[AsteriskBrasil] ASterisk GUi2.0

[Marcel Morais Luna]

Essas regras foram criadas automaticamente pelo Asterisk GUI2.0
ao criar gráficamente pela ferramenta.

 == Using SIP RTP CoS mark
5                                                                                        

    -- Executing [7532214545 em DLPN_n_celular:1]
Macro("SIP/6106-092964e0",
"trunkdial-failover-0.3,SIP/trunk_1/7532214545,,trunk_1,") in new
stack                                                                                          

    -- Executing [s em macro-trunkdial-failover-0.3:1]
GotoIf("SIP/6106-092964e0", "0?1-fmsetcid,1") in new stack       
    -- Executing [s em macro-trunkdial-failover-0.3:2]
GotoIf("SIP/6106-092964e0", "0?1-setgbobname,1") in new stack    
    -- Executing [s em macro-trunkdial-failover-0.3:3]
Set("SIP/6106-092964e0", "CALLERID(num)=") in new stack          
    -- Executing [s em macro-trunkdial-failover-0.3:4]
GotoIf("SIP/6106-092964e0", "0?1-dial,1") in new stack           
    -- Executing [s em macro-trunkdial-failover-0.3:5]
Set("SIP/6106-092964e0", "CALLERID(all)=") in new stack          
    -- Executing [s em macro-trunkdial-failover-0.3:6]
Goto("SIP/6106-092964e0", "1-dial,1") in new stack               
    -- Goto
(macro-trunkdial-failover-0.3,1-dial,1)                                                                  

    -- Executing [1-dial em macro-trunkdial-failover-0.3:1]
Dial("SIP/6106-092964e0", "SIP/trunk_1/7532214545") in new
stack                                                                                                                  

  == Using SIP RTP CoS mark
5                                                                                        

[Nov 12 10:25:02] WARNING[9784]: chan_sip.c:4599 create_addr: No such
host: trunk_1                                  
[Nov 12 10:25:02] WARNING[9784]: app_dial.c:1528 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 -
Unknown)                                                                                                               

  == Everyone is busy/congested at this time
(1:0/0/1)                                                               
    -- Executing [1-dial em macro-trunkdial-failover-0.3:2]
GotoIf("SIP/6106-092964e0", "0 > 0 ?1-CHANUNAVAIL,1:1-out,1") in new
stack                                                                                                        

    -- Goto (macro-trunkdial-failover-0.3,1-out,1)
    -- Executing [1-out em macro-trunkdial-failover-0.3:1]
Hangup("SIP/6106-092964e0", "") in new stack
  == Spawn extension (macro-trunkdial-failover-0.3, 1-out, 1) exited
non-zero on 'SIP/6106-092964e0' in macro 'trunkdial-failover-0.3'
  == Spawn extension (DLPN_n_celular, 7532214545, 1) exited non-zero on
'SIP/6106-092964e0'
       > HTTP Manager add header action: waitevent
       > HTTP Manager add header advancedmode: yes
       > HTTP Manager add header mansession_id: 34b48079
       > HTTP Manager add header rwaccess: yes
       > HTTP Manager add header username: admin
       > HTTP Manager add header action: waitevent
       > HTTP Manager add header advancedmode: yes
       > HTTP Manager add header mansession_id: 34b48079
       > HTTP Manager add header rwaccess: yes
       > HTTP Manager add header username: admin
       > HTTP Manager add header action: ping
       > HTTP Manager add header advancedmode: yes
       > HTTP Manager add header mansession_id: 34b48079
       > HTTP Manager add header rwaccess: yes
       > HTTP Manager add header username: admin
       > HTTP Manager add header action: ping
       > HTTP Manager add header advancedmode: yes

-- 
*Marcel Morais Luna*
Analista de Infra-estrutura
E-mail: marcel.luna em ufba.br



Eduardo - Ustel escreveu:
> set verbose 20
> set debug level 5
>
> e faça uma chamada usando o tronco que esta tendo problema, e manda a tela 
> para analisar o que pode ser
> ----- Original Message ----- 
> From: "Marcel Morais Luna" <marcel.luna em ufba.br>
> To: <asteriskbrasil em listas.asteriskbrasil.org>
> Sent: Thursday, November 12, 2009 9:24 AM
> Subject: Re: [AsteriskBrasil] ASterisk GUi2.0
>
>
> fiz as mudanças
> coloquei somente ulaw/alaw e mesmo assim diz que "A pessoal não está
> disponivel..."
> ja até reinicie o asterisk
>
> sera que preciso fazer mais alguma outra coisa?
> no console
> eu coloco o
> show sip registry
>
> e mostra o servidor principal (trunk) com a conexao establecida normlamente
>
> 1 sip registrations
>
> se eu coloca na extensions
>
> _X., SIP/trunk é pra qualquer coisa q eu digitar ele ir correto?
>
> fiz esse teste e nada
>
> Continua dando o mesmo erro
>
>