similar to: Question about echo cancelation

Displaying 20 results from an estimated 100 matches similar to: "Question about echo cancelation"

2008 Oct 13
1
Need help for debuging
I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash? It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened. This asterisk is using as ACD for over hundred agents. #> thread apply all bt ........ ........ Thread 6 (process 20135): #0
2013 Nov 27
2
Asterisk uses 105% CPU
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
2006 Oct 13
1
Asterisk (meetme) and SMP/HT OK?
In the past, there have been reports of problems with Asterisk with multiple processors and/or HyperThreading. I'm having a !@#$ of a problem with an HPDL380 with 2 3.4gHz Xeon processors, 2 gb RAM -- if I got 24 hours I'd think I had died and gone to heaven :) Am I missing something obvious like "Asterisk is single CPU, single core?" I can't access the ILO so I
2023 Nov 09
1
help with crash
2023-11-08 18:14:13] ERROR[571246][C-000017e2] : Got 19 backtrace records # 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed() # 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref() # 2: [0x58e660] asterisk stasis_cache.c:824 update_create() # 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec() # 4: [0x586b90] asterisk stasis.c:1380 dispatch_message() # 5: [inlined] asterisk
2005 Apr 26
10
Ctrl-c crashes R when run as sudo (PR#7819)
I tried to submit this in R, but not sure if it worked. When running R as sudo, using ctrl-c dumps me to the command line. Hitting exit to exit the terminal window results in R taking 100% of resources. I am using R-2.1.0 on Fedora Core 3. Thanks. Manuel
2006 Dec 21
2
asterisk crashed
our * crashed twice in a month with segmentation fault & a core dump. here's the stack trace: #0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6 #1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6 #2 0xb7e17090 in strdup () from /lib/tls/libc.so.6 #3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879 #4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at
2006 Mar 21
7
Multiple processes
Does anyone have any ideas why my recently updated * 1.2.5 system should spawn multiple * process at seemingly random intervals? Regards L:ee ########################################### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ -------------- next part -------------- An HTML attachment was scrubbed...
2006 Jun 18
1
Unexpected ezstream exit
Hi, I've been working on a customized version of ezstream and I've been noticing that with a particular setup after playing particular tracks ezstream terminates. I say terminates because there are no error messages, it just exits. I've tried this on both my version and the stock version available via icecast.org just be certain and it happens with both versions. I'm
2007 Dec 22
1
v1.1.beta12 released
http://dovecot.org/releases/1.1/beta/dovecot-1.1.beta12.tar.gz http://dovecot.org/releases/1.1/beta/dovecot-1.1.beta12.tar.gz.sig Still not a release candidate, maybe the next one.. This release fixes a lot of bugs and adds some new sanity checks. Fixes quite a lot of mbox problems. v1.1.beta11 (no other versions) had a potential security hole where memory was free()d multiple times. \Recent
2005 Jan 14
2
ezstream keeps getting glibc error on song
Hello everyone, I just setup an mp3 streaming server on an old laptop I had laying around using icecast and ezstream. The problem I am having is that when ezstream gets to a certain song in a playlist I get the error "*** glibc detected *** free(): invalid next size (fast): 0x080576e0 ***". I'm not sure if it is doing this because of the song number it is or if it is a
2004 Apr 22
2
Avoiding IAX destroy deadlock
On one of my 3 * servers I get this after 2 or 3 IAX2 calls Apr 22 15:54:39 NOTICE[1150495040]: chan_iax2.c:1271 iax2_destroy: Avoiding IAX destroy deadlock And as if that wasn't enough I get a never ending stream of this error flying off the top of the screen. At which point I can no longer make any calls into or out of the box. Any commands issued at the CLI prompt are ignored so I have
2008 Jul 17
1
OpenH323 and ptlib version for asterisk 1.4.21.1
Hi what version of openh323 and pwlib are suggested for asterisk 1.4.21.1.? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser
2003 Apr 14
2
SIP hanging
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
2004 Aug 18
1
Hangups - SIGFPE in dsp.c
Hi, I'm running the latest CVS HEAD version of asterisk, and I'm experiencing hangups during voice conversation. This happens quite regularely and often. The problem is in dsp.c, line 1235, where it says accum /= len; But `len', at this point, is 0, resulting in a SIGFPE. The routine ast_frame *i4l_read() in channels/chan_modem_i4l.c:411 is setting p->fr.datalen to
2006 Nov 16
5
spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13
Hi, I'm using spandsp-0.0.3 [http://www.soft-switch.org/downloads/snapshots/spandsp/ spandsp-20061116.tar.gz] on a bristuffed asterisk (1.2.13) [http://212.91.251.199/~junghanns.net/downloads/bristuff-0.3.0- PRE-1v.tar.gz] libtiff is at version 3.6.0 Running on: Linux router2 2.6.17-2-686 #1 SMP Wed Sep 13 16:34:10 UTC 2006 i686 GNU/Linux Debian testing distro. I've tried many
2008 Jun 27
2
usb - audio asterisk crashes
I am using usb-audio for Console/Dsp with asterisk. it is crashing 1.4.21 and also svn. During the brief times its working the audio is choppy but understandable. I have used aplay and arecord at the same time on the same wave file and they work fine every time and I have done it MANY times. Asterisk failes after 1 or 2 times. Any ideas on something I can try? Jerry
2004 Jan 15
12
capacity testing
Hello all. I'm new to asterisk and have been using and testing it for about a week now. My initial hope has been to use it as a sip<->h323 gateway to tie SIP & H323 based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks. I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. I have tried a couple CVS version from the past
2009 Mar 21
2
1.6.2 beta 1 crash
Hi, I'm starting testing 1.6.2 beta. CentOs 5.2 I found my first crash, first I have [Mar 20 20:30:41] WARNING[11201]: res_config_mysql.c:611 update_mysql: Attempted to update column 'useragent' in table 'sip', but column does not exist! [Mar 20 20:30:41] ERROR[11201]: res_config_mysql.c:581 update_mysql: MySQL RealTime: Updating on column 'lastms', but
2008 Dec 07
1
Echo Cancelation
Hi All I Have an ISDN 30 circuit passing through an asterisk box to a legacy pbx, all is working well but I have had a problem that modems do not work, I thought of turning off echo cancelation but I cann t seem to find the ial switch do do it, could someone point me in the right direction to enable /disbale ec on a zap channel per call? Thanks Robb
2003 Sep 15
2
echo cancelation
HI all, Having a mental block today - can someone confirm which direction the echo cancelation applies to for the Zap PRI channels? ie. is it removing traces of the transmitted data to the PSTN from the received data, *or* is it removing traces of the data transmitted to Asterisk from the data received back from Asterisk? Got a configuration that is based on a call made from the PSTN to a SIP