search for: do_monitor

...ly... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 4980 (do_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 4980 (do_monitor): Got it eventually... chan_sip.c line 4980 (do_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 4980 (do_monitor): Got it eventually... chan_sip.c line 4980 (do_monitor): Error obtaining mutex...

Hello, Using asterisk 1.8.24 on CentOS 6.4 I notice that the asterisk process is using between 105 en 110 % CPU : PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1765 root 20 0 2508m 102m 8864 S 105.8 2.7 102:11.55 asterisk 2682 mysql 20 0 627m 29m 6204 S 0.7 0.8 1:59.51 mysqld 1 root 20 0 19228 1508 1220 S 0.0 0.0 0:00.75 init

Hi, I'm using Asterisk version (Asterisk CVS-05/30/04-16:28:04) on Debian Woody. Sometimes I get this warning message: Jun 14 13:32:41 WARNING[10251]: chan_zap.c:5044 do_monitor: select return -1: Bad file descriptor When that is happening, Asterisk gets slow and close all remote active connections (asterisk -vvvcr). VoIP call alse gets bad at this time. That message appears many times, like if Asterisk were in loop. I've downloaded the version from the CVS (cvs c...

Did anybody get this error message before: chan_zap.c:5044 do_monitor: select return -1: Bad file descriptor When it's happening, Asterisk gets freezed and talkers can not hear each other. This message appears like in a loop at the server's screen. thank you Oz _________________________________________________________________ MSN Messenger: instale grátis...

I have an extension defined like this: exten => do_monitor,1,Answer() exten => do_monitor,n,NoOp(Just got '${CfMC_ActionID}') exten => do_monitor,n,ChanSpy(${CfMC_WhoHear},qX) exten => do_monitor,n,Hangup() I use an AMI packet like this: Action: Originate Channel: Agent/1001 Exten: do_monitor Context: cfmc_cdi_private Priority: 1 Variab...

...ages: May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:236 __ast_pthread_mutex_lock: chan_sip.c line 3116 (sip_alloc): Deadlock? waited 460 sec for mutex '&iflock'? May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:239 __ast_pthread_mutex_lock: chan_sip.c line 11257 (do_monitor): '&iflock' was locked here. May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:236 __ast_pthread_mutex_lock: pbx.c line 2017 (ast_extension_state_del): Deadlock? waited 460 sec for mutex '&hintlock'? May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:239 __ast_p...

...s that I haven't replaced yet.) I connected that extension to an FXO port on a Digium AEX410 card, and set that channel to have the context "doorphone". The problem is that the extension is never executed. With debugging on, I see: Nov 23 16:37:23] DEBUG[20077]: chan_dahdi.c:11896 do_monitor: Monitor doohicky got event Ring Begin on channel 12 [Nov 23 16:37:23] DEBUG[20077]: sig_analog.c:3621 analog_handle_init_event: channel (12) - signaling (5) - event (ANALOG_EVENT_RINGBEGIN) [Nov 23 16:37:24] DEBUG[20077]: chan_dahdi.c:11896 do_monitor: Monitor doohicky got event Ring/Answered on c...

...e as long as 7000 milliseconds (or about 3 or 4 rings on an ear piece). below is some of the log -- note 2 seconds to get from Ring Begin to In Use and a total of 7 seconds before the sip phone rings. anyway to speed this process up? cheers charles Apr 5 16:15:00 DEBUG[7010]: chan_zap.c:6639 do_monitor: Monitor doohicky got event Ring Begin on channel 1 Apr 5 16:15:02 DEBUG[7010]: chan_zap.c:6639 do_monitor: Monitor doohicky got event Ring/Answered on channel 1 Apr 5 16:15:02 DEBUG[6986]: devicestate.c:187 do_state_change: Changing state for Zap/1 - state 2 (In use) Apr 5 16:15:02 DEBUG[7549]:...

...uct (data=0x81be208) at chan_sip.c:1315 p = (struct sip_pvt *) 0x81be208 #4 0x08056c3e in ast_sched_runq (con=0x8172f28) at sched.c:373 current = (struct sched *) 0x8174868 tv = {tv_sec = 1135275568, tv_usec = 989877} x = 0 res = 1083432672 #5 0x40927e28 in do_monitor (data=0x0) at chan_sip.c:11253 res = 0 sip = (struct sip_pvt *) 0x0 peer = (struct sip_peer *) 0x0 t = 1135275568 fastrestart = 0 lastpeernum = -1 curpeernum = 6 reloading = 0 #6 0x40024b63 in start_thread () from /lib/tls/libpthread....

I met a strange SIP problem recently. In an ordinary procedure, when asterisk loads sip module, a series of functions are called sequentially: load_module()->restart_monitor()->ast_pthread_create()->pthread_create()->do_monitor() However in my system, pthread_create() failed to create a child thread to execute do_monitor(), (though pthread_create() returns a successful signal to asterisk.) Therefore my sip monitor can't monitor the incoming network flow and it results that no SIP packet will be caught. Could anyo...

Ok I know this may have been covered and I did have a look back in the archives but didn't find anthing so I am asking it now.. Many moons ago the MySQL CDR functions and MySQL Voicemail functions had to be removed from the main asterisk code because of licensing issues.. Now there is new MySQL stuff like MySQL FRIENDS for SIP and IAX definitions.. So how is it that these options

...modules/chan_sip.so #2 0x41ba1659 in handle_request (p=0x815eea0, req=0x43131e5c, sin=0x43131e4c) at chan_sip.c:4953 #3 0x41b9d304 in sipsock_read (id=0x80e8da8, fd=9, events=1, ignore=0x0) at chan_sip.c:5281 #4 0x08051420 in ast_io_wait (ioc=0x80e54d8, howlong=0) at io.c:268 #5 0x41b998d4 in do_monitor (data=0x0) at chan_sip.c:5383 #6 0x4003b2b6 in start_thread () from /lib/tls/libpthread.so.0 Here is a partial dump of sip.c starting with the comment line 4952. The call in stack frame #1 is the call to get_destination() /* Get destination right away */...

...ces. I know neither of these devices provide QOS but we usually don't run into any problems. I notice this problem when we are in a conference call with one of the employees at the remote office and a third party. They are disconnected after a minute. Here is the error: Chan_sip.c 11452 do_monitor: Disconnecting call 'SIP/103" for lack of RTP activity in 61 seconds. Any ideas? Jason Adams Sumo Systems 4694 Cemetery Road Suite 310 Hilliard, OH 43026 Phone | 614.433.9906 ext: 102 Fax | 614.433.9931 E-mail | jadams@sumosystems.net -------------- next part --------------...

...bc.so.6 #2 0x403be83e in __sip_destroy (p=0x8248c88, lockowner=1) at chan_sip.c:2278 #3 0x403b1728 in sip_destroy (p=0x49) at chan_sip.c:2366 #4 0x403bd18a in __sip_autodestruct (data=0x8248c88) at chan_sip.c:1370 #5 0x08056cda in ast_sched_runq (con=0x81940d8) at sched.c:373 #6 0x403c8a27 in do_monitor (data=0x0) at chan_sip.c:12201 #7 0x40021a62 in pthread_start_thread () from /lib/libpthread.so.0 #8 0x401e85ca in clone () from /lib/libc.so.6 In frame #4, sip_destroy() was called with the pointer address 0x8248c88. But in frame #3, the pointer address became 0x49. Then when it called __sip_...

...in_zero = "\000\000\000\000\000\000\000"} res = 526 len = 16 __PRETTY_FUNCTION__ = "sipsock_read" #14 0x0000000000496b9a in ast_io_wait (ioc=0x80279afa0, howlong=1000) at io.c:288 res = 1 x = 0 origcnt = 1 #15 0x0000000809069046 in do_monitor (data=0x0) at chan_sip.c:19867 res = 1000 dialog = (struct sip_pvt *) 0x0 t = 1259149207 reloading = 0 __PRETTY_FUNCTION__ = "do_monitor" #16 0x0000000000506f79 in dummy_start (data=0x80279acc0) at utils.c:861 __cleanup_info__ = {pthread_cle...

...e 'es') -- Playing 'digits/2' (language 'es') -- Playing 'digits/2' (language 'es') -- Playing 'digits/2' (language 'es') -- Playing 'vm-isunavail' (language 'es') Jan 18 18:06:17 NOTICE[16386]: chan_sip.c:11213 do_monitor: Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11 seconds Jan 18 18:06:17 WARNING[17340]: file.c:583 ast_readaudio_callback: Failed to write frame == Spawn extension (default, 331222, 3) exited non-zero on 'SIP/172.25.92.153-085340d0' The channels has RTP...

...t (p=0x81944d8, req=0xbe7fe7ac, sin=0xbe7fe79c, recount=0x666d7464, nounlock=0x666d7464) at chan_sip.c:11083 #26 0x4048150d in sipsock_read (id=0x813ed80, fd=15, events=1, ignore=0x0) at chan_sip.c:11377 #27 0x080558dd in ast_io_wait (ioc=0x8162320, howlong=1718449252) at io.c:284 #28 0x404776a9 in do_monitor (data=0x0) at chan_sip.c:11536 #29 0x40034cc4 in pthread_detach () from /lib/libpthread.so.0 #30 0x40201037 in clone () from /lib/libc.so.6 Please anyone can help me with a suggestion? I can also post asterisk debug trace anyway. Thaks for all, flavio -- ******************************** * (o<...

...g on my logs this message: Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160 on channel 3 Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160 on channel 3 (repeated much more then what I will show here). I see that it comes from static void* do_monitor(void *data) in chan_zap.c, but I do not understand what does it mean, and now why is it spamming my logs. Can anyone give me a hint...? - diego

hi,all i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to ReceiveFAX, link to a E1 (DE410P) using dahdi this can receive the fax from E1 successfully, but i see many error message in the log like this: [Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called with no recorded file descriptor. when i receive a 5 pages fax, i will see this error message over 200 lines..... it

Hi, Using 1.4.26.1 & DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] & same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? Thanks, --Barry