similar to: Avoided initial deadlock asterisk v 1.2.12.1 SIP clients IAX2 termination.

Hi all, Can tell me somebody what meen : channel.c: Avoided initial deadlock Our customer makes calls with our softphone (with IAX2). Sometimes the softphon freezes. The call is ACTIVE but the user cant hang it up. At this time in the log file (asterisk/messages) appear the next line: channel.c: Avoided initial deadlock..... we use: SVN-branch-1.2-r46176M with VoIP channel (ADSL) Can you help

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

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

I never so this error. I am using H323 with Asterisk 1.2.6 Any idea what can be the problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060401/2e8ad498/attachment.htm

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

Hi guys, First of all, I know that this server must be upgraded asap, I'm just wondering if anyone of you has already faced this problem and , if so, would the upgrade solve my problems... CAPI version 0.6 Asterisk 1.2.5 AGI scripts are being used Main problems: -Dropped Calls - ps aux | grep asterisk shows that asterisk (that is started with safe_asterisk) is generating multiple

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',

An Asterisk box at customer site shows these messages pretty regularly. This causes one way voice, the called party cannot hear the calling party. Apr 7 11:59:44 WARNING[18406] channel.c: Avoided initial deadlock for '0x817b790', 10 retries! Apr 7 14:47:46 WARNING[18406] channel.c: Avoided initial deadlock for '0x81a4380', 10 retries! Apr 7 14:58:53 WARNING[18406] channel.c:

Hi! We have an Asterisk Bussines Edition ABE-A.1-6 on a PowerEdge 2850 with 4Gb RAM. It was working 24/7 without any for a month, but for not related causes I rebooted it a week ago. Yesterday the machine suddenly stop working, with a kernel panic. We was watching logs, and found in /var/log/asterisk just before the machine hung the messages posted avobe(is the first time we see it). Anyone

Hi, After upgrading from: Zaptel 1.2.9.1 Asterisk 1.2.12.1 with bristuff-0.3.0-PRE-1s to Zaptel 1.2.10 Asterisk 1.2.13 with brustuff-0.3.0-PRE-1v I get the following error when connecting my Xlite Softphone: --- cut --- Nov 4 17:33:45 WARNING[4430]: chan_sip.c:1090 __sip_xmit: sip_xmit of 0x886df58 (len 486) to 192.168.9.9:31308 returned -1: Operation not permitted --- cut --- It seems to be

Hi, My asterisk sometimes stop responding to iax calls. In the log, I've found this: Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[13002] chan_sip.c: update_call_counter(1409) - decrement call limit counter Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29

Anyone have any experience with a Japanese local VoIP termination supplier? I've emailed a few companies looking to setup some PSTN to SIP and SIP to PSTN termination, but no luck so far. Thanks, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100505/5068aaab/attachment.htm

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

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

Hi guys, How are you keeping ? I use 'smbmount' and a custom script to mount remote shares from Windows desktops and back them up to a Linux server. Everything works great except that I get "Resource deadlock avoided" when trying to copy 'Outlook.pst'. This of course is due to users' having their Outlook open when my backup script runs. Is anyone aware of a way to

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 -------------- next part -------------- An HTML attachment was scrubbed...

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070628/6b5382fc/attachment.htm

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

I have been seeing this problem for a long time and it occurs in 1.4.0b2 (as well as 1.2.0-1.2.12.1). If the internet connection is lost and I have SIP services that require me to register, any SIP devices attached to the system stop working. I have an IAX phone connected to one of my servers that I've been having this problem with which will work fine (and filover to the PSTN) the

Hello, I am using chan_capi 0.3.5 and Asterisk CVS-v1-0-12/29/04-15:32:48 on a SuSE Linux 9.0 with Kernel 2.4.21-99-default In the system is a AVM C4 with one port connected to PSTN at PTP BRI and another one to an ISDN PBX with an PMP BRI. The system is running fine, but I have regualary this error, and then chan_capi is not working anymore. Jan 18 15:29:46 WARNING[2919]: Avoided initial