similar to: Avoided deadlock

I don't use my iconnect account much, and so I can't say precisely when outbound calls began to fail. But fail dramatically they do, and I'm not sure how to interpret what Asterisk is doing that their server is choking on. I get these messages in droves, even after I quickly hang up the call: -- Called 12195551212@iconnect Dec 1 02:37:56 WARNING[14865]: chan_sip.c:604

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, December 01, 2004 7:07 AM Subject: Asterisk-Users Digest, Vol 5, Issue 6 > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Hi All, I'm trying to hook up asterisk (CVS-HEAD-02/09/05-13:44:11 ) to a ADIT 600 via MGCP. Got it working really nice but now have a pretty bad problem: 1. When I perform a flash on the telephone, I usually get a second dialtone, but when I dial, dialtone doesn't break. If I flash back and forth a few times, it will eventually give me no dialtone.. here if I dial, it successfully

Hello everybody, I would like to implement "callback" function. When I call a person and his extension is busy I can press, for example, 5 and get a callback when his phone is not busy anymore. When I create a call file and copy it to spool call folder asterisk makes a call. One problem is that when extension is still busy my phone rings and I get busy tone of the person who I am

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

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',

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 all, is there any way to force all sip peers to re-authenticate themselves? thanks, Paradise Dove

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

does * support call return? i want when the operator transfers a call if the transferee is busy or doesn't answer the call the call return back to operator again... this feature may be called: call return on busy call return on no answer Paradise Dove

Hi all, i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco ATA-186 3.1.1 atamgcp We are used to make an special ;) blind transfer like (Flash)Number(Hangup before anyone answers or ring). Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp If one waits until the last one rings, then hangup, everything is fine. If one waits until the last one

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

Has anyone here done this? I got it compiled just fine but when I make a call I do not get any audio going either way. The * box is not behind any sort of firewall or nat. My H323 client (gnomemeeting) is behind NAT but I have it set up properly to work through NAT and it will talk correctly with my other regular x86 box running H323. One odd thing I note is that when looking at the UDP traffic

can anybody help me to choose the most reliable fax solution for * . after googling the net i found that there are at least two solutions for this, app_rxfax+spandsp and iaxmodem+hylafax. - what's the differences between these two? - which one's better? why? thanks

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

Hello! After upgrading my production machine to 1.2.1(used to be 1.2.0) on friday i experienced strange behaviour yesterday, i received deadlock-avoided-messages and channels refusing to hangup on span1(used for inbound calls), both messages in all cases paired: Jan 9 17:40:01 WARNING[30003] chan_zap.c: Ring requested on channel 0/17 already in use on span 1. Hanging up owner. Jan 9 17:40:01

SOFTWARE FreePBX 2.1.3 CentOS 4.4 Asterisk 1.2.13 Zaptel 1.2.10 Sangoma Wanpipe 2.3.4.5 I had an Asterisk server lock up on me today after 95 days of up time. Had to manually kill the Asterisk process and then restart. Nothing out of the ordinary in terms of memory use as far as I could tell. Seems to be running fine now. Here is the log file. I deleted the stuff in the middle to keep it

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

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',

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...