Olá Rafael, tudo bem?<br><div>Tente isso no zapata.conf<br><br>[trunkgroups]<br><br>[channels]<br>context=seu-contexto<br>switchtype=euroisdn<br>signalling=pri_cpe<br>;rxwink=300</div><div><span class="q"><br>usecallerid=yes
<br>hidecallerid=no<br>callwaiting=yes
<br>usecallingpres=yes<br></span></div><div>restrictcid=no</div><div><span class="q"><br>callwaitingcallerid=yes<br>threewaycalling=yes<br>transfer=yes<br>cancallforward=yes<br>callreturn=yes<br>echocancel=yes<br></span>
</div><div>echocancelwhenbridged=yes</div><div><span class="q"><br>rxgain=0.0<br>txgain=0.0
<br></span></div><div>group=1<br>callgroup=1<br>immediate=no<br>callerid=asreceived<br>musiconhold=sua-musica-em espera<br>group=1</div><div><span class="q"><br>channel=>1-15<br>channel=>17-31<br><br><br></span></div>
Abraço e boa sorte<br><br>Josué<br><br><div><span class="gmail_quote">2006/9/13, Rafael Augusto <<a href="mailto:rafael.augusto@govoip.com.br">rafael.augusto@govoip.com.br</a>>:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Dio, segue o erro ao efetuar o dial.<br><br><br>Executing Macro("SIP/200-0a0fbac0", "dialout-trunk|2|100|") in new stack<br> -- Executing GotoIf("SIP/200-0a0fbac0", "1?3:2)") in new stack
<br> -- Goto (macro-dialout-trunk,s,3)<br> -- Executing Macro("SIP/200-0a0fbac0", "user-callerid") in new stack<br> -- Executing DBget("SIP/200-0a0fbac0", "AMPUSER=DEVICE/200/user") in new
<br>stack<br> -- DBget: varname=AMPUSER, family=DEVICE, key=200/user<br> -- DBget: set variable AMPUSER to 200<br> -- Executing DBget("SIP/200-0a0fbac0",<br>"AMPUSERCIDNAME=AMPUSER/200/cidname") in new stack
<br> -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=200/cidname<br> -- DBget: set variable AMPUSERCIDNAME to 200<br> -- Executing GotoIf("SIP/200-0a0fbac0", "0?5") in new stack<br> -- Executing SetCallerID("SIP/200-0a0fbac0", ""200" <200>") in new stack
<br> -- Executing NoOp("SIP/200-0a0fbac0", "Using CallerID "200" <200>") in<br>new stack<br> -- Executing Macro("SIP/200-0a0fbac0", "record-enable|200|OUT") in new
<br>stack<br> -- Executing GotoIf("SIP/200-0a0fbac0", "0 > 0?2:4") in new stack<br> -- Goto (macro-record-enable,s,4)<br> -- Executing AGI("SIP/200-0a0fbac0",<br>"recordingcheck|20060913-143611|1158172571.7") in new stack
<br> -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck<br> recordingcheck|20060913-143611|1158172571.7: Outbound recording not<br>enabled<br> -- AGI Script recordingcheck completed, returning 0<br> -- Executing NoOp("SIP/200-0a0fbac0", "No recording needed") in new
<br>stack<br> -- Executing Macro("SIP/200-0a0fbac0", "outbound-callerid|2") in new<br>stack<br> -- Executing DBget("SIP/200-0a0fbac0",<br>"USEROUTCID=AMPUSER/200/outboundcid") in new stack
<br> -- DBget: varname=USEROUTCID, family=AMPUSER, key=200/outboundcid<br> -- DBget: set variable USEROUTCID to 200<br> -- Executing GotoIf("SIP/200-0a0fbac0", "0?4") in new stack<br> -- Executing SetCallerID("SIP/200-0a0fbac0", "Rota PABX") in new stack
<br> -- Executing GotoIf("SIP/200-0a0fbac0", "0?6") in new stack<br> -- Executing SetCallerID("SIP/200-0a0fbac0", "200") in new stack<br> -- Executing NoOp("SIP/200-0a0fbac0", "CallerID set to 200") in new
<br>stack<br> -- Executing SetGroup("SIP/200-0a0fbac0", "OUT_2") in new stack<br> -- Executing CheckGroup("SIP/200-0a0fbac0", "30") in new stack<br> -- Executing SetVar("SIP/200-0a0fbac0", "DIAL_NUMBER=100") in new stack
<br> -- Executing SetVar("SIP/200-0a0fbac0", "DIAL_TRUNK=2") in new stack<br> -- Executing AGI("SIP/200-0a0fbac0", "fixlocalprefix") in new stack<br> -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
<br> -- AGI Script fixlocalprefix completed, returning 0<br> -- Executing SetVar("SIP/200-0a0fbac0", "OUTNUM=100") in new stack<br> -- Executing Cut("SIP/200-0a0fbac0", "custom=OUT_2|:|1") in new stack
<br> -- Executing GotoIf("SIP/200-0a0fbac0", "0?16") in new stack<br> -- Executing Dial("SIP/200-0a0fbac0", "ZAP/g1/100") in new stack<br> == Everyone is busy/congested at this time (1:0/1/0)
<br> -- Executing Goto("SIP/200-0a0fbac0", "s-CONGESTION|1") in new stack<br> -- Goto (macro-dialout-trunk,s-CONGESTION,1)<br> -- Executing NoOp("SIP/200-0a0fbac0", "Dial failed due to CONGESTION")
<br>in new stack<br> -- Executing Macro("SIP/200-0a0fbac0", "outisbusy") in new stack<br> -- Executing PlayTones("SIP/200-0a0fbac0", "Busy") in new stack<br> -- Executing Macro("SIP/200-0a0fbac0", "hangupcall") in new stack
<br> -- Executing ResetCDR("SIP/200-0a0fbac0", "w") in new stack<br> -- Executing NoCDR("SIP/200-0a0fbac0", "") in new stack<br> -- Executing Wait("SIP/200-0a0fbac0", "5") in new stack
<br> -- Executing Hangup("SIP/200-0a0fbac0", "") in new stack<br><br>Abraços,<br><br>Rafael<br><br><br><br><br><br><br><br>Message: 2<br>Date: Wed, 13 Sep 2006 11:35:07 -0300 (ART)<br>From: Dio Makibara <
<a href="mailto:dioedu@yahoo.com.br">dioedu@yahoo.com.br</a>><br>Subject: Re: [AsteriskBrasil] Config. E1 ISDN com Siemens Hicom 150<br>To: Rafael Augusto <<a href="mailto:rafael_jcn@yahoo.com.br">rafael_jcn@yahoo.com.br
</a>><br>Cc: Asterisk Brasil <<a href="mailto:asteriskbrasil@listas.asteriskbrasil.org">asteriskbrasil@listas.asteriskbrasil.org</a>><br>Message-ID: <<a href="mailto:20060913143507.3048.qmail@web52504.mail.yahoo.com">
20060913143507.3048.qmail@web52504.mail.yahoo.com</a>><br>Content-Type: text/plain; charset="iso-8859-1"<br><br>Rafael,<br><br> Envie as mensagens para lista, pois a chance de ser respondida é maior.<br><br> Mas aparentemente as configurações estão corretas. Informe o que está sendo
<br>exibido no console do asterisk ao tentar efetuar uma ligação.<br><br> Diógenes Makibara<br><br><br>Rafael Augusto <<a href="mailto:rafael_jcn@yahoo.com.br">rafael_jcn@yahoo.com.br</a>> escreveu: Dio, segue configuração
<br>do zaptel e zapata, o extensions é tranquilo, se poder me ajudar, desde de<br>já agradeço.<br><br> zaptel.conf<br> span=1,1,0,ccs,hdb3,crc4<br>bchan=1-15<br>dchan=16<br>bchan=17-31<br><br> loadzone = br<br>defaultzone = br
<br><br> zapata.conf<br><br> [channels]<br> language=br<br>context=from-pstn<br>signalling=fxs_ks<br>rxwink=300 ; Atlas seems to use long (250ms) winks<br>;<br>; Whether or not to do distinctive ring detection on FXO lines
<br>;<br>usecallerid=yes<br>hidecallerid=no<br>callwaiting=yes<br>usecallingpres=yes<br>callwaitingcallerid=yes<br>threewaycalling=yes<br>transfer=yes<br>cancallforward=yes<br>callreturn=yes<br>echocancel=yes<br>echocancelwhenbridged=no
<br>echotraining=800<br>rxgain=0.0<br>txgain=0.0<br>group=0<br>callgroup=1<br>pickupgroup=1<br>group=1<br><br> group = 1<br>context =ext-ddr<br>signalling=pri_net<br>overlapdial=yes<br>immediate=no<br>channel => 1-15<br>
channel => 17-31<br><br><br> Connected to Asterisk 1.2.10 currently running on govoip (pid = 19866)<br>Verbosity is at least 3<br>govoip*CLI> pri show span 1<br>Primary D-channel: 16<br>Status: Provisioned, In Alarm, Down, Active
<br>Switchtype: National ISDN<br>Type: Network<br>Window Length: 0/7<br>Sentrej: 0<br>SolicitFbit: 0<br>Retrans: 0<br>Busy: 0<br>Overlap Dial: -1<br>T200 Timer: 1000<br>T203 Timer: 10000<br>T305 Timer: 30000<br>T308 Timer: 4000
<br>T313 Timer: 4000<br>N200 Counter: 3<br><br> Agraços,<br><br> Rafael<br><br><br><br>----------------------------------------<br>Estação VoIP 2006<br>5 e 6 Dezembro<br>Curitiba PR<br><a href="http://www.estacaovoip.com.br">
http://www.estacaovoip.com.br</a><br><br>_______________________________________________<br>LIsta de discussões AsteriskBrasil.org<br><a href="mailto:AsteriskBrasil@listas.asteriskbrasil.org">AsteriskBrasil@listas.asteriskbrasil.org
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AsteriskBrasil.org:<br><a href="http://www.asteriskbrasil.org">http://www.asteriskbrasil.org</a><br></blockquote></div><br>