similar to: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x8446b50', 10 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

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

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

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 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

Hi, I installed H323, however when I make a call from SIP Phone -> Asterisk H323 -> Provider H323 the provider can hear me, but I cannot hear nothing. The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP. Any thoughts? -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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',

I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if I call the traffic still go throw the asterisk. How come? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060617/8f4449fa/attachment.htm

Hi, someone can help me, I am ot able to find the way how to user Ajax.updaterto test if the request give some positive or negative result. I am able only to return the result inside a div. An example is appreciated. _______________________________________________ Rails-spinoffs mailing list Rails-spinoffs-1W37MKcQCpIf0INCOvqR/iCwEArCW2h5@public.gmane.org

Hi, I installed this program but I am not able to configure, it does not want to work. Someone can help me?

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, someone know a good webphone, possibily a free one Thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060627/0e83bc29/attachment.htm

Hi, I download the last svn and I also look around but I cannot find the source, I only found the patch http://bugs.digium.com/print_bug_page.php?bug_id=8919 any one can help me out. thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070209/6780fde6/attachment.htm

There is a way to force the order of the codecs in the sip.conf since the allow seams to let know only the accepted codec. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071022/619b8f2b/attachment.htm

I'm trying to add ooh323c to my asterisk 1.2.7.1 install and did an svn update of asterisk-addons and followed the readme in asterisk-ooh323c and I get through the .configure with no errors. But make causes: rpath /usr/local/lib -L./ooh323c/src -version-info 1:1:0 -lpthread make: rpath: Command not found make: [libchan_h323.la] Error 127 (ignored) I'm not real sure what to try to fix

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

Hello, I have asterisk 1.4 with ooh323c addons installed. (As I am a newbie in voip world...my question might be idiot...! ;) Please forgive me!) I succeed to make H323 call when ooh323c is configured as gateway (gatekeeper=DISABLE in ooh323.conf). When I put gatekeeper= ip_address, and add an account as follow : [aaa] type=friend username=aaa password=xxxx host=dynamic context=test

hi has any one used OOH323C i tried this it is installed but do not know how to configure has any one used this, what is the best h323 addon to use with asterisk

Hi, I'm using Asterisk-1.4.7.1. Everything was working fine. Now I'm trying to Install Asterisk-addons-1.4.2. The procedure I followed is as... # cd asterisk-addons-1.4.2 #./configure #make menuselect #make #make install Everything is going fine except make install. I've tried many times, but the same error I'm gettiing--- The error is--- asterisk-addons-1.4.2]# make install