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<DIV><SPAN class=970565815-09052007><FONT face=Arial size=2>SIP channel hang ups
are progressively getting worse and I'm really grasping at straws here trying to
find out what the cause is. The problem start, once a week or so the SIP
phones couldn't communicate with the server, though there was no error message
on the server and everything appeared fine on the server. It's now doing
it multiple times a day and I fear having to go back to our old phone system if
I can't find a fix in the near future. When the SIP channel locks up the
only fix is to restart Asterisk. SIP RELOAD & RELOAD CHAN_SIP do no
good.</FONT></SPAN></DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial size=2>Here's a few things
I've noticed and changes I've made in hopes of making it better. First,
I've currently got 71 active SIP channels when only 2 people are on the
phone. This doesn't happen every time, but could be part of the
cause. The 'ghost' channels are all INVITES, how do I clear these without
rebooting the system?</FONT></SPAN></DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial
size=2>10.200.26.116
716 0a2a959d3d3
00102/00000 unkn No Init:
INVITE<BR>10.200.26.115
715 1dee947d485
00102/00000 unkn No Init:
INVITE<BR>10.200.26.104
704 28808764699
00102/00000 unkn No Init:
INVITE<BR>10.200.26.104
704 36d3e88f59c
00102/00000 unkn No Init:
INVITE<BR>10.200.26.104
704 0e00060800d
00102/00000 unkn No Init:
INVITE<BR></FONT></SPAN></DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial size=2>Second, I've gone
through and basically redone my extensions.conf to have it flow much smoother
and clearer. I thought for sure my problem was coming from a loop
somewhere in extensions.conf, but I'm now certain my extensions.conf is fine
(but I'm glad I redid it, much easier to follow now).</FONT></SPAN></DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial size=2>Third, I removed
'qualify=yes' from my sip.conf. I had read where people were having SIP
channel lockups with this enabled, I again thought I had found the problem...but
alas...In addition I had seen someone suggest setting REINVITE=NO, in addition
to CANREINVITE=NO...no good.</FONT></SPAN></DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial size=2>Fourth, I downgraded
all my GXP-2000's to the latest released version of the software (1.1.1.14),
some were on a newer version that I'm not sure where it came from
(1.1.2.x). I also removed the 2 phones that were on 1.1.3.x (they can't be
downgraded), as those apparently had lock up issues as well...again thought I
had found the problem...</FONT></SPAN></DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial size=2>Fifth, I installed
the latest SVN of 1.4 last night in hopes it was a known issue that had been
fixed....nope....</FONT></SPAN></DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial size=2>We don't have a very
complicated setup at all. The server is running CentOS 4, it has two
TDM-400 cards with 6 FXS & 2 FXO. We have about 25 GXP-2000
phones. My dialplan is nice and clean now. </FONT></SPAN></DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial size=2>If no one has any
further suggestions I'm to the point of opening a bug report with digium.
I've read a ton on other people who have had this problem and followed the fixes
for those people, but I can't seem to get to the bottom of it. I have
multiple SIP DEBUG console logs and DEBUG/VERBOSE set to 4 logs around the time
SIP stops responding.</DIV></FONT></SPAN>
<DIV><SPAN class=970565815-09052007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial
size=2>SIP.CONF:</FONT></SPAN></DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial
size=2>[general]<BR>bindport=5060<BR>bindaddr=0.0.0.0</FONT></SPAN></DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial
size=2>disallow=all </FONT></SPAN></DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial
size=2>allow=ulaw <BR>allow=gsm<BR>context=from-internal</FONT></SPAN></DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial
size=2>allowsubscribe=yes<BR>notifyhold=no<BR>limitonpeers=yes<BR></FONT></SPAN></DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial
size=2>[701]<BR>type=friend<BR>secret=blahblah<BR>port=5060<BR>host=dynamic<BR>dtmfmode=rfc2833<BR>dial=SIP/701<BR>context=from-internal<BR>canreinvite=no<BR>reinvite=no<BR><A
href="mailto:mailbox=701@default">mailbox=701@default</A><BR>call-limit=9<BR>allowsubscribe=yes<BR><BR></FONT></SPAN><SPAN
class=970565815-09052007><FONT face=Arial size=2>Thanks for any
help,</FONT></SPAN></DIV>
<DIV><SPAN class=970565815-09052007><FONT face=Arial
size=2>Ken</DIV></FONT></SPAN></BODY></HTML>