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Discussion: Elastix + ippi

  1. #1
    Membre Junior
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    juillet 2012
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    Exclamation Elastix + ippi

    Bonjour,
    Je suis nouveau en téléphonie sur ip, j ai installer un serveur avec elastix, j ai pris une offre ippi.

    J'ai 2 téléphone aastra 6731i + soft phones.

    Les téléphones communique via asterisk, ca jusque la ca marche.

    Avec Ippi,

    SIP TRUNK:
    ippi- out
    username=bsm
    type=peer
    secret=password
    nat=yes
    host=ippi.fr
    fromuser=bsm
    fromdomain=ippi.fr
    canreinvite=no

    ippi in
    type=peer
    host=ippi.fr
    nat=yes
    insecure=port,invite
    context=from-trunk
    canreinvite=no
    qualify=yes

    registrer
    bsm:password@ippi.fr

    sip show registry
    ippi.fr:5060 N bsm 105 Registered Tue, 27 Jan 2015 16:51:09
    1 SIP registrations.

    sip show peers
    1000/1000 192.168.1.10 D No No A 49765 OK (5 ms) softphone
    1001/1001 192.168.1.120 D No No A 5060 OK (9 ms) aastra
    1002/1002 192.168.1.121 D No No A 5060 OK (24 ms) aastra
    ippi-in 213.xxx.xxx.xxx Yes Yes 5060 OK (130 ms)
    ippi-out/bsm 213.xxx.xxx.xxx Yes Yes 5060 Unmonitored

    sip show channels
    192.168.1.121 (None) faa82903030xxxx (nothing) No Rx: REGISTER <guest>
    213.xxx.xxx.xxx (None) 89d65087-d5xxxx (nothing) No Rx: OPTIONS <guest>
    2 active SIP dialogs

    Je n'arrive ni a sortir un appel, ni rentrer un appel,

    Sur l interface ippi, dans mes connections, ippi voit bien mon server
    Sur mes téléphone, j ai juste mis l extension d elastix , le password et l ip du server, je n'ai rien renseigner d ippi

    Quelqu'un pourrais me donner un coup de main?

  2. #2
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    novembre 2013
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    Bonjour
    Je suis surpris qu'il y ai dans le "sip show peers" un trunk ippi-in et un trunk ippi-out, c'est peut être normal sur certaines versions mais je n'ai jamais vu ça.

    Sinon, as tu bien crée les routes entrantes/sortantes ?

    Si oui, il faudrait montrer ce qui passe dans les logs au moment d'un appel entrant ou sortant.

  3. #3
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    J'ai la dernière version de Elastix 2.5.0-2

    Route sortante:
    j ai mis juste dans thunk sequence for matched routes: 0--> mon trunk sip
    et optional destination on congestion --> trunks --> mon trunk sip

    Route sortante:
    N] sda => 334.... mon tel
    prefixe au cid -->sda1
    choix destination --> trunks --> mon sip

    Log appel


    <--- Transmitting (no NAT) to 213.215.45.230:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 213.215.45.230:5060;branch=0;received=213.215.45.2 30
    From: sip:pinger@ippi.fr;tag=00e77702
    To: sip:89.2.111.99:55024;tag=as38c8c520
    Call-ID: 89d65087-43642f87-2cc42@213.215.45.230
    CSeq: 1 OPTIONS
    Server: FPBX-2.11.0(11.14.1)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:192.168.1.37:5060>
    Accept: application/sdp
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog '89d65087-43642f87-2cc42@213.215.45.230' in 32000 ms (Method: OPTIONS)
    Really destroying SIP dialog '89d65087-64632f87-4ac42@213.215.45.230' Method: OPTIONS

    <--- SIP read from UDP:192.168.1.121:1036 --->
    INVITE sip:0616251476@192.168.1.37:5060;user=phone SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.121;branch=z9hG4bK5609e37ed3e378bda.0c10 6e28a4c2b7831
    Route: <sip:192.168.1.37:5060;lr>
    Max-Forwards: 70
    From: "1002" <sip:1002@192.168.1.37:5060>;tag=d9fef05245
    To: <sip:0616251476@192.168.1.37:5060;user=phone>
    Call-ID: f356d01b17c88310
    CSeq: 7358 INVITE
    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
    Allow-Events: talk, hold, conference, LocalModeStatus
    Contact: "1002" <sip:1002@192.168.1.121:5060;transport=udp>;+sip.i nstance="<urn:uuid:00000000-0000-1000-8000-00085D3F115B>"
    Supported: path, 100rel, replaces
    User-Agent: Aastra 6731i/3.2.2.3077
    Content-Type: application/sdp
    Content-Length: 616

    v=0
    o=MxSIP 0 1 IN IP4 192.168.1.121
    s=SIP Call
    c=IN IP4 89.2.111.99
    t=0 0
    m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:106 BV16/8000
    a=rtpmap:107 BV32/16000
    a=rtpmap:113 L16/16000
    a=rtpmap:110 PCMU/16000
    a=rtpmap:111 PCMA/16000
    a=rtpmap:112 L16/8000
    a=rtpmap:98 G726-16/8000
    a=rtpmap:97 G726-24/8000
    a=rtpmap:115 G726-32/8000
    a=rtpmap:96 G726-40/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=silenceSupp:on - - - -
    a=fmtp:18 annexb=yes
    a=fmtp:101 0-15
    a=ptime:30
    a=sendrecv
    <------------->
    --- (15 headers 26 lines) ---
    Sending to 192.168.1.121:5060 (no NAT)
    Sending to 192.168.1.121:5060 (no NAT)
    Using INVITE request as basis request - f356d01b17c88310
    Found peer '1002' for '1002' from 192.168.1.121:1036

    <--- Reliably Transmitting (no NAT) to 192.168.1.121:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 192.168.1.121;branch=z9hG4bK5609e37ed3e378bda.0c10 6e28a4c2b7831;received=192.168.1.121
    From: "1002" <sip:1002@192.168.1.37:5060>;tag=d9fef05245
    To: <sip:0616251476@192.168.1.37:5060;user=phone>;tag= as027f963c
    Call-ID: f356d01b17c88310
    CSeq: 7358 INVITE
    Server: FPBX-2.11.0(11.14.1)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="01fa75c8"
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog 'f356d01b17c88310' in 6400 ms (Method: INVITE)

    <--- SIP read from UDP:192.168.1.121:1036 --->
    ACK sip:0616251476@192.168.1.37:5060;user=phone SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.121;branch=z9hG4bK5609e37ed3e378bda.0c10 6e28a4c2b7831
    Route: <sip:192.168.1.37:5060;lr>
    Max-Forwards: 70
    From: "1002" <sip:1002@192.168.1.37:5060>;tag=d9fef05245
    To: <sip:0616251476@192.168.1.37:5060;user=phone>;tag= as027f963c
    Call-ID: f356d01b17c88310
    CSeq: 7358 ACK
    User-Agent: Aastra 6731i/3.2.2.3077
    Content-Length: 0

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

    <--- SIP read from UDP:192.168.1.121:1036 --->
    INVITE sip:0616251476@192.168.1.37:5060;user=phone SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.121;branch=z9hG4bKf92fdabc0a6cc4a76.ad42 85990873d9d57
    Route: <sip:192.168.1.37:5060;lr>
    Max-Forwards: 70
    From: "1002" <sip:1002@192.168.1.37:5060>;tag=d9fef05245
    To: <sip:0616251476@192.168.1.37:5060;user=phone>
    Call-ID: f356d01b17c88310
    CSeq: 7359 INVITE
    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
    Allow-Events: talk, hold, conference, LocalModeStatus
    Authorization: Digest username="1002",realm="asterisk",nonce="01fa75c8", uri="sip:0616251476@192.168.1.37:5060;user=phone", response="945b7445f18fb0655b39e321abe13d6e",algori thm=MD5
    Contact: "1002" <sip:1002@192.168.1.121:5060;transport=udp>;+sip.i nstance="<urn:uuid:00000000-0000-1000-8000-00085D3F115B>"
    Supported: path, 100rel, replaces
    User-Agent: Aastra 6731i/3.2.2.3077
    Content-Type: application/sdp
    Content-Length: 616

    v=0
    o=MxSIP 0 1 IN IP4 192.168.1.121
    s=SIP Call
    c=IN IP4 89.2.111.99
    t=0 0
    m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:106 BV16/8000
    a=rtpmap:107 BV32/16000
    a=rtpmap:113 L16/16000
    a=rtpmap:110 PCMU/16000
    a=rtpmap:111 PCMA/16000
    a=rtpmap:112 L16/8000
    a=rtpmap:98 G726-16/8000
    a=rtpmap:97 G726-24/8000
    a=rtpmap:115 G726-32/8000
    a=rtpmap:96 G726-40/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=silenceSupp:on - - - -
    a=fmtp:18 annexb=yes
    a=fmtp:101 0-15
    a=ptime:30
    a=sendrecv
    <------------->
    --- (16 headers 26 lines) ---
    Sending to 192.168.1.121:5060 (no NAT)
    Using INVITE request as basis request - f356d01b17c88310
    Found peer '1002' for '1002' from 192.168.1.121:1036
    Found RTP audio format 0
    Found RTP audio format 18
    Found RTP audio format 106
    Found RTP audio format 107
    Found RTP audio format 113
    Found RTP audio format 110
    Found RTP audio format 111
    Found RTP audio format 112
    Found RTP audio format 98
    Found RTP audio format 97
    Found RTP audio format 115
    Found RTP audio format 96
    Found RTP audio format 9
    Found RTP audio format 8
    Found RTP audio format 101
    Found audio description format PCMU for ID 0
    Found audio description format G729 for ID 18
    Found unknown media description format BV16 for ID 106
    Found unknown media description format BV32 for ID 107
    Found audio description format L16 for ID 113
    Found unknown media description format PCMU for ID 110
    Found unknown media description format PCMA for ID 111
    Found audio description format L16 for ID 112
    Found unknown media description format G726-16 for ID 98
    Found unknown media description format G726-24 for ID 97
    Found audio description format G726-32 for ID 115
    Found unknown media description format G726-40 for ID 96
    Found audio description format G722 for ID 9
    Found audio description format PCMA for ID 8
    Found audio description format telephone-event for ID 101
    Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw|g726|slin|g729|g722|slin16)/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 89.2.111.99:3000
    Looking for 0616251476 in from-internal (domain 192.168.1.37)
    list_route: hop: <sip:1002@192.168.1.121:5060;transport=udp>

    <--- Transmitting (no NAT) to 192.168.1.121:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.1.121;branch=z9hG4bKf92fdabc0a6cc4a76.ad42 85990873d9d57;received=192.168.1.121
    From: "1002" <sip:1002@192.168.1.37:5060>;tag=d9fef05245
    To: <sip:0616251476@192.168.1.37:5060;user=phone>
    Call-ID: f356d01b17c88310
    CSeq: 7359 INVITE
    Server: FPBX-2.11.0(11.14.1)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:0616251476@192.168.1.37:5060>
    Content-Length: 0


    <------------>
    Audio is at 11672
    Adding codec 100004 (alaw) to SDP
    Adding codec 100008 (g729) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (no NAT) to 213.215.45.230:5060:
    INVITE sip:0616251476@ippi.fr SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK1f8ddc17
    Max-Forwards: 70
    From: "bsmnimes" <sip:bsmnimes@192.168.1.37>;tag=as5756b27b
    To: <sip:0616251476@ippi.fr>
    Contact: <sip:bsmnimes@192.168.1.37:5060>
    Call-ID: 551dadc303ea8819446183030592179a@192.168.1.37:5060
    CSeq: 102 INVITE
    User-Agent: FPBX-2.11.0(11.14.1)
    Date: Wed, 28 Jan 2015 11:14:14 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 283

  4. #4
    Membre Junior
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    v=0
    o=root 2083461824 2083461824 IN IP4 192.168.1.37
    s=Asterisk PBX 11.14.1
    c=IN IP4 192.168.1.37
    t=0 0
    m=audio 11672 RTP/AVP 8 18 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv

    ---

    <--- SIP read from UDP:213.215.45.230:5060 --->
    SIP/2.0 403 GW call denied
    Via: SIP/2.0/UDP 192.168.1.37:5060;received=89.2.111.99;rport=55024 ;branch=z9hG4bK1f8ddc17
    From: "bsmnimes" <sip:bsmnimes@192.168.1.37>;tag=as5756b27b
    To: <sip:0616251476@ippi.fr>;tag=a910c8153188470b28416 23c513a131f.f481
    Call-ID: 551dadc303ea8819446183030592179a@192.168.1.37:5060
    CSeq: 102 INVITE
    Server: OpenSIPS (1.8.2-tls (i386/linux))
    Content-Length: 0

    <------------->
    --- (8 headers 0 lines) ---
    Transmitting (no NAT) to 213.215.45.230:5060:
    ACK sip:0616251476@ippi.fr SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK1f8ddc17
    Max-Forwards: 70
    From: "bsmnimes" <sip:bsmnimes@192.168.1.37>;tag=as5756b27b
    To: <sip:0616251476@ippi.fr>;tag=a910c8153188470b28416 23c513a131f.f481
    Contact: <sip:bsmnimes@192.168.1.37:5060>
    Call-ID: 551dadc303ea8819446183030592179a@192.168.1.37:5060
    CSeq: 102 ACK
    User-Agent: FPBX-2.11.0(11.14.1)
    Content-Length: 0


    ---
    [2015-01-28 12:14:14] WARNING[2828][C-00000007]: chan_sip.c:23027 handle_response_invite: Received response: "Forbidden" from '"bsmnimes" <sip:bsmnimes@192.168.1.37>;tag=as5756b27b'
    Scheduling destruction of SIP dialog '551dadc303ea8819446183030592179a@192.168.1.37:506 0' in 32000 ms (Method: INVITE)
    Audio is at 11830
    Adding codec 100004 (alaw) to SDP
    Adding codec 100008 (g729) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (no NAT) to 213.215.45.230:5060:
    INVITE sip:ippi.fr SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK0a8ef638
    Max-Forwards: 70
    From: "TECHNIQUE" <sip:bsmnimes@192.168.1.37>;tag=as68744ae8
    To: <sip:ippi.fr>
    Contact: <sip:bsmnimes@192.168.1.37:5060>
    Call-ID: 4c3eb6796ccd8ac108ea1b810630e3dd@192.168.1.37:5060
    CSeq: 102 INVITE
    User-Agent: FPBX-2.11.0(11.14.1)
    Date: Wed, 28 Jan 2015 11:14:14 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 281

    v=0
    o=root 662821603 662821603 IN IP4 192.168.1.37
    s=Asterisk PBX 11.14.1
    c=IN IP4 192.168.1.37
    t=0 0
    m=audio 11830 RTP/AVP 8 18 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv

    ---

    <--- SIP read from UDP:213.215.45.230:5060 --->
    SIP/2.0 404 unknown service
    Via: SIP/2.0/UDP 192.168.1.37:5060;received=89.2.111.99;rport=55024 ;branch=z9hG4bK0a8ef638
    From: "TECHNIQUE" <sip:bsmnimes@192.168.1.37>;tag=as68744ae8
    To: <sip:ippi.fr>;tag=a910c8153188470b2841623c513a131f .3cd1
    Call-ID: 4c3eb6796ccd8ac108ea1b810630e3dd@192.168.1.37:5060
    CSeq: 102 INVITE
    Server: OpenSIPS (1.8.2-tls (i386/linux))
    Content-Length: 0

    <------------->
    --- (8 headers 0 lines) ---
    Transmitting (no NAT) to 213.215.45.230:5060:
    ACK sip:ippi.fr SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK0a8ef638
    Max-Forwards: 70
    From: "TECHNIQUE" <sip:bsmnimes@192.168.1.37>;tag=as68744ae8
    To: <sip:ippi.fr>;tag=a910c8153188470b2841623c513a131f .3cd1
    Contact: <sip:bsmnimes@192.168.1.37:5060>
    Call-ID: 4c3eb6796ccd8ac108ea1b810630e3dd@192.168.1.37:5060
    CSeq: 102 ACK
    User-Agent: FPBX-2.11.0(11.14.1)
    Content-Length: 0


    ---
    Scheduling destruction of SIP dialog '4c3eb6796ccd8ac108ea1b810630e3dd@192.168.1.37:506 0' in 32000 ms (Method: INVITE)
    Scheduling destruction of SIP dialog 'f356d01b17c88310' in 6400 ms (Method: INVITE)

    <--- Reliably Transmitting (no NAT) to 192.168.1.121:5060 --->
    SIP/2.0 404 Not Found
    Via: SIP/2.0/UDP 192.168.1.121;branch=z9hG4bKf92fdabc0a6cc4a76.ad42 85990873d9d57;received=192.168.1.121
    From: "1002" <sip:1002@192.168.1.37:5060>;tag=d9fef05245
    To: <sip:0616251476@192.168.1.37:5060;user=phone>;tag= as6417ee0f
    Call-ID: f356d01b17c88310
    CSeq: 7359 INVITE
    Server: FPBX-2.11.0(11.14.1)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Content-Length: 0


    <------------>

    <--- SIP read from UDP:192.168.1.121:1036 --->
    ACK sip:0616251476@192.168.1.37:5060;user=phone SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.121;branch=z9hG4bKf92fdabc0a6cc4a76.ad42 85990873d9d57
    Route: <sip:192.168.1.37:5060;lr>
    Max-Forwards: 70
    From: "1002" <sip:1002@192.168.1.37:5060>;tag=d9fef05245
    To: <sip:0616251476@192.168.1.37:5060;user=phone>;tag= as6417ee0f
    Call-ID: f356d01b17c88310
    CSeq: 7359 ACK
    User-Agent: Aastra 6731i/3.2.2.3077
    Content-Length: 0

  5. #5
    Membre Junior
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    <------------->
    --- (10 headers 0 lines) ---
    Really destroying SIP dialog 'faa82903030cd663' Method: REGISTER

    <--- SIP read from UDP:192.168.1.120:3225 --->
    REGISTER sip:192.168.1.37:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bKc0fa309f5c440d9a5
    Route: <sip:192.168.1.37:5060;lr>
    Max-Forwards: 70
    From: "1001" <sip:1001@192.168.1.37:5060>;tag=0086b14fa3
    To: "1001" <sip:1001@192.168.1.37:5060>
    Call-ID: 82c6e967c0f82f06
    CSeq: 43973 REGISTER
    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
    Allow-Events: talk, hold, conference, LocalModeStatus
    Authorization: Digest username="1001",realm="asterisk",nonce="19d20289", uri="sip:192.168.1.37:5060",response="1f1d761876bb 8acddb17e13f7f53824f",algorithm=MD5
    Contact: "1001" <sip:1001@192.168.1.120:5060;transport=udp>;+sip.i nstance="<urn:uuid:00000000-0000-1000-8000-00085D3F11A7>"
    Supported: path, gruu
    User-Agent: Aastra 6731i/3.2.2.3077
    Content-Length: 0

    <------------->
    --- (15 headers 0 lines) ---
    Sending to 192.168.1.120:5060 (no NAT)
    Sending to 192.168.1.120:5060 (no NAT)

    <--- Transmitting (no NAT) to 192.168.1.120:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bKc0fa309f5c440d9a5 ;received=192.168.1.120
    From: "1001" <sip:1001@192.168.1.37:5060>;tag=0086b14fa3
    To: "1001" <sip:1001@192.168.1.37:5060>;tag=as67cad833
    Call-ID: 82c6e967c0f82f06
    CSeq: 43973 REGISTER
    Server: FPBX-2.11.0(11.14.1)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1f695d73"
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog '82c6e967c0f82f06' in 32000 ms (Method: REGISTER)

    <--- SIP read from UDP:192.168.1.120:3225 --->
    REGISTER sip:192.168.1.37:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bK8babe24be6dbfaec0
    Route: <sip:192.168.1.37:5060;lr>
    Max-Forwards: 70
    From: "1001" <sip:1001@192.168.1.37:5060>;tag=0086b14fa3
    To: "1001" <sip:1001@192.168.1.37:5060>
    Call-ID: 82c6e967c0f82f06
    CSeq: 43974 REGISTER
    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
    Allow-Events: talk, hold, conference, LocalModeStatus
    Authorization: Digest username="1001",realm="asterisk",nonce="1f695d73", uri="sip:192.168.1.37:5060",response="80cf13c3c40e 1f8295d81b0d79b76b2b",algorithm=MD5
    Contact: "1001" <sip:1001@192.168.1.120:5060;transport=udp>;+sip.i nstance="<urn:uuid:00000000-0000-1000-8000-00085D3F11A7>"
    Supported: path, gruu
    User-Agent: Aastra 6731i/3.2.2.3077
    Content-Length: 0

    <------------->
    --- (15 headers 0 lines) ---
    Sending to 192.168.1.120:5060 (no NAT)
    Reliably Transmitting (no NAT) to 192.168.1.120:5060:
    OPTIONS sip:1001@192.168.1.120:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK4c39b038
    Max-Forwards: 70
    From: "Unknown" <sip:Unknown@192.168.1.37>;tag=as43139ce5
    To: <sip:1001@192.168.1.120:5060;transport=udp>
    Contact: <sip:Unknown@192.168.1.37:5060>
    Call-ID: 394609a3490804d8623a39245408239b@192.168.1.37:5060
    CSeq: 102 OPTIONS
    User-Agent: FPBX-2.11.0(11.14.1)
    Date: Wed, 28 Jan 2015 11:14:20 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Content-Length: 0


    ---

    <--- Transmitting (no NAT) to 192.168.1.120:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bK8babe24be6dbfaec0 ;received=192.168.1.120
    From: "1001" <sip:1001@192.168.1.37:5060>;tag=0086b14fa3
    To: "1001" <sip:1001@192.168.1.37:5060>;tag=as67cad833
    Call-ID: 82c6e967c0f82f06
    CSeq: 43974 REGISTER
    Server: FPBX-2.11.0(11.14.1)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Expires: 120
    Contact: <sip:1001@192.168.1.120:5060;transport=udp>;expire s=120
    Date: Wed, 28 Jan 2015 11:14:20 GMT
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog '4f5417b739d97bdb12893c8a72311750@192.168.1.37:506 0' in 6400 ms (Method: NOTIFY)
    Reliably Transmitting (no NAT) to 192.168.1.120:5060:
    NOTIFY sip:1001@192.168.1.120:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK4d50b6d6
    Max-Forwards: 70
    From: "Unknown" <sip:Unknown@192.168.1.37>;tag=as2a002cf2
    To: <sip:1001@192.168.1.120:5060;transport=udp>
    Contact: <sip:Unknown@192.168.1.37:5060>
    Call-ID: 4f5417b739d97bdb12893c8a72311750@192.168.1.37:5060
    CSeq: 102 NOTIFY
    User-Agent: FPBX-2.11.0(11.14.1)
    Event: message-summary
    Content-Type: application/simple-message-summary
    Content-Length: 87

    Messages-Waiting: no
    Message-Account: sip:*97@192.168.1.37
    Voice-Message: 0/0 (0/0)

    ---
    Scheduling destruction of SIP dialog '82c6e967c0f82f06' in 32000 ms (Method: REGISTER)

    <--- SIP read from UDP:192.168.1.120:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK4c39b038
    From: "Unknown" <sip:Unknown@192.168.1.37>;tag=as43139ce5
    To: <sip:1001@192.168.1.120:5060;transport=udp>;tag=65 5704210
    Call-ID: 394609a3490804d8623a39245408239b@192.168.1.37:5060
    CSeq: 102 OPTIONS
    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
    Server: Aastra 6731i/3.2.2.3077
    Supported: path
    Content-Length: 0

    <------------->
    --- (10 headers 0 lines) ---
    Really destroying SIP dialog '394609a3490804d8623a39245408239b@192.168.1.37:506 0' Method: OPTIONS

    <--- SIP read from UDP:192.168.1.120:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK4d50b6d6
    From: "Unknown" <sip:Unknown@192.168.1.37>;tag=as2a002cf2
    To: <sip:1001@192.168.1.120:5060;transport=udp>;tag=33 65219826
    Call-ID: 4f5417b739d97bdb12893c8a72311750@192.168.1.37:5060
    CSeq: 102 NOTIFY
    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
    Allow-Events: talk, hold, conference, LocalModeStatus
    Contact: "1001" <sip:1001@192.168.1.120:5060;transport=udp>;+sip.i nstance="<urn:uuid:00000000-0000-1000-8000-00085D3F11A7>"
    Server: Aastra 6731i/3.2.2.3077
    Supported: path
    Content-Length: 0

    <------------->
    --- (12 headers 0 lines) ---
    Really destroying SIP dialog '4f5417b739d97bdb12893c8a72311750@192.168.1.37:506 0' Method: NOTIFY
    Really destroying SIP dialog 'f356d01b17c88310' Method: ACK

  6. #6
    Membre Senior
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    Route sortante:
    j ai mis juste dans thunk sequence for matched routes: 0--> mon trunk sip
    et optional destination on congestion --> trunks --> mon trunk sip
    OK

    Route sortante:
    N] sda => 334.... mon tel
    prefixe au cid -->sda1
    choix destination --> trunks --> mon sip
    C'est pas plutôt route entrante ? Si oui pourquoi as tu mis le trunk comme destination, la destination devrait plutôt être l'extension interne.

    Je ne comprends pas ce Received response: "Forbidden" from '"bsmnimes" <sip:bsmnimes@192.168.1.37>. Cette IP, c'est bien l'IP locale du serveur ?

    Sinon, même si un trace SIP est utile pour une analyse fine, les logs de l'asterisk en lui même en sont général plus faciles a lire, je pense qu'il faut mieux commencer par regarder ces logs, avant de se plonger dans les traces SIP.

  7. #7
    Membre Junior
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    alors j ai mis en route entrante
    vers l extension de mon tel 1002

    tel 1002 ip 1.121
    serveur ip 1.37

  8. #8
    Membre Junior
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    quand j appelle de mon portable, vers mon sip
    j ai trouver une trace

    [2015-01-28 14:44:48] NOTICE[2005][C-00000002]: chan_sip.c:25545 handle_request_invite: Call from 'ippi-in' (213.215.45.230:5060) to extension 's' rejected because extension not found in context 'from_ippi'

    pour moi extension from_ippi, c'est ici, et la configuration viens du site ippi
    [ippi_incoming]
    ; configuration des appels entrants depuis ippi
    type=peer
    host=ippi.fr
    context=from_ippi
    nat=yes
    canreinvite=no

  9. #9
    Membre Junior
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    quand j'appelle

    voici les logs,

    [2015-01-28 15:03:55] WARNING[4474]: res_pjsip_pubsub.c:2858 pubsub_on_rx_publish_request: No registered publish handler for event presence
    [2015-01-28 15:03:55] WARNING[4474]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
    [2015-01-28 15:04:03] WARNING[4910][C-00000007]: channel.c:4816 ast_prod: Prodding channel 'PJSIP/1000-00000002' failed
    [2015-01-28 15:04:03] WARNING[1871]: res_pjsip_pubsub.c:2858 pubsub_on_rx_publish_request: No registered publish handler for event presence
    [2015-01-28 15:04:03] WARNING[1871]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo

  10. #10
    Membre Senior
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    Ok, as tu édité cette conf avec l'interface web ou en dur dans les fichiers de conf comme c'est dit sur le site de ippi ?

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