[asterisk-users] SIP Problems continue...

[Eric "ManxPower" Wieling]

Go back to 1.2.x and see if it fixes the problem.

Ken Williams wrote:
> 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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