similar to: sipsock_read: BAD! BAD! BAD!

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...

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

How do I allow traceroute to reach my server? Pings work fine but traceroute stops at the last hop before my server. If I shut off the firewall it reaches it fine. PING danicar.net (24.222.246.120): 56 data bytes 64 bytes from 24.222.246.120: icmp_seq=0 ttl=237 time=104.0 ms 64 bytes from 24.222.246.120: icmp_seq=1 ttl=237 time=74.9 ms 64 bytes from 24.222.246.120: icmp_seq=2 ttl=237 time=90.6

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

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

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 have two Asterisk Systems. One on of those, when I execute this: Dial("SIP/teleglobe-007931d0", "SIP/13033372500 at teleglobe|60|oL(4007520000:60000:30000)") ... It causes Asterisk to immediately read out the time limit of the call (66,792 minutes), as soon as the other end answers, even though we aren't down to 60s remaining yet. Asterisk then goes into an infinite

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

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,

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

I've been getting this Warning message for a while now.. Apr 27 13:56:45 WARNING[1142106560]: chan_sip.c:5775 sipsock_read: Recv error: Resource temporarily unavailable and from what I can tell, this warning coinsides with a dropped call.. I'm running Cisco Gateways with Cisco ATA's (running 3.1 firmware) and I am doing Re-invites with NAT & STUN (and in some cases RTP aware

Hi, What do you think that the problem might be if a program has a segmentation fault at the same library call? The library call is from libpthread.so.0 and the call itself is "pthread_mutex_locl ( )". I have enclosed the core dump information below. The program comes up and then does the segmentation fault. (gdb) bt #0 0x40035944 in pthread_mutex_lock () from

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

I 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

Hi all, I have seen a lot of message talking about asterisk crashed when using queue and mixmonitor together. I do use both in our system and also get the crash (segfault) randomly. I don't know it is related to the reason above as I have no idea about how it happened. I get the core dump below. If anybody has any idea about the root cause of the crash, please tell me. Asterisk 1.4.13

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'