Gracias por la ayuda de todos el problema ya lo solucione viendo el error un poco vi que estaba esto "Seize Timeout" y llame a la operadora porque recorde que una vez moises en esta lista dijo que cuando salia eso es porque la operadora no estaba respondiendo le dije al operador eso y cambio una variable en la central de ellos y funciono perfecto.<br>
<br><div class="gmail_quote">2012/11/12 alberto topp <span dir="ltr"><<a href="mailto:alberto_topp@yahoo.com.ar" target="_blank">alberto_topp@yahoo.com.ar</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<table border="0" cellpadding="0" cellspacing="0"><tbody><tr><td style="font:inherit" valign="top">Fijate los codecs, si es un E1 deberia ser codec G.711 ley-A.<br><br>En que codec esta recibiendo el mensaje SIP: INVITE en G.711 ley u como 1ra. prioridad?, En caso afirmativo cambiar en el telefono para G.711 ley A.<br>
<br><br><br>--- El <b>lun 12-nov-12, GEORGE H <i><<a href="mailto:hgeorge123@gmail.com" target="_blank">hgeorge123@gmail.com</a>></i></b> escribió:<br><blockquote style="border-left:2px solid rgb(16,16,255);margin-left:5px;padding-left:5px">
<br>De: GEORGE H <<a href="mailto:hgeorge123@gmail.com" target="_blank">hgeorge123@gmail.com</a>><br>Asunto: Re: [asterisk-r2] dtmf r2 venezuela<br>Para: "Jose Daniel Yribarren" <<a href="mailto:jdyribarren@compusan.com.ve" target="_blank">jdyribarren@compusan.com.ve</a>><br>
Cc: <a href="mailto:asterisk-r2@lists.digium.com" target="_blank">asterisk-r2@lists.digium.com</a><br>Fecha: lunes, 12 de noviembre de 2012, 17:10<div><div class="h5"><br><br><div>Haciendo prueba todavia no puedo sacar llamadas este es el log de la llamada en dtmfr2<br>
<br>[Nov 12 15:38:12] VERBOSE[3584] app_dial.c: -- Called
DAHDI/g12/04145859332<br>[Nov 12 15:38:12] DEBUG[3584] channel.c: Set channel DAHDI/1-1 to read format ulaw<br>
[Nov 12 15:38:12] DEBUG[3584] channel.c: Set channel SIP/6430-00000000 to read format alaw<br>[Nov 12 15:38:20] WARNING[3584] chan_dahdi.c: Chan 1 - Seize Timeout Expired!<br>[Nov 12 15:38:20] ERROR[3584] chan_dahdi.c: Chan 1 - Protocol error. Reason = Seize Timeout, R2 State = Seize Transmitted, MF state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08<br>
DNIS = <a href="tel:04145859332" value="+584145859332" target="_blank">04145859332</a>, ANI = 6430, MF = 0x20<br>[Nov 12 15:38:20] ERROR[3584] chan_dahdi.c: MFC/R2 protocol error on chan 1: Seize Timeout<br>[Nov 12 15:38:20] DEBUG[3584] channel.c: Hanging up channel 'DAHDI/1-1'<br>
[Nov 12 15:38:20] DEBUG[3584] chan_dahdi.c: dahdi_hangup(DAHDI/1-1)<br>[Nov 12 15:38:20] DEBUG[3584] chan_dahdi.c: Hangup: channel: 1 index = 0, normal = 13, callwait = -1, thirdcall = -1<br>[Nov 12 15:38:20] DEBUG[3584] chan_dahdi.c: Set option TDD MODE, value: OFF(0) on DAHDI/1-1<br>
[Nov 12 15:38:20] DEBUG[3584] chan_dahdi.c: Updated conferencing on 1, with 0 conference users<br>[Nov 12 15:38:20] VERBOSE[3584] chan_dahdi.c: -- Hungup 'DAHDI/1-1'<br>[Nov 12 15:38:20] VERBOSE[3584] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)<br>
[Nov 12 15:38:20] DEBUG[3584] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.<br>[Nov 12 15:38:20] DEBUG[3584] app_macro.c: Executed application: Dial<br>[Nov 12 15:38:20] DEBUG[3584] pbx.c: Result of 'DIALSTATUS' is 'CHANUNAVAIL'<br>
[Nov 12 15:38:20] DEBUG[3584] pbx.c: Result of 'HANGUPCAUSE' is '111'<br>[Nov 12 15:38:20] DEBUG[3584] pbx.c: Launching 'NoOp'<br>[Nov 12 15:38:20] VERBOSE[3584] pbx.c: -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/6430-00000000", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 111") in new stack<br>
[Nov 12 15:38:20] DEBUG[3584] app_macro.c: Executed application: Noop<br>[Nov 12 15:38:20] DEBUG[3584] pbx.c: Result of 'DIALSTATUS' is 'CHANUNAVAIL'<br>[Nov 12 15:38:20] DEBUG[3584] pbx.c: Launching 'Goto'<br>
[Nov 12 15:38:20] VERBOSE[3584] pbx.c: -- Executing [s@macro-dialout-trunk:21] Goto("SIP/6430-00000000", "s-CHANUNAVAIL,1") in new stack<br>[Nov 12 15:38:20] VERBOSE[3584] pbx.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)<br>
[Nov 12 15:38:20] DEBUG[3584] app_macro.c: Executed application: Goto<br>[Nov 12 15:38:20] DEBUG[3584] pbx.c: Result of 'HANGUPCAUSE' is '111'<br>[Nov 12 15:38:20] DEBUG[3584] pbx.c: Function result is '0'<br>
[Nov 12 15:38:20] DEBUG[3584] pbx.c: Expression result is '0'<br>[Nov 12 15:38:20] DEBUG[3584] pbx.c: Result of 'HANGUPCAUSE' is '111'<br>[Nov 12 15:38:20] DEBUG[3584] pbx.c: Function result is '111'<br>
[Nov 12 15:38:20] DEBUG[3584] pbx.c: Launching 'Set'<br><br><br><br><div>El 9 de noviembre de 2012 10:44, Jose Daniel Yribarren <span dir="ltr"><<a rel="nofollow" href="http://mc/compose?to=jdyribarren@compusan.com.ve" target="_blank">jdyribarren@compusan.com.ve</a>></span> escribió:<br>
<blockquote style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">has una prueba saca la llamada por el canal que sepas que es saliente por ejemplo el canal 33, create una troncal y en <table>
<tbody><tr><td colspan="2"><h4 style="color:gray">Outgoing Settings</h4></td></tr><tr><td><a rel="nofollow" style="border-bottom-style:dashed;border-bottom-width:1px;font-size:16px;text-decoration:none;font-family:verdana,arial,helvetica,sans-serif;border-bottom-color:rgb(204,204,204)">Zap Identifier (trunk name)</a> colocas 33<br>
<br><br>y luego saca la llamada por ese canal.. y verifica si sale</td></tr></tbody></table><br><div>2012/11/9 <span dir="ltr"><<a rel="nofollow" href="http://mc/compose?to=hgeorge123@gmail.com" target="_blank">hgeorge123@gmail.com</a>></span><div>
<div><br>
<blockquote style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Ok voy a probar poner primero las salientes y luego las entrantes con respecto a cuales son salientes y cuales entrantes no es problema porque un e1 completo es saliente y el otro entrante y las entrantes funcionan perfecto<br>
<div>George Hernandez<br>
Telf : <a rel="nofollow">(58) 4145859332</a><br>
<br>
-----Original Message-----<br>
</div><div>From: Jose Daniel Yribarren <<a rel="nofollow" href="http://mc/compose?to=jdyribarren@compusan.com.ve" target="_blank">jdyribarren@compusan.com.ve</a>><br>
Sender: <a rel="nofollow" href="http://mc/compose?to=asterisk-r2-bounces@lists.digium.com" target="_blank">asterisk-r2-bounces@lists.digium.com</a><br>
</div><div><div>Date: Fri, 9 Nov 2012 10:09:14<br>
To: <<a rel="nofollow" href="http://mc/compose?to=asterisk-r2@lists.digium.com" target="_blank">asterisk-r2@lists.digium.com</a>><br>
Reply-To: <a rel="nofollow" href="http://mc/compose?to=asterisk-r2@lists.digium.com" target="_blank">asterisk-r2@lists.digium.com</a><br>
Subject: Re: [asterisk-r2] dtmf r2 venezuela<br>
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</div></div></blockquote></div></div></div><br><br clear="all"><div><div><div><br></div>-- <br><div><b><font color="#000066">Jose Daniel Yribarren C</font></b></div>
<div><i>Soporte Tecnico <br><font style="font-family:arial,helvetica,sans-serif" size="4"><br></font></i><b style="font-family:arial,helvetica,sans-serif;color:rgb(0,0,153)"><font size="4">Compusan C.A</font></b><font size="4"><span style="font-family:arial narrow,sans-serif;color:rgb(0,5,101)" lang="es-MX"><i><span style="font-family:arial black,sans-serif"> </span> <font><span style="font-family:comic sans ms,sans-serif"> <br>
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