similar to: Asterisk STILL loosing IAX user's registration

Hello list, hope some one could help me find the answer. Asterisk 1.4.16.2 installd as no-root user The main issue is that every now and then, cd * box seems to loose the user's registrations, there is nothing in the console, absolutely no messages, only when another friend trys to dial an extension I can see this on messages logs [Jan 26 10:35:46] WARNING[23015] app_dial.c: Unable to

Hello list, hope some one could help me find the answer. Asterisk 1.4.16.2 installd as non-root user The main issue is that every now and then, cd * box seems to loose the user's registrations, there is nothing in the console, absolutely no messages, only when another friend trys to dial an extension I can see this on messages logs

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! I am running an * 0.7.2 on an X86 debian stable 2.4.25 (with backports.org). The HW I am using is Digium's E100P on an HP DL 380. Quite often it crashes, e.g. after a call has finished. Below some logs form the * Console as well as from the /var/log/asterisk/messages (Replaced some stuff with XXX). Any idea what there could be the reason for this segmentaion fault? What other

Hi! As reported earlier this week, I have problems with a sometimes-crashing Asterisk. In most of the cases safe_asterisk is able to restart it. But sometimes it crashes, so that manual interaction is necessary. The seg-faults and crashes occurs, right after call between a SIP Terminal and a legacy PSTN Terminal (PRI/Euro-ISDN), but only if the SIP Terminal hangs up as first. No problem, if the

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

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 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, 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

Hi, My asterisk is frequently dead by itself. It leaves messages: /usr/sbin/safe_asterisk: line 40: 24890 Segmentation fault (core dumped) asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY} Asterisk ended with exit status 139 Asterisk exited on signal 11. Automatically restarting Asterisk. Anyone has any idea of the cause? Thanks.. Best Regards, Stevanus

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 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

I have a handful of 7912's connected to my asterisk 1.2.14 server. (6 to be exact). I get the X on the display sometimes for loosing registration. I have the config file for the 7912's SipRegInterval: 60 and asterisk is the default. ; maxexpirey=3600 ;defaultexpirey=120 I've not changed them. How can I keep these phones online and stop loosing registration? Thanks, Jerry

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

After just having update to the latest CVS I am getting the following message when I call VoicemailMain(): -- Executing VoiceMailMain("SIP/2302-1a12", "") in new stack -- Playing 'vm-login' (language 'en') -- Playing 'vm-password' (language 'en') -- Playing 'vm-youhave' (language 'en') -- Playing

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",

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, I noticed that asterisk manager interface will only accept the originate commands in sequential order. For example, if I want to ring two extensions through the AMI, and while first extension is ringing, AMI won't execute and ring second extension until first extension has answered the call. Anybody has any ideas as I had the same results even tested with telnet commands to AMI interface.

I was trying to enable CDR in a mysql-database when the following occured : asterisk*CLI> cdr status CDR logging: enabled CDR mode: simple CDR output unanswered calls: yes CDR registered backend: cdr_manager CDR registered backend: cdr-custom asterisk*CLI> exit [Jun 29 21:56:52] Executing last minute cleanups [root at asterisk asterisk]# vi modules.conf [root at asterisk asterisk]# service