Displaying 20 results from an estimated 20000 matches similar to: "Channel Lock problems"
2007 Aug 30
0
WARNING[22292]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x82f2fe0', 9 retries!
Hello!
I clear remarks in Makefile:
DEBUG_THREADS = -DDEBUG_THREADS -DDETECT_DEADLOCKS
But same things in CLI:
Aug 30 18:16:31 WARNING[22292]: channel.c:780 channel_find_locked:
Avoided initial deadlock for '0x82f2fe0', 9 retries!
-- Zap/32-1 is proceeding passing it to Zap/31-1
-- Zap/32-1 is ringing
-- Accepting call from '2177' to '7141278' on channel
2006 Nov 22
0
channel_find_locked: Avoided deadlock ... messages - What to do?
What are these?
Nov 22 09:35:23 WARNING[7127]: channel.c:787 channel_find_locked:
Avoided deadlock for '0xf6c06778', 10 retries!
Nov 22 09:35:24 WARNING[7127]: channel.c:787 channel_find_locked:
Avoided deadlock for '0xf6c06778', 10 retries!
Nov 22 09:35:24 WARNING[7127]: channel.c:787 channel_find_locked:
Avoided deadlock for '0xf6c06778', 10 retries!
Nov 22 09:35:25
2010 Nov 24
2
Avoided deadlock Error
My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem
is :
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
deadlock for '0x861f6d8', 9 retries!
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
deadlock for '0x85a6420', 9 retries!
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
2007 May 29
2
channel_find_locked: Avoided deadlock
Hi
i have 20 people calling agents calling
when ever they calling i get this below error
May 30 00:46:57 WARNING[2840]: channel.c:785 channel_find_locked: Avoided
deadlock for '0x8b2f50', 10 retries!
and the voice go choppy, and voice breakages
iam using Latest SVN, any suggestion to come over this problem
ram
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2006 Feb 08
1
channel.c: Avoided deadlock for '0x91a8b20', 10 retries!
Dear users,
Couple of days ago I have updated my * to 1.2.4 with ZAP 1.2.3
Since the upgrade I am having these problems:
Feb 7 16:21:18 WARNING[387] channel.c: Avoided deadlock for '0x91a8b20', 10
retries!$
Feb 7 16:23:16 WARNING[16176] channel.c: Avoided deadlock for '0x91a8b20',
10 retries!$
Feb 7 16:23:28 WARNING[16176] channel.c: Avoided deadlock for '0x91a8b20',
2008 Dec 04
0
Deadlock ? I hope i am wrong
I have thousands if this messages in the logs:
Dec 4 10:53:43 NOTICE[26310]: app_queue.c:1980 wait_for_answer: No one is
answering queue 'COMMERCIAL-WT' (2/0/0)
Dec 4 10:53:43 WARNING[5602]: channel.c:889 channel_find_locked: Warning:
Avoided contention wait for '0xb77482c8', 10 retries! RETURN = NULL
Dec 4 10:53:43 WARNING[5602]: channel.c:889 channel_find_locked: Warning:
2006 Oct 18
0
Please explain these SIP errors
Hi,
sometimes on by Asterisk 1.2.10 box I get these errors, there are about 50 active SIP channels so I
dont know if calls are getting dropped or not. Should I be worried?
2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided deadlock for
'0xb7341470', 10 retries!
-- Executing GotoIf("SIP/sipCSC-b737f9e8", "0 ? 15") in new stack
2006 Oct 19
0
Please help with these SIP errors
Hi,
sometimes on my Asterisk 1.2.10 box I get these errors, there are about 50 active SIP channels so I
dont know if calls are getting dropped or not. Should I be worried?
2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided deadlock for
'0xb7341470', 10 retries!
-- Executing GotoIf("SIP/sipCSC-b737f9e8", "0 ? 15") in new stack
2009 Sep 08
2
1.2 AGI Deadlock
I am running 1.2.34 (also tried on 1.2.32) and whenever I launch an AGI, I
get the "avoided deadlock" message below.
*CLI> == Spawn extension (CONTEXT3, 6080, 8) exited non-zero on
'SIP/3211-1-081c40a8'
-- Executing NoOp("SIP/3211-1-081c40a8", "") in new stack
-- Executing AGI("SIP/3211-1-081c40a8", "diallocal.agi") in new
2007 Jan 27
1
Via EPIA channel_find_locked: Avoided initial deadlock
In asterisk 1.2 branch SVN 51363
zaptel svn 1980
libpri svn 393
addons svn 332
My equipment is a Via EPIA minit-itx CN10000 1.2ghz, 1gb ram and a
tdm400p (4fxo).
A call comes from zap, a SIP ulaw receives the call, talks for a while
and when SIP users tries to park the call, then dozens of...
WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial
deadlock for '0x91bb840', 10
2009 Sep 27
0
channel.c:780 channel_find_locked: Avoided deadlock
Hi All.
I have many days reading and research about asterisk and vicidial. I thing
this issue is about asterisk and doesnt about vicidial. Isn't it?
I have a problem with theses application (I already ask for help in vicidial
forums), but I can not fix it.
I have debian 5 with asterisk 1.2.24 and vicidial 2.0.4. This server has a
IAX tunnel with another asterisk server B which connect to
2006 Mar 28
1
Redirect problem/bug/feature
I have a major problem with SIP redirects, or maybe just do not understand
how they are supposed to work. We are using Cisco 7940s and 7960s with SIP
version 6.3. Asterisk 1.2.5.
A call come in to extension 944 over the IAX trunk. Extension 944 has
forward all to extension 904 set on the phone. According to the dialplan.
extension 904 should ring for 90 seconds, then ring another extension, and
2006 Jan 31
2
Asterisk hangs on 1.2.1
Anyone have any idea what's causing this or how to debug it? I'm pretty
sure the root cause is with chan_sccp.so, but not sure how to prove it.
I recently upgraded from CVS-head to 1.2.1 and the chan_sccp from
12-17-2005. Now, once or twice a week, I get this on the console:
Jan 31 10:39:08 WARNING[10586]: channel.c:784 channel_find_locked:
Avoided deadlock for '0xbf1013e0',
2008 Jan 09
1
Help! channel_find_deadlocked: Avoided initial deadlock for ...
Hope someone can help.
I have a situation where asterisk is sending a SIP CANCEL message before the Dial() timeout has hit. It doesn't always do it.
Normally, we send an INVITE to the ITSP. They respond with a 100 Trying, then a 180 Ringing, or 183 Session Progress. It seems to be at this point that Asterisk starts the dial timer. Normally, when no more replies have been received by the dial
2005 Jul 07
0
Parial Hang with cvs-HEAD and queues/agentcallbacklogin
Hi
Last night I upgraded an asterisk install from cvs of early this year to
current cvs head and all seemed to be working OK, but now I'm having several
problems which seem to be related to queues. First off queues don't work,
there's no error message, the channel just seems to hang - cli output as
follows:
-- Executing Answer("Local/00@local-208c,2", "") in
2007 Jun 28
0
Avoided deadlock for '0x864e70', 10 retries!
Hi
iam using 1.2.X SVN
iam keep getting the below message
Jun 28 23:07:31 WARNING[2692]: channel.c:785 channel_find_locked: Avoided
deadlock for '0x864e70', 10 retries!
any help
ram
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2007 Aug 29
1
WARNING[11439]
Hi All!
Yesterday has established Asterisk 1.2.21.1 on Gentoo.
Prompt the reason of the following message:
Aug 29 14:06:24 WARNING[11439]: channel.c:780 channel_find_locked:
Avoided initial deadlock for '0x815d548', 9 retries!
--
wbr. Eugeniy Khvastunov,
[FMGH-UANIC]
eugeniy.khvastunov at digma.ua
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Name:
2005 May 26
1
deadlock
All out of the blue I get these errors?
Any Ideas why
Please help
May 26 09:54:28 WARNING[3964]: channel.c:507 ast_channel_walk_locked:
Avoided initial deadlock for 'SIP/301-b9d0', 10 retries!
May 26 09:54:30 WARNING[3964]: channel.c:507 ast_channel_walk_locked:
Avoided initial deadlock for 'SIP/301-b9d0', 10 retries!
May 26 09:54:33 WARNING[3964]: channel.c:507
2007 May 09
1
Replaces header
I'm tying to use park and announce for call park on Asterisk 1.4.2 but
I'm having trouble getting it to work properly. This use to work with an
older version of Asterisk.
A telephone on the PSTN calls an IP phone. The IP phone is assigned
extension 3-8396. 3-8396 answers the call and attempts to perform a
blind transfer to x700, the parking lot number. The transfer gets to
Asterisk,
2008 Jan 30
2
sipsock_read: BAD! BAD! BAD!
Does anyone know the cause of these BAD BAD BAD messages?
I think I lost all my calls when it happened too. We have nagios running against IAX and nagios reports that IAX is down. It would seem that the entire application locks up when this happens and calls are dropped.
Connected to Asterisk 1.2.14 currently running on flexo (pid = 26846)
Verbosity is at least 3
flexo*CLI> show channels