<div>Hi,</div>
<div> </div>
<div>Which asterisk version are you using. try setting call-limit value in sip.conf and see if it makes any difference.</div>
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<div><br><br> </div>
<div class="gmail_quote">On Thu, Oct 21, 2010 at 1:29 PM, GBR Icasiano, Ryan A. <span dir="ltr"><<a href="mailto:raicasiano@globalbridgeresources.com">raicasiano@globalbridgeresources.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">Hi,<br><br>Here is the scenario:<br>1. 1st phone calls and asterisk dials to extension no.<br>2. Extension answers 1st caller(which makes it busy).<br>
2. 2nd phone calls and asterisk dials to extension no.<br>3. 2nd phone hears a BUSY tone, but have to wait for the timeout to expire(in DIAL cmd) before proceeding to the next step in dialplan<br>4. Get the current DIALSTATUS, but it returns NOANSWER, instead of BUSY<br>
<br>the problem is, since the 2nd caller hears a busy tone, it should not wait for the timeout to expire, and proceed immediately in fetching the DIALSTATUS.<br>I also tried this scenario and used DEV_STATE, but it always returns NOT_INUSE<br>
<br>I already assigned qualify=yes in my sip configuration but still to no avail.<br><br>any ideas?<br><br>regards,<br><br>RYAN ICASIANO<br><br>--<br>_____________________________________________________________________<br>
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</blockquote></div>
<div><br><br clear="all"><br>-- <br>Thanks & Regards,<br>Godson Gera<br><a href="http://blog.godson.in/2010/10/asterisk-vs-freeswitch-channel-tracking.html">FreeSWITCH Asterisk Billing Consultant</a><br></div>