FÓRUM DigiVoice

Tenho o mesmo problema, utilizando a operadora VOIP Transit. Tenho um Asterisk em produção com as mesmas configurações do SIP funcionando perfeitamente.

Boa tarde,

estou com o mesmo problema do colega sajunior, porém tenho um tronco E1. Quando tento fazer uma ligação o Meucci retorna uma mensagem dizendo que todas as linhas estão ocupadas, sendo que ninguém as está utilizando.

Alguém tem idéia de como resolver isso?

No aguardo.

geordano

Alguma novidade deste caso, pois o erro também ocorre aqui, ai basta fazer um reload que tudo volta ao normal.

Bom dia,

segue o log do meucci:

sip*CLI>
       Verbosidade 50 ligada em TODAS as portas.

    -- Executing Set("SIP/4520-08239a70", "INTRACOMPANYROUTE=YES") in new stack
    -- Executing Macro("SIP/4520-08239a70", "dialout-trunk|10|08007077272||") in new stack
    -- Executing Set("SIP/4520-08239a70", "DIAL_TRUNK=10") in new stack
    -- Executing Set("SIP/4520-08239a70", "DIAL_NUMBER=08007077272") in new stack
    -- Executing Set("SIP/4520-08239a70", "ROUTE_PASSWD=") in new stack
    -- Executing GotoIf("SIP/4520-08239a70", "1?noauth") in new stack
    -- Goto (macro-dialout-trunk,s,6)
    -- Executing GotoIf("SIP/4520-08239a70", "0?disabletrunk|1") in new stack
    -- Executing Set("SIP/4520-08239a70", "_NODEST=") in new stack
    -- Executing Set("SIP/4520-08239a70", "DIAL_TRUNK_OPTIONS=twW") in new stack
    -- Executing Set("SIP/4520-08239a70", "GROUP()=OUT_10") in new stack
    -- Executing Macro("SIP/4520-08239a70", "user-callerid|SKIPTTL") in new stack
    -- Executing NoOp("SIP/4520-08239a70", "user-callerid: device 4520") in new stack
    -- Executing Set("SIP/4520-08239a70", "AMPUSER=4520") in new stack
    -- Executing GotoIf("SIP/4520-08239a70", "0?report") in new stack
    -- Executing GotoIf("SIP/4520-08239a70", "0?start") in new stack
    -- Executing Set("SIP/4520-08239a70", "REALCALLERIDNUM=4520") in new stack
    -- Executing NoOp("SIP/4520-08239a70", "REALCALLERIDNUM is 4520") in new stack
    -- Executing Set("SIP/4520-08239a70", "AMPUSER=4520") in new stack
    -- Executing Set("SIP/4520-08239a70", "AMPUSERCIDNAME=Katia Freiberger") in new stack
    -- Executing GotoIf("SIP/4520-08239a70", "0?report") in new stack
    -- Executing Set("SIP/4520-08239a70", "AMPUSERCID=4520") in new stack
    -- Executing Set("SIP/4520-08239a70", "CALLERID(all)="Katia Freiberger" <4520>") in new stack
    -- Executing Set("SIP/4520-08239a70", "REALCALLERIDNUM=4520") in new stack
    -- Executing NoOp("SIP/4520-08239a70", "TTL:  ARG1: SKIPTTL") in new stack
    -- Executing GotoIf("SIP/4520-08239a70", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,23)
    -- Executing NoOp("SIP/4520-08239a70", "Using CallerID "Katia Freiberger" <4520>") in new stack
    -- Executing Macro("SIP/4520-08239a70", "record-enable|4520|OUT") in new stack
    -- Executing GotoIf("SIP/4520-08239a70", "0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing AGI("SIP/4520-08239a70", "recordingcheck|20101025-095733|1288007853.342") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20101025-095733|1288007853.342: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/4520-08239a70", "No recording needed") in new stack
    -- Executing GotoIf("SIP/4520-08239a70", "1?skipoutcid") in new stack
    -- Goto (macro-dialout-trunk,s,15)
    -- Executing GotoIf("SIP/4520-08239a70", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,17)
    -- Executing AGI("SIP/4520-08239a70", "fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing Set("SIP/4520-08239a70", "OUTNUM=08007077272") in new stack
    -- Executing Set("SIP/4520-08239a70", "custom=DGV/G1") in new stack
    -- Executing GotoIf("SIP/4520-08239a70", "1?gocall") in new stack
    -- Goto (macro-dialout-trunk,s,22)
    -- Executing GotoIf("SIP/4520-08239a70", "0?customtrunk") in new stack
    -- Executing Dial("SIP/4520-08239a70", "DGV/G1/08007077272|300|twW") in new stack
Oct 25 09:57:33 NOTICE[12272]: chan_dgv.c:14745 dgv_request: Asked to get a channel DGV/G1/08007077272 using format AST_FORMAT_ULAW:ffWaveULaw
Oct 25 09:57:33 NOTICE[12272]: chan_dgv.c:14761 dgv_request: Group G1
Oct 25 09:57:33 WARNING[12272]: chan_dgv.c:14915 dgv_request: dgv_request: Port 1 in use or reserved!
Oct 25 09:57:33 NOTICE[12272]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'DGV' (cause 34 - Circuit/channel congestion)
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing Goto("SIP/4520-08239a70", "s-CONGESTION|1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing GotoIf("SIP/4520-08239a70", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,3)
    -- Executing NoOp("SIP/4520-08239a70", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
    -- Executing Macro("SIP/4520-08239a70", "outisbusy|") in new stack
    -- Executing Playback("SIP/4520-08239a70", "all-circuits-busy-now|noanswer") in new stack
    -- Playing 'all-circuits-busy-now' (language 'pt_BR')
sip*CLI>



Parece que ele não está conseguindo puxar o outro tronco para fazer as ligações.

O que posso fazer para solucionar esse problema que os outros colegas também estão enfrentando?

Atenciosamente,
Geordano Arend