SIP/5520-0000000b is ringing
    -- SIP/5520-0000000b answered SIP/5501-0000000a
    -- Remotely bridging SIP/5501-0000000a and SIP/5520-0000000b
    [Nov 14 21:51:34] WARNING[3087]: chan_sip.c:8738 process_sdp: Unsupported SDP media type in offer: video 0 RTP/AVP 34
    == Spawn extension (phones, 5520, 2) exited non-zero on 'SIP/5501-0000000a'
    == Extension Changed 5501[subscriptions] new state Idle for Notify User 5503


    et pas de vidéo....
    Une idée ?

    Merci

  2. #2
    Membre Junior
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    La meme tace avec en plus les debug sip activés, ca serait plus pratique pour comprendre:
    Code:
    sip set debug on
    puis relancer le test
    Découvrez QualiSIP, http://www.open-appliance.com une solution de tests pour stresser vos infrastructures Télécom, et superviser la qualité de vos services voix.

  3. #3
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    Voila le résultat, je me connecte en ssh avec putty c'est le max que je puisse extraire:

    --- (14 headers 0 lines) ---
    Sending to 192.168.1.41:5060 (no NAT)
    Reliably Transmitting (no NAT) to 192.168.1.41:5060:
    OPTIONS sip:5503@192.168.1.41:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK3adf2a63
    Max-Forwards: 70
    From: "asterisk" <sip:asterisk@192.168.1.220>;tag=as7ac5f647
    To: <sip:5503@192.168.1.41:5060;transport=udp>
    Contact: <sip:asterisk@192.168.1.220:5060>
    Call-ID: 12cf549f6f453b9b134b5da37a7db21d@192.168.1.220:506 0
    CSeq: 102 OPTIONS
    User-Agent: Asterisk PBX 1.8.6.0
    Date: Tue, 15 Nov 2011 19:04:36 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Length: 0


    ---

    <--- Transmitting (no NAT) to 192.168.1.41:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.41;branch=z9hG4bK843c192f58aaf3a64;recei ved=192.168.1.41
    From: <sip:5503@192.168.1.220:5060>;tag=09aa42ea0c
    To: <sip:5503@192.168.1.220:5060>;tag=as042c4e18
    Call-ID: bf0fa8af85b3dda3
    CSeq: 100984 REGISTER
    Server: Asterisk PBX 1.8.6.0
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Expires: 120
    Contact: <sip:5503@192.168.1.41:5060;transport=udp>;expires =120
    Date: Tue, 15 Nov 2011 19:04:36 GMT
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog 'bf0fa8af85b3dda3' in 32000 ms (Method: REGISTER)

    <--- SIP read from UDP:192.168.1.41:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK3adf2a63
    From: "asterisk" <sip:asterisk@192.168.1.220>;tag=as7ac5f647
    To: <sip:5503@192.168.1.41:5060;transport=udp>;tag=362 1404576
    Call-ID: 12cf549f6f453b9b134b5da37a7db21d@192.168.1.220:506 0
    CSeq: 102 OPTIONS
    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
    Server: Aastra 55i/3.2.0.1011
    Supported: path
    Content-Length: 0

    <------------->
    --- (10 headers 0 lines) ---
    Really destroying SIP dialog '12cf549f6f453b9b134b5da37a7db21d@192.168.1.220:50 60' Method: OPTIONS

    <--- SIP read from UDP:192.168.1.11:45688 --->
    BYE sip:5520@192.168.1.220:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.11:45688;branch=z9hG4bK-d8754z-d464ac1d05087289-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:5501@192.168.1.11:45688;transport=udp>
    To: <sip:5520@192.168.1.220>;tag=as1606e0bb
    From: "5501"<sip:5501@192.168.1.220>;tag=ce578432
    Call-ID: MmQxNGNiMGEwYjFiZDM2MjQ3MzViNjA3ZDM3MTEwMGY.
    CSeq: 3 BYE
    User-Agent: X-Lite 4 release 4.1 stamp 63214
    Authorization: Digest username="5501",realm="asterisk",nonce="61f12305", uri="sip:5520@192.168.1.220:5060",response="184bcc 94cc53ebec83743cb0860427d7",algorithm=MD5
    Content-Length: 0

    <------------->
    --- (11 headers 0 lines) ---
    Sending to 192.168.1.11:45688 (no NAT)
    Scheduling destruction of SIP dialog 'MmQxNGNiMGEwYjFiZDM2MjQ3MzViNjA3ZDM3MTEwMGY.' in 32000 ms (Method: BYE)

    <--- Transmitting (no NAT) to 192.168.1.11:45688 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.11:45688;branch=z9hG4bK-d8754z-d464ac1d05087289-1---d8754z-;received=192.168.1.11;rport=45688
    From: "5501"<sip:5501@192.168.1.220>;tag=ce578432
    To: <sip:5520@192.168.1.220>;tag=as1606e0bb
    Call-ID: MmQxNGNiMGEwYjFiZDM2MjQ3MzViNjA3ZDM3MTEwMGY.
    CSeq: 3 BYE
    Server: Asterisk PBX 1.8.6.0
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Length: 0


    <------------>
    set_destination: Parsing <sip:5520@192.168.1.151:5060> for address/port to send to
    set_destination: set destination to 192.168.1.151:5060
    Audio is at 5060
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x4 (ulaw) to SDP
    Reliably Transmitting (no NAT) to 192.168.1.151:5060:
    INVITE sip:5520@192.168.1.151:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK1f095ee8
    Max-Forwards: 70
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
    To: <sip:5520@192.168.1.151:5060>;tag=393068593
    Contact: <sip:5501@192.168.1.220:5060>
    Call-ID: 219adec357d78ce34c2273335129e215@192.168.1.220:506 0
    CSeq: 105 INVITE
    User-Agent: Asterisk PBX 1.8.6.0
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    X-asterisk-Info: SIP re-invite (External RTP bridge)
    Content-Type: application/sdp
    Content-Length: 203

    v=0
    o=root 993097821 993097824 IN IP4 192.168.1.220
    s=Asterisk PBX 1.8.6.0
    c=IN IP4 192.168.1.220
    t=0 0
    m=audio 7872 RTP/AVP 8 0
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=sendrecv

    ---

    <--- SIP read from UDP:192.168.1.151:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK1f095ee8
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
    To: <sip:5520@192.168.1.151:5060>;tag=393068593
    Call-ID: 219adec357d78ce34c2273335129e215@192.168.1.220:506 0
    CSeq: 105 INVITE
    Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
    Content-Length: 0

    <------------->
    --- (8 headers 0 lines) ---

    <--- SIP read from UDP:192.168.1.151:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK1f095ee8
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
    To: <sip:5520@192.168.1.151:5060>;tag=393068593
    Call-ID: 219adec357d78ce34c2273335129e215@192.168.1.220:506 0
    CSeq: 105 INVITE
    Contact: <sip:5520@192.168.1.151:5060>
    Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
    Content-Type: application/sdp
    Content-Length: 179

    v=0
    o=userX 20000001 20000001 IN IP4 192.168.1.151
    s=Asterisk PBX 1.8.6.0
    c=IN IP4 192.168.1.151
    t=0 0
    m=audio 10500 RTP/AVP 8 0
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    <------------->
    --- (10 headers 8 lines) ---
    set_destination: Parsing <sip:5520@192.168.1.151:5060> for address/port to send to
    set_destination: set destination to 192.168.1.151:5060
    Transmitting (no NAT) to 192.168.1.151:5060:
    ACK sip:5520@192.168.1.151:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK49f26b0c
    Max-Forwards: 70
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
    To: <sip:5520@192.168.1.151:5060>;tag=393068593
    Contact: <sip:5501@192.168.1.220:5060>
    Call-ID: 219adec357d78ce34c2273335129e215@192.168.1.220:506 0
    CSeq: 105 ACK
    User-Agent: Asterisk PBX 1.8.6.0
    Content-Length: 0


    ---
    Scheduling destruction of SIP dialog '219adec357d78ce34c2273335129e215@192.168.1.220:50 60' in 6400 ms (Method: INVITE)
    set_destination: Parsing <sip:5520@192.168.1.151:5060> for address/port to send to
    set_destination: set destination to 192.168.1.151:5060
    Reliably Transmitting (no NAT) to 192.168.1.151:5060:
    BYE sip:5520@192.168.1.151:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK33cc2e46
    Max-Forwards: 70
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
    To: <sip:5520@192.168.1.151:5060>;tag=393068593
    Call-ID: 219adec357d78ce34c2273335129e215@192.168.1.220:506 0
    CSeq: 106 BYE
    User-Agent: Asterisk PBX 1.8.6.0
    X-Asterisk-HangupCause: Normal Clearing
    X-Asterisk-HangupCauseCode: 16
    Content-Length: 0


    ---
    == Spawn extension (phones, 5520, 2) exited non-zero on 'SIP/5501-00000019'
    set_destination: Parsing <sip:5503@192.168.1.41:5060;transport=udp> for address/port to send to
    set_destination: set destination to 192.168.1.41:5060
    Reliably Transmitting (no NAT) to 192.168.1.41:5060:
    NOTIFY sip:5503@192.168.1.41:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK73d30871
    Max-Forwards: 70
    From: <sip:5501@192.168.1.220:5060>;tag=as77c8c620
    To: <sip:5503@192.168.1.220:5060>;tag=560a44aa4f
    Contact: <sip:5501@192.168.1.220:5060>
    Call-ID: 4baa7bb7f47d8d4c
    Seq: 164 NOTIFY
    User-Agent: Asterisk PBX 1.8.6.0
    Subscription-State: active
    Event: dialog
    Content-Type: application/dialog-info+xml
    Content-Length: 209

    <?xml version="1.0"?><dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="62" state="full" entity="sip:5501@192.168.1.220:5060"><dialog id="5501"><state>terminated</state>
    </dialog>
    </dialog-info>

    ---
    == Extension Changed 5501[subscriptions] new state Idle for Notify User 5503

    <--- SIP read from UDP:192.168.1.151:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK33cc2e46
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
    To: <sip:5520@192.168.1.151:5060>;tag=393068593
    Call-ID: 219adec357d78ce34c2273335129e215@192.168.1.220:506 0
    CSeq: 106 BYE
    Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
    Content-Length: 0

    <------------->
    --- (8 headers 0 lines) ---

    <--- SIP read from UDP:192.168.1.151:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK33cc2e46
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
    To: <sip:5520@192.168.1.151:5060>;tag=393068593
    Call-ID: 219adec357d78ce34c2273335129e215@192.168.1.220:506 0
    CSeq: 106 BYE
    Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
    Content-Length: 0

    <------------->
    --- (8 headers 0 lines) ---
    Really destroying SIP dialog '219adec357d78ce34c2273335129e215@192.168.1.220:50 60' Method: INVITE

    <--- SIP read from UDP:192.168.1.41:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK73d30871
    From: <sip:5501@192.168.1.220:5060>;tag=as77c8c620
    To: <sip:5503@192.168.1.220:5060>;tag=560a44aa4f
    Call-ID: 4baa7bb7f47d8d4c
    CSeq: 164 NOTIFY
    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
    Allow-Events: talk, hold, conference, LocalModeStatus
    Server: Aastra 55i/3.2.0.1011
    Supported: path
    Content-Length: 0

    <------------->
    --- (11 headers 0 lines) ---
    SIP Response message for INCOMING dialog NOTIFY arrived
    serveur*CLI> sip set debug off
    SIP Debugging Disabled


    C'est un peu obscure pour moi, si ça vous parle n’hésitez pas a m'éclairer!!!!
    Merci

  4. #4
    Membre Senior
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    Il faudrait que tu débranches ton aastra 55i au moment ou tu fais ton test de visio entre ta caméra et ton PC.. et re-poster le résultat ..

  5. #5
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    Bonjour,

    J'ai déconnecté tous les postes sauf le SPA 3102 qui visiblement ne pollue pas trop le log, et voici tout ce que j'ai pu copier

    From: <sip:5520@192.168.1.220>;tag=as76046a24
    To: "5501"<sip:5501@192.168.1.220>;tag=e6894cea
    Contact: <sip:5520@192.168.1.220:5060>
    Call-ID: YTgzNzgwOWZjYmIyOWEwMmUzZjFjMjg3MTRmODcxMDg.
    CSeq: 102 ACK
    User-Agent: Asterisk PBX 1.8.6.0
    Content-Length: 0


    ---
    set_destination: Parsing <sip:5520@192.168.1.151:5060> for address/port to send to
    set_destination: set destination to 192.168.1.151:5060
    Audio is at 5060
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x4 (ulaw) to SDP
    Reliably Transmitting (no NAT) to 192.168.1.151:5060:
    INVITE sip:5520@192.168.1.151:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK36ef104f
    Max-Forwards: 70
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
    To: <sip:5520@192.168.1.151:5060>;tag=1292519073
    Contact: <sip:5501@192.168.1.220:5060>
    Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
    CSeq: 104 INVITE
    User-Agent: Asterisk PBX 1.8.6.0
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    X-asterisk-Info: SIP re-invite (External RTP bridge)
    Content-Type: application/sdp
    Content-Length: 202

    v=0
    o=root 752446342 752446344 IN IP4 192.168.1.11
    s=Asterisk PBX 1.8.6.0
    c=IN IP4 192.168.1.11
    t=0 0
    m=audio 59984 RTP/AVP 8 0
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=sendrecv

    ---

    <--- SIP read from UDP:192.168.1.151:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK36ef104f
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
    To: <sip:5520@192.168.1.151:5060>;tag=1292519073
    Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
    CSeq: 104 INVITE
    Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
    Content-Length: 0

    <------------->
    --- (8 headers 0 lines) ---

    <--- SIP read from UDP:192.168.1.151:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK36ef104f
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
    To: <sip:5520@192.168.1.151:5060>;tag=1292519073
    Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
    CSeq: 104 INVITE
    Contact: <sip:5520@192.168.1.151:5060>
    Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
    Content-Type: application/sdp
    Content-Length: 179

    v=0
    o=userX 20000001 20000001 IN IP4 192.168.1.151
    s=Asterisk PBX 1.8.6.0
    c=IN IP4 192.168.1.151
    t=0 0
    m=audio 10500 RTP/AVP 8 0
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    <------------->
    --- (10 headers 8 lines) ---
    set_destination: Parsing <sip:5520@192.168.1.151:5060> for address/port to send to
    set_destination: set destination to 192.168.1.151:5060
    Transmitting (no NAT) to 192.168.1.151:5060:
    ACK sip:5520@192.168.1.151:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK7ee8a187
    Max-Forwards: 70
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
    To: <sip:5520@192.168.1.151:5060>;tag=1292519073
    Contact: <sip:5501@192.168.1.220:5060>
    Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
    CSeq: 104 ACK
    User-Agent: Asterisk PBX 1.8.6.0
    Content-Length: 0


    ---

    <--- SIP read from UDP:192.168.1.11:45996 --->
    BYE sip:5520@192.168.1.220:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.11:45996;branch=z9hG4bK-d8754z-2173cd01d15af29a-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:5501@192.168.1.11:45996;transport=udp>
    To: <sip:5520@192.168.1.220>;tag=as76046a24
    From: "5501"<sip:5501@192.168.1.220>;tag=e6894cea
    Call-ID: YTgzNzgwOWZjYmIyOWEwMmUzZjFjMjg3MTRmODcxMDg.
    CSeq: 3 BYE
    User-Agent: X-Lite 4 release 4.1 stamp 63214
    Authorization: Digest username="5501",realm="asterisk",nonce="75510cc6", uri="sip:5520@192.168.1.220:5060",response="3e6e4c 61dbb314c9ae372b80e3f83b20",algorithm=MD5
    Content-Length: 0

    <------------->
    --- (11 headers 0 lines) ---
    Sending to 192.168.1.11:45996 (no NAT)
    Scheduling destruction of SIP dialog 'YTgzNzgwOWZjYmIyOWEwMmUzZjFjMjg3MTRmODcxMDg.' in 32000 ms (Method: BYE)

    <--- Transmitting (no NAT) to 192.168.1.11:45996 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.11:45996;branch=z9hG4bK-d8754z-2173cd01d15af29a-1---d8754z-;received=192.168.1.11;rport=45996
    From: "5501"<sip:5501@192.168.1.220>;tag=e6894cea
    To: <sip:5520@192.168.1.220>;tag=as76046a24
    Call-ID: YTgzNzgwOWZjYmIyOWEwMmUzZjFjMjg3MTRmODcxMDg.
    CSeq: 3 BYE
    Server: Asterisk PBX 1.8.6.0
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Length: 0


    <------------>
    set_destination: Parsing <sip:5520@192.168.1.151:5060> for address/port to send to
    set_destination: set destination to 192.168.1.151:5060
    Audio is at 5060
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x4 (ulaw) to SDP
    Reliably Transmitting (no NAT) to 192.168.1.151:5060:
    INVITE sip:5520@192.168.1.151:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK12711fef
    Max-Forwards: 70
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
    To: <sip:5520@192.168.1.151:5060>;tag=1292519073
    Contact: <sip:5501@192.168.1.220:5060>
    Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
    CSeq: 105 INVITE
    User-Agent: Asterisk PBX 1.8.6.0
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    X-asterisk-Info: SIP re-invite (External RTP bridge)
    Content-Type: application/sdp
    Content-Length: 204

    v=0
    o=root 752446342 752446345 IN IP4 192.168.1.220
    s=Asterisk PBX 1.8.6.0
    c=IN IP4 192.168.1.220
    t=0 0
    m=audio 11108 RTP/AVP 8 0
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=sendrecv

    ---

    <--- SIP read from UDP:192.168.1.151:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK12711fef
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
    To: <sip:5520@192.168.1.151:5060>;tag=1292519073
    Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
    CSeq: 105 INVITE
    Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
    Content-Length: 0

    <------------->
    --- (8 headers 0 lines) ---

    <--- SIP read from UDP:192.168.1.151:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK12711fef
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
    To: <sip:5520@192.168.1.151:5060>;tag=1292519073
    Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
    CSeq: 105 INVITE
    Contact: <sip:5520@192.168.1.151:5060>
    Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
    Content-Type: application/sdp
    Content-Length: 179

    v=0
    o=userX 20000001 20000001 IN IP4 192.168.1.151
    s=Asterisk PBX 1.8.6.0
    c=IN IP4 192.168.1.151
    t=0 0
    m=audio 10500 RTP/AVP 8 0
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    <------------->
    --- (10 headers 8 lines) ---
    set_destination: Parsing <sip:5520@192.168.1.151:5060> for address/port to send to
    set_destination: set destination to 192.168.1.151:5060
    Transmitting (no NAT) to 192.168.1.151:5060:
    ACK sip:5520@192.168.1.151:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK5ab0f8d3
    Max-Forwards: 70
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
    To: <sip:5520@192.168.1.151:5060>;tag=1292519073
    Contact: <sip:5501@192.168.1.220:5060>
    Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
    CSeq: 105 ACK
    User-Agent: Asterisk PBX 1.8.6.0
    Content-Length: 0


    ---
    Scheduling destruction of SIP dialog '498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:50 60' in 6400 ms (Method: INVITE)
    set_destination: Parsing <sip:5520@192.168.1.151:5060> for address/port to send to
    set_destination: set destination to 192.168.1.151:5060
    Reliably Transmitting (no NAT) to 192.168.1.151:5060:
    BYE sip:5520@192.168.1.151:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK19c8feec
    Max-Forwards: 70
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
    To: <sip:5520@192.168.1.151:5060>;tag=1292519073
    Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
    CSeq: 106 BYE
    User-Agent: Asterisk PBX 1.8.6.0
    X-Asterisk-HangupCause: Normal Clearing
    X-Asterisk-HangupCauseCode: 16
    Content-Length: 0


    ---
    == Spawn extension (phones, 5520, 2) exited non-zero on 'SIP/5501-0000002c'
    == Extension Changed 5501[subscriptions] new state Idle for Notify User 5503 (queued)

    <--- SIP read from UDP:192.168.1.151:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK19c8feec
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
    To: <sip:5520@192.168.1.151:5060>;tag=1292519073
    Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
    CSeq: 106 BYE
    Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
    Content-Length: 0

    <------------->
    --- (8 headers 0 lines) ---

    <--- SIP read from UDP:192.168.1.151:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK19c8feec
    From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
    To: <sip:5520@192.168.1.151:5060>;tag=1292519073
    Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
    CSeq: 106 BYE
    Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
    Content-Length: 0

    <------------->
    --- (8 headers 0 lines) ---
    Really destroying SIP dialog '498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:50 60' Method: INVITE

  6. #6
    Membre Senior
    Date d'inscription
    septembre 2010
    Messages
    410
    Downloads
    1
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    0
    Apparemment tu appelles avec X-Lite, on voit que X-Lite propose du G711u et G711a.

    Une fois que tu as décroché, est-ce que tu actives bien la video en affichant l'onglet video dans X-Lite ? Parce qu'il n'y a aucune trace de codec video dans le log que tu nous donnes ..

  7. #7
    Membre
    Date d'inscription
    août 2011
    Messages
    63
    Downloads
    1
    Uploads
    0
    Oui j'utilise même X lite 4 et il y a une fonction qui permet de faire un appel vidéo direct, du coup j'ai installé x lite sur un autre PC et la j'arrive a avoir la vidéo. Je pense que mon problème vient de la camera, je vais lui faire un petit retour usine et reprendre la configuration depuis le début j'ai probablement loupé quelque chose...

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