similar to: Asterisk Seg faulting.... No core dump.

Hi all, I have installed some Asterisk machine, all with the same problem. My typical configuration is: - Asterisk 1.2.14 (or 1.4.0beta3) - CentOS 4.4 server. The problem is this: When I start Asterisk with the default init script (/etc/init.d/asterisk start) distributed with the source, and kill (or kill -9) Asterisk-pid, then safe_asterisk doesn't correctly work (it dies and not restart

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

My Asterisk installation crashes frequently. Since it's a random event I am not able to reproduce it so I can't say what is making it crash. Here's a snippet of /var/log/asterisk/full just when it crashes: Sep 18 13:42:51 DEBUG[378] chan_zap.c: disabled echo cancellation on channel 31 Sep 18 13:42:51 VERBOSE[378] logger.c: -- Hungup 'Zap/31-1' Sep 18 13:42:51

I have setup Asterisk on 2 Fedora Core 8 machines, and have made it to record all incoming calls. One of the box that have Asterisk 1.4.18 is properly merging calls and the other box that has Asterisk 1.4.15 is recording the calls but not merging them, I have made sure that SOX is installed on the box. Here is the Dialplan of both the machines : exten => 1234,1,Answer() exten =>

Hi *, does someone has a directory that works with the Cisco 7960 and astdb or mysql/ldap? Regards, Andreas _________________________________________________________________ Gaming galore at http://xtramsn.co.nz/gaming !

Hi - I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't make it work- currently, asterisk will not startup because of a bad module. Below are some log files/config files. If anyone has any suggestions, I'd appreciate it. I used Trixbox 2.0 and followed instructions on (http:// sangoma.editme.com/wanpipe-linux-asterisk-atHome) - no problems running through or

Hello list, When you have an asterisk box connected between the VoIP phones and an PSTN gateway what is the role of asterisk. Proxy server: stateful or stateless? From what i read in the: "Understanding the SIP, second edition" from Alan B. Johnston i think that asterisk is a stateful proxy server as well as registration server. Am I right? Can asterisk be configured to work as

Hi List, While I know that "upping" ulimit will fix the issue I am trying to understand what will cause it. I have a few set ups that are almost exactly the same yet some machines used to give this error often and others don't. I also noticed the error a lot more on my boxes running 1.4.X. TIA. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server. My PBX has experienced several core dumps the last couple of days and I am not sure if this is what's causing it, but it always seems to happen when a particular extension on a grandstream phone uses ChanSpy SIP group. I have not been able to locate where the core dump file is being saved. I can't find it in my

Hi, Is there any Royalty one needs to pay when using the inbuilt exisimg asterisk on hold music or when using any other mp3 from a music album. I think we need to pay for the later, but I am not sure if we need to pay for the inbuilt asterisk(freepbx) on hold music. -- Deepak --------------------------------- Yahoo! Answers - Get better answers from someone who

Hi, Since I upgraded from Asterisk 1.2.18 to 1.4.17 I've been experiencing high CPU utilization from the chan_sip module. I've notice the more sip peers I have loaded, the higher the CPU load goes when there are no active calls. I am currently using a Pentium 4 3.0Ghz with CentOS 4 Kernel 2.6.9-42.0.2.EL. I currently have 1558 sip peers loaded in Asterisk and the current CPU load is

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Recently safe_asterisk is failing to pick up ASTSBINDIR. I've never had this problem before and even when I move to back versions I have the issue. I did upgrade safe_asterisk and the init.d scripts a version or so ago but even when I try older ones I still have the problem. When I hard code the location things seem to work. The problem that occurs is: cat:

Hi All, I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the asterisk daemon not the safe_asterisk daemon so when asterisk is running and I ssh tot he server then 'asterisk -vr' to attach to the asterisk console there are no colors. If I use the safe_asterisk script to start asterisk, the colors are fine when I attach through SSH. I found this in the init

Hi All; How can I test manager.conf? Can I telnet to the asterisk machine at the port 5038 and send and receive commands to test if the manager is working fine? How? Regards Bilal

Hello, I've found a couple of previous posts on this subject, but with no posted resolution... I'm attempting to run * as a non-root user (asterisk), following the guidelines on the wiki: http://voip-info.org/tiki-index.php?page=Asterisk%20non-root I can run * as my new user with "/usr/sbin/asterisk -vvvvc" without problem. However, I'm unable to run * using

Hi All, I'm interested in how people are "clustering" Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible. I

Hello, I have a few entries for sip peers in sip.conf with different name and username, like [TestSIPUser] type=peer host=dynamic username=testuser secret=1234 context=test_context [TestNewUser] type=peer host=dynamic username=newsipuser secret=3456 context=test_context When a call is made from any of these peers I want to get the username of the peer. for eg:- If a call is being made from

Hi, I configured Asterisk to run as "asterisk" user, but I see that a user can anyway get a root sheet using !command in CLI. I understood that it's something related to safe_asterisk and TTY console, but modifying the script safe_asterisk I wasn't able to disable this root access. Can someone help me? Thanks. -------------- next part -------------- An HTML attachment was