Nobody is using r option anywhere in my dialplan, thats 4 sure. And im also not using any PSTN line to connect to outside world. my system is based on voip only.<br><br>SIP PHONE<---------->ASTERISK<---------->CARRIER-OUT
<br><br>No GUI is involved. i edit the conf files myself.<br><br><div><span class="gmail_quote">On 5/31/07, <b class="gmail_sendername">Eric ManxPower Wieling</b> <<a href="mailto:eric@fnords.org">eric@fnords.org</a>> wrote:
</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Rizwan Hisham wrote:<br>> Hi all,<br>> when a user dials any number, asterisk automatically generates ringing
<br>> which<br>> caller can hear, and after 2 - 3 rings asterisk detects that the called<br>> user<br>> is busy, then caller hears busy tone. for example user hears---<br>> tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing
<br>> at the start so that user hears only beep beep beep if the called user is<br>> busy. I have used the R and r options in Dial application but they dont<br>> work.<br><br>Remove the "r" option from Dial.
<br><br>I assume you have the following:<br><br>SIP Phone <-> Asterisk w/FXO card <-> POTS line<br><br>If you are using AMP or any other GUI for Asterisk, then my advice is<br>not valid, since those GUIs take over everything, hide the important
<br>stuff, and add options to Dial that you never see.<br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>asterisk-users mailing list
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-- <br>Rizwan Hisham<br>Software Engineer<br>AXVOICE Inc.