similar to: How to avoid "Avoiding initial deadlock...."

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:

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

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

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

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

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

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',

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

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

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

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

Periodically (as in sometimes several times a day and sometimes never) I get A channel.c: voided initial deadlock for '0x82*****', 10 retries! The ***** figure is different each time. When this happens an active call (in or out) is dropped. The setup is as follows: handset --SIP--> Asterisk 1.2.12.1 --IAX2--> Terminating supplier (don't know which software they are running).

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

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

Hello all, I'm running Asterisk 1.2.35 with chan_mgcp activated. The process host around 2,4K users. Along the day I've got some debug reports like : Feb 24 22:25:42 DEBUG[28546] channel.c: Avoiding deadlock for 'MGCP/aaln/1 at 028421223635-1' Feb 24 22:29:04 DEBUG[28670] channel.c: Avoiding initial deadlock for 'MGCP/aaln/1 at 028421223635-1' Then, at

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

Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c:

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

Greetings to all, I ma having a problem with channel variables on a couple of our Asterisk boxes. Here is the setup. Asterisk on customer's site (1.2.5), using IAX to our external GW (1.2.5), IAX to PSTN GW (1.0.10), E1/PRI to PSTN. On the External GW, we also have an IAX trunk to a VOIP provider if for some reason the E1 is down. If the DIALSTATUS is CHANUNAVAIL, which should be

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...