similar to: asterisk frequently dead

Went from 1.6.0.6 to 1.6.0.8 and resulted in segmentation fault. Reverted to 1.6.0.6 and back to normal. ------------------ Linux asterisk.hulber.com 2.6.18-128.1.1.el5 #1 SMP Mon Jan 26 13:58:24 EST 2009 x86_64 x86_64 x86_64 GNU/Linux Apr 3 11:49:56 asterisk kernel: asterisk[3780]: segfault at 00002ce1ac0537a8 rip 0000003e980715a8 rsp 00007fff5bf00c30 error 4 Apr 3 11:50:00 asterisk

Hi iam not able to start asterisk give me following error any help STARTING ASTERISK /usr/sbin/safe_asterisk: line 42: 4633 Illegal instruction (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY} Asterisk ended with exit status 132 Asterisk exited on signal 4. Automatically restarting Asterisk. /usr/sbin/safe_asterisk: line 42: 4637 Illegal

Hi I was trying to have realtime voicemail working with ODBC Driver. Everything works fine with MySQL Realtime access, BUT as I want to implement ODBC Storage as well, it seems that everything should go through ODBC ( what I read on voip-info wiki page ) But I do not manage to make it work with ODBC. Outside Asterisk, ODBC works fine, I can access my databases & tables ! Asterisk fails to

Hi all, I use asterisk-1.6.2.7 and asterisk addon version 1.6.2.1. It started ok with out cdr_addon_mysql.o. But when I put cdr_addon_mysql.o in to modules folder, it fail at start and the following out has been thrown: ---------- [root at localhost modules]# /usr/sbin/safe_asterisk: line 145: 13270 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTBINDIR}/asterisk -f ${CLIARGS}

Hi all, I've just built a new installation of CentOS release 5.3 (Final) and have installed both <http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.1.tar.gz>Asterisk 1.6.1.1 and subsequently Asterisk 1.6.0.10 (thinking that I was maybe trying to be too cutting edge) on a Dell PowerEdge sc440 server (nothing complex - Pentium Dual core 2ghz - 1gb ram - 70gb

Hi, Does anyone know how to configure speex in asterisk 1.0.10? I've successfully installed it but cannot get any idea how to set the quality, etc.. Thanks Regards, Stevanus

Hi guys, I 've tried asterisk 1.2.2. It work flawlessly for about 3 days then at the third days I activated setting jitterbuffer=yes and suddenly there is no voice when the call is picked up. It's really weird as if asterisk stops sending rtp packet. I've checked asterisk log and found nothing suspicious. Just weird :S. I tried it in 3 asterisk server and all of them are having

Hi, Does anyone know how to make this thing (7940) work with asterisk (chan_sccp module) ? I've set the configuration according to the wiki and now the phone just keep asking for CTLSEP<xxx>.tlv from my tftp server. In the cisco's web interface, I found this in the device logs : 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 ...

Hi, I tried to connect cisco 7910 into asterisk system using chan_sccp.so. But I got a major issue : - when I called from 7910 to another sip phone in the same asterisk server, the call took place normally. - when I called from 7910 to another sip phone in different asterisk server, the call is answered but I cannot hear nor say anything. The phone just immediately lose its tone. - when I got

Hi, I've spotted weird crash of Asterisk cvs Stable. I have defined queue in queues.conf : [prodaja] music = default announce = queue-markq strategy = ringall context = from-pstn timeout = 15 retry = 5 maxlen = 0 announce-holdtime = no announce-frequency = 30 announce-holdtime = yes monitor-format = gsm|wav|wav49 monitor-join = yes eventwhencalled = yes member => Agent/1000

i m using asterisk 1.8.3 on a centos 5.5 computer when i try to change my name on the voicemail asterisk dump core here what i got on the console -- User ended message by pressing # -- <SIP/6672-00000002> Playing 'auth-thankyou.alaw' (language 'fr') -- <SIP/6672-00000002> Playing 'vm-review.alaw' (language 'fr') -- Saving message as

Hi folks, I'm running Asterisk 11 (at the moment - planning to u/grade to v13.7 LTS), I've just configured the voicemail function, and it's mostly working fine... except when I try to leave a voicemail! This crashes asterisk with no entries in the messages log. The system is running on Centos 6 (or maybe 6.5, I'm not sure how to check this). uname -a returns: Linux

Configuration: Linux wpbx 2.4.9-13 #1 Tue Oct 30 20:11:04 EST 2001 i686 unknown P4 2.5 GHz, 1 GB RAM T400P with 3 T1s plugged in. A flow of 46 calls (spread out over 3 T1s). Each call gets transferred (Dial) to the SIP platform and stays for 5 min. Case 1. Asterisk built out of CVS Mar. 19. Test was running for 3 days. Segmentation fault. Case 2. Asterisk built out of CVS Apr. 1. Test was running

Hey all, I have set up my cisco 30vip using chan_skinny because chan_sccp wont register. The problem I am having is, everytime a call is sent to the phone Skinny/200@jason it rings once, then asterisk segfaults. heres the output -- Executing Answer("SIP/4437821638-7588", "") in new stack -- Executing Dial("SIP/4437821638-7588",

Hi, Recently, I've just installed asterisk along with AMP.. Everything seems to work fine, just when I tried to record my voice via ivr, asterisk won't play the file if I call it. When I test by dialing *99, the record is played, but when I call straight to the digital receptionist, it just stand there about 7 seconds, playing no sound at all and then hung up.. I use AMP version

Hi, I've tried to upgrade my asterisk to 1.2.3 again after disastrous bug incident yesterday but when I called and the phone was picked up, there was simply busy tone... Weird, is this another bug in asterisk 1.2.3? Currently, I rollback again to asterisk 1.0.10...:( Is there any configuration change issue in 1.2.3 cause I've just used my configuration that worked in asterisk1.2.2 ?

I have Asterisk 1.2.29, Zaptel 1.2.24 and Freepbx Setup for a queue up to 15 agents through a PRI line, it was working fine for more than 1 year, suddenly, when there is a load on the queue, the asterisk service disconnects and the calls are dropped. And the service starts again after few seconds, and so on. I am not using fax. I checked PRI by zttool and there are no alarms. The cdr logs

I upgraded * via svn and it did not work !!! 1. asterisk-addon did not compile! pbx:/usr/local/src/svn-versions/asterisk-addons # make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` make -C format_mp3 all make[1]: Entering directory `/usr/local/src/svn-versions/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes

I've just installed asterisk on a Debian Linux (apt-get it) And i have placed two sip phones in sip.conf and i'm testing parking with them I have phone1-SIP/1000 and phone2-SIP/1007 The following happens if i park from calling party and everything is OK 1. Pickup Phone2 and call to Phone1 2. Talk 3. Phone2 dials #700 and parks the call (it is placed in 701) 4. Phone2 is hangup 5. Pickup

Hi, After days tinkering with this digium card (TDM04B), I notice that this card has a slow response in detecting ring signal from pstn and hanging up when the call is over. The delay can consume up to several seconds... Is this normal? Best regards, Stevanus