[asterisk-users] Asterisk stops responding to SIP/ZAP

[Ken Williams]

The problem has pretty much been there from the beginning.  I may
re-arrange cards and see if it happens on one particular channel or if
the problem moves with cards.

Thanks for the response. 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Yuan LIU
Sent: Saturday, April 21, 2007 12:52 AM
To: asterisk-users at lists.digium.com
Subject: RE: [asterisk-users] Asterisk stops responding to SIP/ZAP

>From: "Ken Williams" <ken at intermountainelectronics.com>
>Date: Fri, 20 Apr 2007 07:27:05 -0600
>
>About once a week or so my Asterisk box stops responding to all phones.
>I can pull up the console, do whatever I want at the CLI but the only 
>way to get things working again is to restart Asterisk altogether.
>
>I finally cranked verbose & debugging way up (and watched my log files 
>go from 1mb/day to 100mb/day), but below I believe contains my problem.
>The next line is 1.5 minutes later where I restart Asterisk.

As a general troubleshooting procedure, you want to ask yourself if you
have made any changes before it stopped working.  If not, and especially
if you can restart and get it working again, I'd suspect some hardware
failure. 
(Assuming the problem is reproduceable - I had times when TDM card
stopped working with no trace of error.)  Try installing on another box.

Yuan Liu

>SIP/701 is a Grandstream GXP-2000 phone (we have about 30 of them in 
>place here).  Zap/3-1 is a Digium TDM400.
>
>I can't quite figure out where my problem is, is it the initial 
>exception, is it not getting hung up completely, does it have to do 
>with the call limit on the SIP channel, perhaps 'no provider found'
>statements?
>
>Any help would be appreciated, I have a relatively simple dial-plan, I 
>can send over relevant bits of it if necessary.
>
>Thanks,
>Ken
>
>[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Exception on 12, channel 3 
>[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Got event On hook(1) on 
>channel 3 (index 0) [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: disabled

>echo cancellation on channel 3 [Apr 19 13:51:13] DEBUG[27722] 
>channel.c: Didn't get a frame from
>channel: Zap/3-1
>[Apr 19 13:51:13] DEBUG[27722] channel.c: Bridge stops bridging 
>channels SIP/701-08ee6120 and Zap/3-1 [Apr 19 13:51:13] DEBUG[27722] 
>channel.c: Hanging up channel 'Zap/3-1'
>[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: zt_hangup(Zap/3-1) [Apr 19 
>13:51:13] DEBUG[27722] chan_zap.c: Hangup: channel: 3 index = 0, normal

>= 12, callwait = -1, thirdcall = -1 [Apr 19 13:51:13] DEBUG[27722] 
>chan_zap.c: disabled echo cancellation on channel 3 [Apr 19 13:51:13] 
>DEBUG[27722] chan_zap.c: Set option TDD MODE, value:
>OFF(0) on Zap/3-1
>[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Updated conferencing on 3, 
>with 0 conference users [Apr 19 13:51:13] VERBOSE[27722] logger.c: -- 
>Hungup 'Zap/3-1'
>[Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state 
>change to be queued on device/channel Zap/3-1 [Apr 19 13:51:13] 
>DEBUG[27722] pbx.c: Spawn extension
>(from-internal,201,2) exited non-zero on 'SIP/701-08ee6120'
>[Apr 19 13:51:13] VERBOSE[27722] logger.c: == Spawn extension 
>(from-internal, 201, 2) exited non-zero on 'SIP/701-08ee6120'
>[Apr 19 13:51:13] DEBUG[27722] pbx.c: Launching 'Hangup'
>[Apr 19 13:51:13] VERBOSE[27722] logger.c: -- Executing 
>[h at from-internal:1] Hangup("SIP/701-08ee6120", "") in new stack [Apr 19

>13:51:13] DEBUG[27722] pbx.c: Spawn extension
>(from-internal,h,1) exited non-zero on 'SIP/701-08ee6120'
>[Apr 19 13:51:13] VERBOSE[27722] logger.c: == Spawn extension 
>(from-internal, h, 1) exited non-zero on 'SIP/701-08ee6120'
>[Apr 19 13:51:13] DEBUG[27722] channel.c: Hanging up channel 
>'SIP/701-08ee6120'
>[Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Hangup call 
>SIP/701-08ee6120, SIP callid fb118a6415d13a1e at 10.200.26.101) [Apr 19 
>13:51:13] DEBUG[27722] chan_sip.c: Updating call counter for incoming 
>call [Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Call from peer '701' 
>removed from call limit 6 [Apr 19 13:51:13] DEBUG[27722] devicestate.c:

>Notification of state change to be queued on device/channel SIP/701 
>[Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state 
>change to be queued on device/channel SIP/701-08ee6120 [Apr 19 
>13:51:13] DEBUG[20432] devicestate.c: No provider found, checking 
>channel drivers for Zap - 3 [Apr 19 13:51:13] DEBUG[20432] 
>devicestate.c: Changing state for Zap/3 - state 0 (Unknown) [Apr 19 
>13:51:13] DEBUG[20432] devicestate.c: No provider found, checking 
>channel drivers for SIP - 701 [Apr 19 13:51:13] DEBUG[20432] 
>chan_sip.c: Checking device state for peer 701 [Apr 19 13:51:13] 
>DEBUG[20432] devicestate.c: Changing state for SIP/701
>- state 1 (Not in use)
>[Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found, 
>checking channel drivers for SIP - 701 [Apr 19 13:51:13] DEBUG[20432] 
>chan_sip.c: Checking device state for peer 701 [Apr 19 13:51:13] 
>DEBUG[20432] devicestate.c: No provider found, checking channel drivers

>for SIP - 701 [Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking 
>device state for peer 701 [Apr 19 13:51:13] DEBUG[20432] devicestate.c:

>Changing state for SIP/701
>- state 1 (Not in use)
>[Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found, 
>checking channel drivers for SIP - 701 [Apr 19 13:51:13] DEBUG[20432] 
>chan_sip.c: Checking device state for peer 701


_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users