similar to: Inbound Call Errors...

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

Greetings I am receiving following error message. Any idea as to why? WARNING[147466]: File chan_sip.c, Line 4530 (sipsock_read): Recv error: Resource temporarily unavailable WARNING[147466]: File chan_sip.c, Line 4530 (sipsock_read): Recv error: Resource temporarily unavailable Frank...

Folks, I know this must be a configuration problem. Just changed servers last nite -- an interim server running 1.6.1.6. Copied all of /etc/asterisk to the new server and fired it up. Now I'm getting: [Oct 14 12:28:10] ERROR[8471]: cdr_pgsql.c:309 pgsql_log: Failed to insert call detail record into database! [Oct 14 12:28:10] ERROR[8471]: cdr_pgsql.c:310 pgsql_log: Reason: ERROR: syntax

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

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 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

Starting playing with asterisk 1.6.1.0 i found the following problems: In the cdr_pgsql, the sql statement is wrong: 2009-06-25 12:17:01 COT LOG: statement: INSERT INTO cdr

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

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

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing

Greetings. Attempting to connect Asterisk to LDAP database using res_config_ldap module. While trying to register sip client (Ekiga softphone), according to slapd.log, asterisk connects to LDAP server, asks for some attributes to modify (they do exist, and asterisk user has all permissions to do that, etc). And then asterisk application just crashes. Without ldap (using just static users'

Hi, Having installed postgresql-devel-7.4-0.3 and postgresql-libs-7.4-0.3 I'm having probs. compiling cdr_pgsql, can anyone offer any pointers as to what I might be missing? I'm hoping I've just missed out something like postgresql-wibblewobble-7.4-0.3 or something ... Below is the result of a make in the cdr source dir which may help those of you in the know thanks... Andy

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've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

Dear All, I just setup the Asterisk with E100P which it's no problem in Dial In but I have problem when outdial. The connection method is like this : E1 PRI <-SIGNAL-1-> MaxLink (PBX) <-SIGNAL-2-> E100P <-> Asterisk <--> SIP \-----> Analog PHone Now when I tried to dial out by SIP X-Lite on Windows, it shows me Connect, Trying,