Have reported the bug. <br><br>
<div class="gmail_quote">On Mon, Mar 17, 2008 at 9:26 PM, Donny Kavanagh <<a href="mailto:donnyk@gmail.com">donnyk@gmail.com</a>> wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Please try the latest version and if you can re-create file a bug @ <a href="http://bugs.digium.com/" target="_blank">http://bugs.digium.com</a><br>
<br>Donny<br><br>
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<div class="Wj3C7c">On Mon, Mar 17, 2008 at 12:08 PM, Rizwan Hisham <<a href="mailto:rizwanhasham@gmail.com" target="_blank">rizwanhasham@gmail.com</a>> wrote:<br></div></div>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
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<div>Hi all,</div>
<div>i have just encountered a strange problem. Maybe its a bug. </div>
<div> </div>
<div>Suppose a is a user is on a call whose call-limit=2. "sip show inuse" shows the inuse counter of the user is set to 1. But after reloading sip (the user is still in conversation and call is up) the "sip show inuse" command shows that its inuse counter is set to zero. It shouldnt do that. i have seen inside the code in chan_sip.c that while doing sip reload, asterisk destroys all the user (not peers) by freeing the memory allocated for all users and destroying their channel variables. As it destroys all the users, their inuse counter is also lost. what i want to do is, before destroying the users, if any user's inuse counter > 0 then save it in some place, rebuiled users after reading the config files and then set their inuse counter again by reading their saved inuse counter.</div>
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<div>Can anybody help me solve this problem. Or have another idea to solve this problem?</div>
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<div>Maybe this has been fixed in later versions of asterisk. I am using ast 1.4.2. I havent checked any other version. <br clear="all"><br>-- <br>Best Regards<br>Rizwan Hisham<br></div><br></div></div>_______________________________________________<br>
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</blockquote></div><br><br>_______________________________________________<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--/" target="_blank">http://www.api-digital.com--</a><br><br>asterisk-dev mailing list<br>
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<br>-- <br>Best Regards<br>Rizwan Hisham<br>