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Discussion: [RESOLU] [IPPI] Appel entrant

  1. #11
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    Citation Envoyé par jean Voir le message
    ok... effectivement, ca cause avec ton asterisk, c'est déjà ça :-)

    next... pousse le détail sur la CLI, doit y a avoir qque chose...
    core set verbose 9
    core set debug 9

    ensuite, .. as tu une extension _+. dans ton contexte ? vu que les numéros arrivent avec ce format ?
    Code:
    *CLI> core set verbose 9
    Verbosity was 7 and is now 9
    *CLI> core set debug 9
    Core debug was 0 and is now 9
    Comment fait-on une extention_+. ?????

  2. #12
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    ben... comme les autres....

    exten => _+.,1,Verbose(1, *** International Formati : 00${EXTEN:1})
    exten => _+.,n,Goto(00${EXTEN:1},1)

    (pour par exemple remplacer le + par 00 mais ce n'est peut etre pas ce que tu recherches)

  3. #13
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    J'ai activer "sip set debug" quand j'ai appellai, il marque:

    Code:
    <--- SIP read from UDP:213.215.45.230:5060 --->
    INVITE sip:+331********@***.***.***.*** SIP/2.0
    Record-Route: <sip:213.215.45.230;lr=on>
    Via: SIP/2.0/UDP 213.215.45.230;branch=z9hG4bK4a2.b4561c11.0
    Via: SIP/2.0/UDP 213.215.45.252:5060;received=213.215.45.252;branch=z9hG4bK2ca377dc;rport=5060
    From: "336********" <sip:336********@ippi.fr>;tag=as24278228
    To: <sip:01********@ippi.fr>
    Contact: <sip:336********@213.215.45.252>
    Call-ID: 15594c9b4ae5bf865302279c7c0cac77@ippi.fr
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 12
    Date: Tue, 28 Jun 2011 22:36:20 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Type: application/sdp
    Content-Length: 340
    DID-info: 336********
    
    v=0
    o=root 17453 17453 IN IP4 213.215.45.252
    s=session
    c=IN IP4 213.215.45.252
    t=0 0
    m=audio 18138 RTP/AVP 8 0 97 3 18 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:97 iLBC/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    
    <------------->
    --- (16 headers 15 lines) ---
      == Using SIP RTP CoS mark 5
    Sending to 213.215.45.230 : 5060 (no NAT)
    Using INVITE request as basis request - 15594c9b4ae5bf865302279c7c0cac77@ippi.fr
    Found peer 'ippi' for '336********' from 213.215.45.230:5060
    
    <--- Reliably Transmitting (NAT) to 213.215.45.230:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 213.215.45.230;branch=z9hG4bK4a2.b4561c11.0;received=213.215.45.230
    Via: SIP/2.0/UDP 213.215.45.252:5060;received=213.215.45.252;branch=z9hG4bK2ca377dc;rport=5060
    From: "336********" <sip:336********@ippi.fr>;tag=as24278228
    To: <sip:01********@ippi.fr>;tag=as0f438622
    Call-ID: 15594c9b4ae5bf865302279c7c0cac77@ippi.fr
    CSeq: 102 INVITE
    Server: Asterisk PBX 1.6.2.5-0ubuntu1.3
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="79decfb5"
    Content-Length: 0
    
    
    <------------>
    Scheduling destruction of SIP dialog '15594c9b4ae5bf865302279c7c0cac77@ippi.fr' in 6400 ms (Method: INVITE)
    Sediad*CLI> 
    <--- SIP read from UDP:213.215.45.230:5060 --->
    ACK sip:+331********@***.***.***.*** SIP/2.0
    Via: SIP/2.0/UDP 213.215.45.230;branch=z9hG4bK4a2.b4561c11.0
    From: "336********" <sip:336********@ippi.fr>;tag=as24278228
    Call-ID: 15594c9b4ae5bf865302279c7c0cac77@ippi.fr
    To: <sip:01********@ippi.fr>;tag=as0f438622
    CSeq: 102 ACK
    Max-Forwards: 70
    User-Agent: OpenSIPS (1.6.3-notls (i386/linux))
    Content-Length: 0
    
    
    <------------->
    --- (9 headers 0 lines) ---
    Sediad*CLI> 
    <--- SIP read from UDP:192.168.1.77:61198 --->
    
    
    
    <------------->
    Reliably Transmitting (NAT) to 192.168.1.77:61198:
    OPTIONS sip:tel1@192.168.1.***:61198;rinstance=54f821032d61eaf2 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK12134893;rport
    Max-Forwards: 70
    From: "asterisk" <sip:asterisk@192.168.1.100>;tag=as7544ff20
    To: <sip:tel1@192.168.1.***:61198;rinstance=54f821032d61eaf2>
    Contact: <sip:asterisk@192.168.1.100>
    Call-ID: 54a2557109e378b826294e30427c4caa@192.168.1.100
    CSeq: 102 OPTIONS
    User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.3
    Date: Tue, 28 Jun 2011 22:36:26 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Content-Length: 0
    
    
    ---
    Sediad*CLI> 
    <--- SIP read from UDP:192.168.1.***:61198 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK12134893;rport=5060
    Contact: <sip:192.168.1.***:61198>
    To: <sip:tel1@192.168.1.**:61198;rinstance=54f821032d61eaf2>;tag=209c83b7
    From: "asterisk"<sip:asterisk@192.168.1.100>;tag=as7544ff20
    Call-ID: 54a2557109e378b826294e30427c4caa@192.168.1.100
    CSeq: 102 OPTIONS
    Accept: application/sdp
    Accept-Language: en
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
    Supported: replaces
    User-Agent: X-Lite 4 release 4.0 stamp 58832
    Content-Length: 0
    
    
    <------------->
    --- (13 headers 0 lines) ---
    Really destroying SIP dialog '54a2557109e378b826294e30427c4caa@192.168.1.100' Method: OPTIONS
    Really destroying SIP dialog '15594c9b4ae5bf865302279c7c0cac77@ippi.fr' Method: ACK
    Mais sans plus

  4. #14
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    asterisk balance un unauthorized....

    next... dans le register,
    register => UTILISATEUR:MOT_DE_PASSE@ippi.fr/NUM_DE_TEL

    le /NUM_DE_TEL designe l'extension... essaie de virer le /num_tel, ou de voir à quoi correspond cette extension (dans le sip ou extensions.conf), et de mettre tes commandes dans ce contexte

  5. #15
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    J'ai retiré le /NUM_DE_TEL, nouveau bug:

    Code:
    <------------->
    --- (16 headers 15 lines) ---
      == Using SIP RTP CoS mark 5
    Sending to 213.215.45.230 : 5060 (no NAT)
    Using INVITE request as basis request - 318196c5321426a359e34d466b8a1fc0@ippi.fr
    Found peer 'ippi' for '331********' from 213.215.45.230:5060
    
    <--- Reliably Transmitting (NAT) to 213.215.45.230:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 213.215.45.230;branch=z9hG4bKffa.88fe2ed6.0;received=213.215.45.230
    Via: SIP/2.0/UDP 213.215.45.252:5060;received=213.215.45.252;branch=z9hG4bK152bcced;rport=5060
    From: "331********" <sip:331********@ippi.fr>;tag=as4acb879b
    To: <sip:01********@ippi.fr>;tag=as5bcdcfc8
    Call-ID: 318196c5321426a359e34d466b8a1fc0@ippi.fr
    CSeq: 102 INVITE
    Server: Asterisk PBX 1.6.2.5-0ubuntu1.3
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b0301d7"
    Content-Length: 0
    
    
    <------------>
    Scheduling destruction of SIP dialog '318196c5321426a359e34d466b8a1fc0@ippi.fr' in 6400 ms (Method: INVITE)
    
    <--- SIP read from UDP:213.215.45.230:5060 --->
    INVITE sip:+331********@***.***.***.*** SIP/2.0
    Record-Route: <sip:213.215.45.230;lr=on>
    Via: SIP/2.0/UDP 213.215.45.230;branch=z9hG4bKffa.88fe2ed6.1
    Via: SIP/2.0/UDP 213.215.45.252:5060;received=213.215.45.252;branch=z9hG4bK152bcced;rport=5060
    From: "331********" <sip:331********@ippi.fr>;tag=as4acb879b
    To: <sip:01********@ippi.fr>
    Contact: <sip:331********@213.215.45.252>
    Call-ID: 318196c5321426a359e34d466b8a1fc0@ippi.fr
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 12
    Date: Wed, 29 Jun 2011 06:14:11 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Type: application/sdp
    Content-Length: 340
    DID-info: 331********
    
    v=0
    o=root 17453 17453 IN IP4 213.215.45.252
    s=session
    c=IN IP4 213.215.45.252
    t=0 0
    m=audio 11904 RTP/AVP 8 0 97 3 18 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:97 iLBC/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    
    <------------->
    --- (16 headers 15 lines) ---
    Ignoring this INVITE request
    Citation Envoyé par jean Voir le message
    et de mettre tes commandes dans ce contexte
    ??? j'ai pas comprit ???
    Dernière modification par nyko77 ; 29/06/2011 à 08h17.

  6. #16
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    ???

  7. #17
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    => je vois pas le pbm dans le traces, le ignoring est plutot du à une retransmission, est ce systematique ?

    => le /NUM_TEL
    il faut je pense qu'il y ait une extension [NUM_TEL] dans le dialplan pour que cela marche...

    Code:
    ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
    ; Format for the register statement is:
    ;       register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
    ;

  8. #18
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    plus précisemment, dans le contexte de reception des appels, tu dois avoir l'extension /NUM_TEL
    exten => NUM_TEL,....

  9. #19
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    Citation Envoyé par jean Voir le message
    => je vois pas le pbm dans le traces, le ignoring est plutot du à une retransmission, est ce systematique ?
    Oui

    Pour le NUM_TEL je l'écrit de quelle manière: +331******** ou 00331******** ou 331******** ou 01******** ???

  10. #20
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    j'ai mit 336********
    quand j'appelle:
    Code:
    Asterisk 1.6.2.5-0ubuntu1.3, Copyright (C) 1999 - 2009 Digium, Inc. and others.
    Created by Mark Spencer <markster@digium.com>
    Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
    This is free software, with components licensed under the GNU General Public
    License version 2 and other licenses; you are welcome to redistribute it under
    certain conditions. Type 'core show license' for details.
    =========================================================================
      == Parsing '/etc/asterisk/asterisk.conf':   == Found
      == Parsing '/etc/asterisk/extconfig.conf':   == Found
    Connected to Asterisk 1.6.2.5-0ubuntu1.3 currently running on Sediad (pid = 741)
    Verbosity was 0 and is now 7
      == Using SIP RTP CoS mark 5
    Sediad*CLI>
    Mode debug:
    Code:
    SIP Debugging enabled
    Sediad*CLI> 
    <--- SIP read from UDP:213.215.45.230:5060 --->
    INVITE sip:331********@***.***.***.*** SIP/2.0
    Record-Route: <sip:213.215.45.230;lr=on>
    Via: SIP/2.0/UDP 213.215.45.230;branch=z9hG4bK4021.e4e3eba1.0
    Via: SIP/2.0/UDP 213.215.45.252:5060;received=213.215.45.252;branch=z9hG4bK4e499720;rport=5060
    From: "331********" <sip:331********@ippi.fr>;tag=as259f1013
    To: <sip:01********@ippi.fr>
    Contact: <sip:331********@213.215.45.252>
    Call-ID: 535a09dd7b98f0d231326e4613d941ab@ippi.fr
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 12
    Date: Fri, 01 Jul 2011 08:11:40 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Type: application/sdp
    Content-Length: 340
    DID-info: 331********
    
    v=0
    o=root 17453 17453 IN IP4 213.215.45.252
    s=session
    c=IN IP4 213.215.45.252
    t=0 0
    m=audio 19328 RTP/AVP 8 0 97 3 18 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:97 iLBC/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    
    <------------->
    --- (16 headers 15 lines) ---
      == Using SIP RTP CoS mark 5
    Sending to 213.215.45.230 : 5060 (no NAT)
    Using INVITE request as basis request - 535a09dd7b98f0d231326e4613d941ab@ippi.fr
    Found peer 'ippi' for '331********' from 213.215.45.230:5060
    
    <--- Reliably Transmitting (NAT) to 213.215.45.230:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 213.215.45.230;branch=z9hG4bK4021.e4e3eba1.0;received=213.215.45.230
    Via: SIP/2.0/UDP 213.215.45.252:5060;received=213.215.45.252;branch=z9hG4bK4e499720;rport=5060
    From: "331********" <sip:331********@ippi.fr>;tag=as259f1013
    To: <sip:01********@ippi.fr>;tag=as26827315
    Call-ID: 535a09dd7b98f0d231326e4613d941ab@ippi.fr
    CSeq: 102 INVITE
    Server: Asterisk PBX 1.6.2.5-0ubuntu1.3
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e0ad280"
    Content-Length: 0
    
    
    <------------>
    Scheduling destruction of SIP dialog '535a09dd7b98f0d231326e4613d941ab@ippi.fr' in 6400 ms (Method: INVITE)
    Sediad*CLI> 
    <--- SIP read from UDP:213.215.45.230:5060 --->
    ACK sip:331********@***.***.***.*** SIP/2.0
    Via: SIP/2.0/UDP 213.215.45.230;branch=z9hG4bK4021.e4e3eba1.0
    From: "331********" <sip:331********@ippi.fr>;tag=as259f1013
    Call-ID: 535a09dd7b98f0d231326e4613d941ab@ippi.fr
    To: <sip:01********@ippi.fr>;tag=as26827315
    CSeq: 102 ACK
    Max-Forwards: 70
    User-Agent: OpenSIPS (1.6.3-notls (i386/linux))
    Content-Length: 0
    
    
    <------------->
    --- (9 headers 0 lines) ---

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