<div dir="ltr">No Dial()<div><br></div><div><ul style="margin-top:0px;margin-bottom:0px;color:rgb(0,0,0);font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px"><li style="margin:0.15em 0px;padding:1px 0px 1px 1em;list-style-type:none;background-image:url(http://www.voip-info.org/voip_info_drupal/themes/garland/images/menu-leaf.gif);background-color:transparent;background-repeat:no-repeat no-repeat">
<strong>R</strong>: Indicate ringing to the calling party when the called party indicates ringing, pass no audio until answered. This is available only if you are using kapejod's<a title="Bristuff" href="http://www.voip-info.org/wiki/view/Bristuff" style="color:rgb(67,105,118)">Bristuff</a>.</li>
<li style="margin:0.15em 0px;padding:1px 0px 1px 1em;list-style-type:none;background-image:url(http://www.voip-info.org/voip_info_drupal/themes/garland/images/menu-leaf.gif);background-color:transparent;background-repeat:no-repeat no-repeat">
<strong>r</strong>: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Without this option, Asterisk will generate ring tones automatically where it is appropriate to do so; however, "r" will force Asterisk to generate ring tones, even if it is not appropriate. For example, if you used this option to force ringing but the line was busy the user would hear "RING RIBEEP BEEP BEEP" (thank you tzanger), which is potentially confusing and/or unprofessional. However, the option is necessary in a couple of places. For example, when you're dialing multiple channels, call progress information is not consistantly passed back. Look at <a title="Asterisk cmd Progress" href="http://www.voip-info.org/wiki/view/Asterisk+cmd+Progress" style="color:rgb(67,105,118)">Progress</a>(), the <em>progressinband</em> setting in sip.conf or <a title="Asterisk cmd Ringing" href="http://www.voip-info.org/wiki/view/Asterisk+cmd+Ringing" style="color:rgb(67,105,118)">Ringing</a>() if you would like to avoid the use of 'r' but have issues with the ringback behaviour of Dial().</li>
</ul><div style><font color="#000000" face="Verdana, Geneva, Arial, Helvetica, sans-serif"><span style="font-size:12px">Sds</span></font></div></div></div><div class="gmail_extra"><br clear="all"><div>Alexandre Alencar<br>
Twitter @alexandreitpro<br><div><a href="http://blog.alexandrealencar.net/" target="_blank">http://blog.alexandrealencar.net/</a><br><a href="http://www.alexandrealencar.net/" target="_blank">http://www.alexandrealencar.net/</a></div>
<div><a href="http://www.alexandrealencar.com" target="_blank">http://www.alexandrealencar.com</a></div><div><a href="http://www.servicosdeti.com.br/" target="_blank">http://www.servicosdeti.com.br/</a></div><div>COBIT, ITIL, CSM, LPI, MCP-I<br>
<div><br></div></div></div>
<br><br><div class="gmail_quote">2013/1/21 Eduardo <span dir="ltr"><<a href="mailto:j.eduardo@newteltelecom.com.br" target="_blank">j.eduardo@newteltelecom.com.br</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div lang="PT-BR" link="blue" vlink="purple">
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<p class="MsoNormal">Galera bom dia!!!!, estou usando o elastix em cima de um
link ISDN tudo funcionando ligações recebidas e originadas porém quando ligo
pro numero não me gera tom de chamada mais o ramal toca e atende normal,quando
faço ligação ele me gera tom normal de chamada,alguém teria uma idéia de que
seria??<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">Grato..<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">José Eduardo.<u></u><u></u></p>
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