[asterisk-users] SIP Problems continue...

Wed May 9 10:34:33 MST 2007

Started with 1.4.1, then 1.4.2, then 1.4.4, now the latest SVN (63478).

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of franco
escalona
Sent: Wednesday, May 09, 2007 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP Problems continue...

whats the asterisk version your using?

On 5/10/07, Ken Williams <ken at intermountainelectronics.com > wrote: 

	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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