[asterisk-users] SIP Problems continue...

[franco escalona]

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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