Problemas con registro de extensiones , iax y sip

Discussion in 'Elastix 2.x' started by wysiwyg, Mar 14, 2011.

  1. wysiwyg

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    Buenas,
    He actualizado las ultimas versiones a FreePBX 2.9.0beta2.1 y resulta que ahora las extensiones que tenia creadas funcionan pero si creo una nueva , bien sea iax o sip, no hay manera de que se registren. Supongo que algun parametro se me esta pasando por alto y no hay manera...

    HP ProLiant DL120 G5 Xeon E3110 3 GHz
    1 x Intel Xeon E3110 / 3 GHz (DualCore)
    1 GB (instalados) / 8 GB (máx.) DDR2 SDRAM ECC 800 MHz PC2-6400
    1 x 250 GB estándar Serial ATA150
    DVD RW

    Distribución Linux: Redhat CentOS release 5.5 (Final)
    Version Asterisk: Asterisk 1.6.2.13, Copyright (C) 1999 - 2010 Digium, Inc. and others.
    Versión Elastix: 2.0.1 (64 bits)
    Freepbx Versión: 2.8.0.4 (ultima estable que funcionaba todo)

    Quisiera que me aclarasen si existe algún problema con la nueva beta o que es lo que pudiese estar fallando....

    dolar*CLI> sip show peers
    Name/username Host Dyn Nat ACL Port Status
    2999/2999 192.168.0.25 D N A 5061 OK (29 ms)
    3001/3001 192.168.0.25 D N A 5060 OK (26 ms)
    3002/3002 192.168.0.26 D N A 5060 OK (309 ms)
    3003/3003 192.168.0.7 D N A 10011 OK (31 ms)
    3004/3004 192.168.0.7 D N A 5061 OK (35 ms)
    3005/3005 192.168.0.24 D N A 5061 OK (18 ms)
    3007/3007 192.168.0.20 D N A 1025 OK (24 ms)
    3008/3008 87.223.235.165 D N 5060 OK (223 ms)
    3009/3009 83.56.151.7 D N 5060 OK (161 ms)
    3010/3010 192.168.0.7 D N A 10005 OK (29 ms)
    3011/3011 192.168.0.7 D N A 10012 OK (35 ms)
    3012/3012 192.168.0.7 D N A 10013 OK (26 ms)
    3013/3013 192.168.0.7 D N A 10002 OK (38 ms)
    3014/3014 192.168.0.20 D N A 1024 OK (38 ms)
    3015/3015 192.168.0.7 D N A 10760 OK (17 ms)
    3016/3016 192.168.0.24 D N A 5060 OK (18 ms)
    4000 (Unspecified) D N A 5060 UNKNOWN


    Parametros Globales sip settings


    Global Settings:
    ----------------
    UDP SIP Port: 5060
    UDP Bindaddress: 0.0.0.0
    TCP SIP Port: Disabled
    TLS SIP Port: Disabled
    Videosupport: No
    Textsupport: No
    Ignore SDP sess. ver.: No
    AutoCreate Peer: No
    Match Auth Username: No
    Allow unknown access: No
    Allow subscriptions: Yes
    Allow overlap dialing: Yes
    Allow promsic. redir: No
    Enable call counters: No
    SIP domain support: Yes
    Realm. auth: No
    Our auth realm asterisk
    Call to non-local dom.: No
    URI user is phone no: No
    Always auth rejects: Yes
    Direct RTP setup: No
    User Agent: FPBX-2.9.0beta2(1.6.2.13)
    SDP Session Name: Asterisk PBX 1.6.2.13
    SDP Owner Name: root
    Reg. context: (not set)
    Regexten on Qualify: No
    Caller ID: Unknown
    From: Domain:
    Record SIP history: Off
    Call Events: Off
    Auth. Failure Events: Off
    T.38 support: No
    T.38 EC mode: Unknown
    T.38 MaxDtgrm: -1
    SIP realtime: Disabled
    Qualify Freq : 60000 ms

    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
    Jitterbuffer forced: No
    Jitterbuffer max size: -1
    Jitterbuffer resync: -1
    Jitterbuffer impl:
    Jitterbuffer log: No

    Network Settings:
    ---------------------------
    SIP address remapping: Enabled using externhost
    Externhost: 87.111.66.50
    Externip: 87.111.66.50:5060
    Externrefresh: 10
    Internal IP: 127.0.0.1:5060
    Localnet: 192.168.0.0/255.255.255.0
    STUN server: 0.0.0.0:0

    Global Signalling Settings:
    ---------------------------
    Codecs: 0x10e (gsm|ulaw|alaw|g729)
    Codec Order: ulaw:20,gsm:20,alaw:20,g729:20
    Relax DTMF: No
    RFC2833 Compensation: No
    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: Yes
    Pedantic SIP support: No
    Reg. min duration 60 secs
    Reg. max duration: 3600 secs
    Reg. default duration: 120 secs
    Outbound reg. timeout: 20 secs
    Outbound reg. attempts: 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: <not set>
    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

    Default Settings:
    -----------------
    Allowed transports: UDP
    Outbound transport: UDP
    Context: from-sip-external
    Nat: Always
    DTMF: rfc2833
    Qualify: 0
    Use ClientCode: No
    Progress inband: Never
    Language: es
    MOH Interpret: default
    MOH Suggest:
    Voice Mail Extension: *97
    Forward Detected Loops: Yes



    Parametros de la extension 4000 que no conecta...


    dolar*CLI> sip show peer 4000


    * Name : 4000
    Secret : <Set>
    MD5Secret : <Not set>
    Remote Secret: <Not set>
    Context : from-internal
    Subscr.Cont. : <Not set>
    Language : es
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Mailbox : 4000@device
    VM Extension : *97
    LastMsgsSent : 32767/65535
    Call limit : 2147483647
    Dynamic : Yes
    Callerid : "device" <4000>
    MaxCallBR : 384 kbps
    Expire : -1
    Insecure : no
    Nat : Always
    ACL : Yes
    T.38 support : No
    T.38 EC mode : Unknown
    T.38 MaxDtgrm: -1
    DirectMedia : No
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Text Support : No
    Ign SDP ver : No
    Trust RPID : Yes
    Send RPID : No
    Subscriptions: Yes
    Overlap dial : Yes
    Forward Loop : Yes
    DTMFmode : rfc2833
    Timer T1 : 500
    Timer B : 32000
    ToHost :
    Addr->IP : (Unspecified) Port 5060
    Defaddr->IP : 0.0.0.0 Port 5060
    Prim.Transp. : UDP
    Allowed.Trsp : UDP
    Def. Username:
    SIP Options : (none)
    Codecs : 0x10e (gsm|ulaw|alaw|g729)
    Codec Order : (ulaw:20,gsm:20,alaw:20,g729:20)
    Auto-Framing : No
    100 on REG : No
    Status : UNKNOWN
    Useragent :
    Reg. Contact :
    Qualify Freq : 60000 ms
    Sess-Timers : Accept
    Sess-Refresh : uas
    Sess-Expires : 1800 secs
    Min-Sess : 90 secs
    Parkinglot :


    Tenia una extension 3000 anterior en iax y se ha conectado perfectamente antes de actualizar el sistema.... no entiendo....




    Gracias de antemano...
     
  2. asepulveda

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    Espero que no lo hayas hecho en un servidor en producción, es una version Beta lo que sucede pueden ser miles de cosas, a mi parecer es algo de permisos
     
  3. wysiwyg

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    Lamentablemente, si asi es, es un servidor en produccion y he sido tan iluso de actualizar la version beta.....

    Hoy mismo me ha llegado otra version de la beta, supongo que ahora empezarán a enviar versiones hasta que esté solucionado.....

    Gracias por tu interes.
    Un cordial saludo.
    jose.
     
  4. asepulveda

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    Entonces supongo que si te dieran una droga experimental para curar la gripa la tomarias no?

    De por si uno nunca debe actualizar un sistema en producción aun y cuando los releases sean estables sin antes probarlo en un conmutador de testing, un BETA jamas se pone en uno en producción, error grave
     
  5. fmvillares

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    de todas formas algo te salio mal porqu8e yo uso la beta ya que soy tester de freepbx y anda de pelos pisando la freepbx 2.8 de elastix no rompe nada ams que la embedded...todo anda bien y con asterisk 1.8.y 1.6
     

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