Re: [AsteriskBrasil] Sip´s não falam entre si no asterisk@home

Jose P. Leitao jose.leitao em oi.com.br
Sexta Dezembro 23 16:12:49 BRT 2005


Sugiro verificarem quais os codecs estão permitidos para os respectivos ramais e se os os mesmos estão habilitados nos clientes, ative o debug do sip na interface CLI, que ele irá informar o motivo da não conexão.

SDS
José Leitão
  ----- Original Message ----- 
  From: Frederico Simões 
  To: asteriskbrasil em listas.asteriskbrasil.org 
  Sent: Friday, December 23, 2005 1:54 PM
  Subject: Re: [AsteriskBrasil] Sip´s não falam entre si no asterisk em home


  Estou com um problema parecido...

  O meu fala de X-lite para X-lite... so que não fala de PAP para PAP!!!

  Se alguem poder solucionar 

  [ext-local]
  include => ext-local-custom
  exten => 10001,1,Macro(exten-vm,novm,10001)
  exten => 10002,1,Macro(exten-vm,novm,10002)
  exten => 1001,1,Macro(exten-vm,1001 em default,1001)
  exten => ${VM_PREFIX}1001,1,Macro(vm,1001)
  exten => 1002,1,Macro(exten-vm,1002 em default,1002)
  exten => ${VM_PREFIX}1002,1,Macro(vm,1002)
  exten => 10021,1,Macro(exten-vm,10021 em default,10021)
  exten => ${VM_PREFIX}10021,1,Macro(vm,10021)
  exten => 10022,1,Macro(exten-vm,10022 em default,10022)
  exten => ${VM_PREFIX}10022,1,Macro(vm,10022)
  exten => 1003,1,Macro(exten-vm,1003 em default,1003)
  exten => ${VM_PREFIX}1003,1,Macro(vm,1003)
  exten => 1004,1,Macro(exten-vm,1004 em default,1004)
  exten => ${VM_PREFIX}1004,1,Macro(vm,1004)
  exten => 1005,1,Macro(exten-vm,1005 em default,1005)
  exten => ${VM_PREFIX}1005,1,Macro(vm,1005)
  exten => 11001,1,Macro(exten-vm,novm,11001)
  exten => 11002,1,Macro(exten-vm,novm,11002)
  exten => 12001,1,Macro(exten-vm,novm,12001)
  exten => 3001,1,Macro(exten-vm,novm,3001)
  exten => 3002,1,Macro(exten-vm,novm,3002)
  exten => 4001,1,Macro(exten-vm,novm,4001)
  exten => 4002,1,Macro(exten-vm,novm,4002)
  exten => 4011,1,Macro(exten-vm,novm,4011)
  exten => 4012,1,Macro(exten-vm,novm,4012)
  exten => 4021,1,Macro(exten-vm,novm,4021)



  [macro-exten-vm]
  exten => s,1,Macro(user-callerid)
  exten => s,2,Setvar(FROMCONTEXT=exten-vm)
  exten => s,3,Macro(record-enable,${ARG2},IN)
  exten => s,4,Macro(dial,${RINGTIMER},${DIAL_OPTIONS},${ARG2})
  exten => s,5,GotoIf($[${CHANNEL:0:5} = Local]?s-${DIALSTATUS},1) ; if the channel is Local, then do not go to voicemail.  This is primarily to avoid vm for call-forwarded extensions in ring groups
  exten => s,6,GotoIf($[${ARG1} = novm]?s-${DIALSTATUS},1) ; no voicemail in use for this extension
  exten => s,7,NoOp(Sending to Voicemail box ${ARG2})
  exten => s,8,Macro(vm,${ARG1},${DIALSTATUS})
  exten => s-BUSY,1,NoOp(Extension is reporting BUSY and has no Voicemail)
  exten => s-BUSY,2,Busy()
  exten => s-BUSY,3,Wait(60)
  exten => s-BUSY,4,NoOp()
  exten => _s-.,1,Congestion()

    Bom dia Asteriskers;

    Sempre trabalhei com o Asterisk instalado ´no braço´. Porém resolvi testar o asterisk em home e estou com um probleminha básico.

    Configurei dois ramais sip com o x-lite. Eles estão logando e o ecotest está ok.
    Porém quando tento falar de um sip (1234) para outro (2000) aparece a menságem no x-lite:( Call failed 486 Busy Here ).

    O ast em home está em uma rede interna juntamente com os outros softphones, exluindo-se a possibilidade de ser uma dificuldade em passar pelo NAT.

    Existe algo a mais que tem que ser feito, além de se configurar os ramais no ast em home para funcionar? 

    Abaixo o log do que acontece no momento que ligo para outro sip:

    Desde já agradeço a atenção de todos.

    Willians Dias
    Vitória E.S.


    login as: root
    root em 192.168.30.125's password:
    Last login: Thu Jan  6 19:27:39 2005 from 192.168.30.161

    Welcome to Asterisk em Home
    -------------------------------------------------

    For access to the Asterisk em Home web GUI use this URL
    http://192.168.30.125

    For help on Asterisk em Home commands you can use from this
    command shell type help-aah.

    [root em asterisk1 ~]# asterisk -r
    Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium.
    Written by Mark Spencer <markster em digium.com>
    =========================================================================
    Connected to Asterisk 1.2.1 currently running on asterisk1 (pid = 2381)
    Verbosity is at least 3
        -- Executing Macro("SIP/2000-7e61", "exten-vm|novm|1234") in new stack
        -- Executing Macro("SIP/2000-7e61", "user-callerid") in new stack
        -- Executing DBget("SIP/2000-7e61", "AMPUSER=DEVICE/2000/user") in new stack
        -- DBget: varname=AMPUSER, family=DEVICE, key=2000/user
        -- DBget: set variable AMPUSER to 2000
        -- Executing DBget("SIP/2000-7e61", "AMPUSERCIDNAME=AMPUSER/2000/cidname") in new stack
        -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=2000/cidname
        -- DBget: set variable AMPUSERCIDNAME to Willians
        -- Executing GotoIf("SIP/2000-7e61", "0?5") in new stack
        -- Executing SetCallerID("SIP/2000-7e61", ""Willians" <2000>") in new stack
        -- Executing NoOp("SIP/2000-7e61", "Using CallerID "Willians" <2000>") in new stack
        -- Executing SetVar("SIP/2000-7e61", "FROMCONTEXT=exten-vm") in new stack
        -- Executing Macro("SIP/2000-7e61", "record-enable|1234|IN") in new stack
        -- Executing GotoIf("SIP/2000-7e61", "0 > 0?2:4") in new stack
        -- Goto (macro-record-enable,s,4)
        -- Executing AGI("SIP/2000-7e61", "recordingcheck|20050106-193009|1105057809.8") in new stack
        -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
      recordingcheck|20050106-193009|1105057809.8: Inbound recording not enabled
        -- AGI Script recordingcheck completed, returning 0
        -- Executing NoOp("SIP/2000-7e61", "No recording needed") in new stack
        -- Executing Macro("SIP/2000-7e61", "dial|49|tr|1234") in new stack
        -- Executing GotoIf("SIP/2000-7e61", "0?4:2") in new stack
        -- Goto (macro-dial,s,2)
        -- Executing GotoIf("SIP/2000-7e61", "0?5:4") in new stack
        -- Goto (macro-dial,s,4)
        -- Executing AGI("SIP/2000-7e61", "dialparties.agi") in new stack
        -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
        -- AGI Script dialparties.agi completed, returning 0
        -- Executing NoOp("SIP/2000-7e61", "Returned from dialparties with no extensions to call") in new stack
        -- Executing SetVar("SIP/2000-7e61", "DIALSTATUS=BUSY") in new stack
        -- Executing GotoIf("SIP/2000-7e61", "0?s-BUSY|1") in new stack
        -- Executing GotoIf("SIP/2000-7e61", "1?s-BUSY|1") in new stack
        -- Goto (macro-exten-vm,s-BUSY,1)
        -- Executing NoOp("SIP/2000-7e61", "Extension is reporting BUSY and has no Voicemail") in new stack
        -- Executing Busy("SIP/2000-7e61", "") in new stack
      == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'SIP/2000-7e61' in macro 'exten-vm'
      == Spawn extension (from-internal, 1234, 1) exited non-zero on 'SIP/2000-7e61'
        -- Executing Macro("SIP/2000-7e61", "hangupcall") in new stack
        -- Executing ResetCDR("SIP/2000-7e61", "w") in new stack
        -- Executing NoCDR("SIP/2000-7e61", "") in new stack
        -- Executing Wait("SIP/2000-7e61", "5") in new stack
      == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/2000-7e61' in macro 'hangupcall'
      == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/2000-7e61'
    asterisk1*CLI>



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