Thanx for ur reply.<br>Im running * 1.4.2. i dont think there is any problem in asterisk because only one user is having this problem.<br><br>User is using Aastra 480i Cordless phone<br>Here is the sip config for that user. Im using call-limit=2 for every user
<br><span style="color: rgb(0, 153, 0);">[saadfarr]</span><br style="color: rgb(0, 153, 0);"><span style="color: rgb(0, 153, 0);">username=saadfarr</span><br style="color: rgb(0, 153, 0);"><span style="color: rgb(0, 153, 0);">
type=friend</span><br style="color: rgb(0, 153, 0);"><span style="color: rgb(0, 153, 0);">secret=123</span><br style="color: rgb(0, 153, 0);"><span style="color: rgb(0, 153, 0);">qualify=no</span><br style="color: rgb(0, 153, 0);">
<span style="color: rgb(0, 153, 0);">nat=yes</span><br style="color: rgb(0, 153, 0);"><span style="color: rgb(0, 153, 0);">insecure=port,invite</span><br style="color: rgb(0, 153, 0);"><span style="color: rgb(204, 102, 0);">
call-limit=2</span><br style="color: rgb(0, 153, 0);"><span style="color: rgb(0, 153, 0);">host=dynamic</span><br style="color: rgb(0, 153, 0);"><span style="color: rgb(0, 153, 0);">dtmfmode=rfc2833</span><br style="color: rgb(0, 153, 0);">
<span style="color: rgb(0, 153, 0);">context=local</span><br style="color: rgb(0, 153, 0);"><span style="color: rgb(0, 153, 0);">canreinvite=yes</span><br style="color: rgb(0, 153, 0);"><span style="color: rgb(0, 153, 0);">
accountcode=1:0:saadfarr</span><br style="color: rgb(0, 153, 0);"><span style="color: rgb(0, 153, 0);">amaflags=default<br></span><br>sip show channels give me the following:<br><span style="color: rgb(0, 153, 0);">Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message
</span><br style="color: rgb(0, 153, 0);"><span style="color: rgb(0, 153, 0);"><a href="http://68.144.208.153">68.144.208.153</a> (None) 1aed3833095 00101/00102 unkn No Rx: OPTIONS</span><br style="color: rgb(0, 153, 0);">
<span style="color: rgb(0, 153, 0);"><a href="http://85.234.144.137">85.234.144.137</a> (None) 256a1914164 00101/01429 unkn No Rx: REGISTER</span><br style="color: rgb(0, 153, 0);"><span style="color: rgb(0, 153, 0);">
<a href="http://216.143.130.70">216.143.130.70</a> 1812923551 70f46ee82be 00102/00000 unkn No Tx: ACK</span><br style="color: rgb(0, 153, 0);"><span style="color: rgb(204, 102, 0);"><a href="http://74.96.225.223">
74.96.225.223</a> saadfarr 0c666dc33b3 00102/00000 unkn No Init: INVITE</span><br style="color: rgb(204, 102, 0);"><span style="color: rgb(204, 102, 0);"><a href="http://74.96.225.223">74.96.225.223</a> saadfarr 522a18fd48c 00102/00000 unkn No Init: INVITE
</span><br style="color: rgb(0, 153, 0);"><span style="color: rgb(0, 153, 0);"><a href="http://66.131.246.220">66.131.246.220</a> foahand2 4e8cfe9c416 00102/00000 unkn No Init: INVITE</span><br style="color: rgb(0, 153, 0);">
<span style="color: rgb(0, 153, 0);"><a href="http://124.29.216.185">124.29.216.185</a> 1212933903 443fdaeb50c 00102/00000 unkn No Init: INVITE</span><br style="color: rgb(0, 153, 0);"><span style="color: rgb(0, 153, 0);">
7 active SIP channels</span><br><br>How do i know which one is dead/zombie channel. I can see 2 channels for user saadfarr. i tried to use soft hangup but it requires channel name.......how do i know the channel name if its a zombie channel.
<br><br><br><br><div><span class="gmail_quote">On 8/17/07, <b class="gmail_sendername">Anthony Cennami</b> <<a href="mailto:acennami@gmail.com">acennami@gmail.com</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;">
Have you looked in show channels/core show channels to see if they have any dead/zombie channels, which you can remove with soft-hangup?<br><br>What version of * are you running?<br><br>What kind of phones?<br><br>What config options are you using in SIP (or other tech) to limit the calls?
<br><br><br><div><div><span class="e" id="q_1147438c6647406b_1"><span class="gmail_quote">On 8/17/07, <b class="gmail_sendername">Rizwan Hisham</b> <<a href="mailto:rizwanhasham@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
rizwanhasham@gmail.com</a>> wrote:</span></span></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><span class="e" id="q_1147438c6647406b_3">
Hi all,<br>Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server?
<br clear="all"><span><br>-- <br>Best Regards<br>Rizwan Hisham<br>Software Engineer<br>Axvoice Inc.<br><a href="http://www.axvoice.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">www.axvoice.com
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http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><br>-- <br><span class="sg">Anthony Cennami
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http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><br>-- <br>Best Regards<br>Rizwan Hisham<br>Software Engineer<br>Axvoice Inc.<br><a href="http://www.axvoice.com">www.axvoice.com
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