similar to: WARNING (Sipsock_read) Recv error: Resource temporaily unavailable

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

I have searched all over the web and have not really found anything related to this error.... The only thing I found is related to a system stops responding on DTMF, which does not happen here... THe following is the output from the CLI: *CLI> 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:2332 sip_alloc: Allocating new SIP call for 640E2D47-E98B11D8-8FDBE54A-5FB5A0CB@65.67.76.30

HI all, Recently, I got the following message from CLI and finally the asterisk will hang. Anyone can tell me how to fix the problem or why it will happen. Thanks. Jun 17 14:18:02 DEBUG[24573] channel.c: Avoiding initial deadlock for 'SIP/1127-008d65f0' Jun 17 14:22:45 ERROR[24696]: chan_sip.c:11337 sipsock_read: We could NOT get the channel lock for SIP/1589-0087cdd0! Jun 17

I too am having this problem reported by Frank Hoonhout. Asterisk runs fine for a few minutes and then stops accepting new calls. (I have a standalone server with SIP phones and I'm not doing any external registration). Asterisk CVS-04/07/03-09:28:50 0x420e0037 in poll () from /lib/i686/libc.so.6 (gdb) info threads 16 Thread 14351 (LWP 7258) 0x420e187e in select () from

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Folks, Just a heads up for everyone. We are aware of the problem and are continuing to work on resolving it. cheers, jerry ===================================================================== Samba ------- http://www.samba.org Centeris ----------- http://www.centeris.com "What man

Greetings I have been running * for about a month now. Configuration. (5) Cisco 79xx IP phones (1) XP100P Pentium III (300mhz) 192meg memory Redat 8.0 (updated) It seems to run for about 3-6 hours, then the process stops. I have noticed, that * does not stop, if I do NOT have it register to other sip servers. (FWD and PCH). Here is are the last few lines in the /var/log/asterisk/messages

I have finally crashed Asterisk for the first time and I'm wondering if anyone has seen this. This is a configuration with SIP endpoints and an IAX2 channel to another Asterisk PBX. The main PBX dropped a core file after a SEGV (signal 11 ) with the following trace: #0 0x42079133 in strchr () from /lib/tls/libc.so.6 #1 0x41bb0f9c in _fini () from /usr/lib/asterisk/modules/chan_sip.so #2

I frequently get this error message, it repeats itself hundred/thousands of times and never stops. chan_sip.c:7467 sipsock_read: Failed to grab lock, trying again... During this period, I can make no SIP calls what-so-ever. The only way I've been able to stop it is to killall -9 asterisk. Doing a restart now doesn't respond. Anyone know why? -- Daniel Jimenez

Hi to all, I've a segmentation fault while using asterisk relatime conf with mysql db. I've cretate sip_buddies and extensions tables into db and edit res_mysql.conf, extconf.conf without any issues. So when I start asterisk and my phone try to register using sip user configured in my db, asterisk stops with Segmentation fault error. Follow post gdb backtrace 0 0x400337c0 in

What happens when you do "stop now" like the error states? Sean -----Original Message----- From: Ryan Parlee [mailto:listbox@jesca.com] Sent: Saturday, April 03, 2004 9:56 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Unabled to exit console No matter what I try, Asterisk won't let me out of the console. If I CTRL+C, of course, the process will terminate. I

Hello. I'm seeing this at the console. NOTICE[81926]: File chan_sip.c, Line 5119 (handle_request): Registration from '<sip:marrandy@192.168.1.1>' failed for '192.168.1.70' What's this all about ? Regards...Martin -- Osborn's Law: Variables won't; constants aren't.

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

I just got the following error messages displayed on my Asterisk console: ========================================== Oct 18 21:18:32 ERROR[22095]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/5058977054-e577! Oct 18 21:18:32 ERROR[22095]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Oct 18 21:18:32 ERROR[22095]: chan_sip.c:11325 sipsock_read: BAD!

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

Hi I have a process which reads a socket using recvfrom. After sometime, for no apparent reason, recvfrom exits with an EINTR causing the sender to barf too. Basically the socket on which these processes were communicating no longer exists. On writing a dtrace, I could determine that a signal was received at that socket causing the EINTR. I do not know of way to see who sent the signal (which

Hi all. I have a strange problem at hand regarding UDP fragmentation on Centos7: Applications are unable to receive UDP packets which have undergone fragmentation UNLESS the netfilter modules are loaded. The problem arose on a application which would run fine on OpenSuse but does not work on Centos7. The application processes UDP data and on Centos only small packets are received and processed,

Dear All, I am using asterisk 1.4.21.2. I have used Originate manager application to to call the two persons. I have called AGI application to call another person. There I have used GET FULL VARIABLE AGI command to get the value. I am able to call another person form AGI. But when one end cut the call another one didn't disconnected. The following errors are displayed in Asterisk console,

Hi all, I'm unable to put app_festival to work. I successfully patched, installed and tested festival (interactive logon and telnet to server port) which seems to work without problems. But when I test it in asterisk I got the following trace in console: -- Executing Answer("SIP/bsenicar-850b", "") in new stack -- Executing

Thank you for your response. Yes, I know something changed, hence the smiley. I have looked at the code (I was able to build it under the new windows bash). It turns out that the error was on a udp receive, not a tcp connection. Didn't know you would get this error. Anyhow near the location of the error there was a remark about IPv6. I hadn't looked at that yet. So I turned IPv6 off at

nmbd has been dying on me occasionally. I'm running mandrake 9.2 with samba3-server-3.0.0-2mdk. We've got our users in ldap but I'd seen this symptom previously when I was using an earlier version of samba that didn't support ldap. There is nothing useful in the logs and a PS shows a nmb process still running. This last time around, I had a ptrace running on both nmbd