[asterisk-users] SIP Problems continue...

[Ken Williams]

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.
 
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?
 
10.200.26.116    716         0a2a959d3d3  00102/00000  unkn  No
Init: INVITE
10.200.26.115    715         1dee947d485  00102/00000  unkn  No
Init: INVITE
10.200.26.104    704         28808764699  00102/00000  unkn  No
Init: INVITE
10.200.26.104    704         36d3e88f59c  00102/00000  unkn  No
Init: INVITE
10.200.26.104    704         0e00060800d  00102/00000  unkn  No
Init: INVITE

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).
 
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.
 
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...
 
Fifth, I installed the latest SVN of 1.4 last night in hopes it was a
known issue that had been fixed....nope....
 
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.  
 
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.
 
SIP.CONF:
 
[general]
bindport=5060
bindaddr=0.0.0.0
disallow=all                   
allow=ulaw                  
allow=gsm
context=from-internal
allowsubscribe=yes
notifyhold=no
limitonpeers=yes

[701]
type=friend
secret=blahblah
port=5060
host=dynamic
dtmfmode=rfc2833
dial=SIP/701
context=from-internal
canreinvite=no
reinvite=no
mailbox=701 at default
call-limit=9
allowsubscribe=yes

Thanks for any help,
Ken
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