<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=us-ascii">
<META content="MSHTML 6.00.6000.16414" name=GENERATOR></HEAD>
<BODY>
<DIV><SPAN class=903423020-01052007><FONT face=Arial size=2>I posted about this
problem last week and thought it was a combination of SIP/ZAP causing issues in
Asterisk. Since then I've realized it's only the SIP channel that's
hanging. When this happens a call can still come in and hit the IVR, but
no one can connect to the server from a SIP client. </FONT></SPAN></DIV>
<DIV><SPAN class=903423020-01052007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=903423020-01052007><FONT face=Arial size=2>I tried reloading
chan_sip.so today when this occurred, and I tried unloading chan_sip.so but was
told the channel was in use. How can I clear SIP connections? With
ZAP channels I can use ZAP DESTROY CHANNEL, but I don't see the equivalent for
SIP. </FONT></SPAN></DIV>
<DIV><SPAN class=903423020-01052007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=903423020-01052007><FONT face=Arial size=2>Any suggestions for
tracking down what's causing SIP to hang? My only option as it stands is
to shutdown asterisk & restart it, I included a piece of the log last week
and am willing to do so again if needed. If I can see which SIP channels
the server thinks are open when the channel hangs I'm hoping this will allow me
to find if it's a common phone or perhaps some dialplan logic gone
bad.</FONT></SPAN></DIV>
<DIV><SPAN class=903423020-01052007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=903423020-01052007><FONT face=Arial
size=2>Thanks,</FONT></SPAN></DIV>
<DIV><SPAN class=903423020-01052007><FONT face=Arial
size=2>Ken</FONT></SPAN></DIV></BODY></HTML>