similar to: Asterisk ended with exit status 1

Hi, I am new to asterisk , i am getting the following error,& the /etc/zaptel.conf file entry is as follows defaultzone=us loadzone=us span=1,1,0,esf,b8zs,yellow bchan=1-23 dchan=24 Parsing '/etc/asterisk/zapata.conf': Found Jul 1 18:33:35 WARNING[16384]: chan_zap.c:664 zt_open: Unable to specify channel 1: No such device or address Jul 1 18:33:35 ERROR[16384]: chan_zap.c:5296

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

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

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

All, I have downloaded and installed openh323 as per the documentation. When the machine now reboots safe_asterisk just keeps restarting. If I start another session and just load asterisk -vvvvvvvgc asterisk loads. If I enter noload chan_h323.so in the modules.conf then safe_asterisk will kick in. Not 100% on Linux yet but I have added the environment variables info into /etc/profile so they

Greetings. I am trying to get AMP up and going on my Asterisk server. I can access the admin pages on my asterisk server via a web browser. I can add and edit things via the web browser and it edits the database accordingly. Everything seems fine except when I try to run 'amportal start'. Below is what I get (Plus tail /var/log/asterisk/full, but the tail of the 'full' log

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! 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

This will sound weird but the command 'asterisk -r -x reload' fails to work when issued by Cron. But it works when I issue it from a bash session. What is not configured correctly? I need to refresh the configuration every a short amount of time. rom root@localhost.localdomain Wed Jun 15 18:42:00 2005 Date: Wed, 15 Jun 2005 18:42:00 -0400 From: root@localhost.localdomain (Cron Daemon)

>Date: Thu, 06 Apr 2006 12:47:00 -0700 >From: Jeremy McNamara <jj@nufone.net> >Subject: Re: [Asterisk-Users] Pickup() h323 >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <44357034.5090802@nufone.net> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Dan Austin wrote: >

I want to share some facts with the Asterisk community. I have been very successful providing a Vonage-type system based on Asterisk. For instance, one company that uses Asterisk and offers a similar service to Vonage is Voyze.com. The key concept is that Asterisk works like a Cisco, for all the intelligence is provided by SQL Server, outside Linux. I don't even save the CDR locally. The

I have no problems lauching asterisk from the command line . . . asterisk -vvvvc However, I'm trying to autostart on boot up, so I'm trying safe_asterisk When I do this, I get: Asterisk ended with exit status 127. Asterisk died with code 127. Aborting. I've looked through the script but can't find what the problem is. I'm running on RedHat Fedora. Thanks for your help.

I'm having a problem with a new installation of asterisk 1.2.5 with a digium dual port T1 (span 1 connected to an outside line, and span 2 connected to a CAC access bank I channel bank with 24 fxs ports). When I start Asterisk (either from safe_asterisk or asterisk -vvvc) it will immediately exit after it initializes. It will start the logger, register applications and functions, register

I tried to use it to create a 'trunk' to Cisco's call manager. The 0.7.1 code worked up to a point. The call would be established, but audio was one-way from the Call Manager. Asterisk with Chan_h323 would not setup the sending rtp stream. The debug results showed the sending stream as using ip:0.0.0.0 I have not checked for a CVS update to see if it is fixed, or if that

Hello! Tried to compile, but : [CC] chan_h323.c -> chan_h323.o chan_h323.c: In function '__oh323_update_info': chan_h323.c:349: error: dereferencing pointer to incomplete type chan_h323.c:350: error: dereferencing pointer to incomplete type chan_h323.c: In function 'oh323_rtp_read': chan_h323.c:790: error: dereferencing pointer to incomplete type chan_h323.c:791: error:

hi all I'm trying to setup a dlink dph-100h phone (actually a dph-100m but with the h.323 software) with asterisk and chan_h323. AFACS, the dph-100h software can only be configured to point to a gatekeeper. I know I don't need to do this, but it's a test before I setup my Symbol Netvision phones (I don't have an access point for them now). First, this is what I have, and what

Hello all, I have problem. I can't makel asterisk addon, asterisk-ooh323. I use Asterisk and addons svn version. OS:redhat EL4 Linux 2.6.9-5.EL #1 Wed Jan 5 19:22:18 EST 2005 i686 i686 i386 GNU/Linux Please help me . [root@asterisk asterisk-ooh323c]# make make all-am make[1]: Entering directory `/usr/local/src/asterisk-addons/asterisk-ooh323c' source='src/chan_h323.c'

Heya, I changed the Makefile of the h323-channel-makefile (I downloaded cvs of a couple of hours ago) so that I don't have to rebuild pwlib and openh323, but can use the precompiled versions. I'm using pwlib 1.8.3 and openh323 1.15.2. There aren't many changes. I replaced OPENH323DIR with OPENH323INC ,which points to /usr/include/openh323 for me and OPENH323LIB, which points to

Hi, all! when my h323 phone dial in Asterisk system, i can hear nothing. and the following is the log slice i picked from /var/log/asterisk/full. ps: i am using red hat AS5 kernel 2.6.18-53.el5,Asterisk-1.4.24.1, pwlib_v1_11_0, openh323_v1_19_0_1. Best Regards! 81948 [May 6 10:07:34] VERBOSE[11579] logger.c: -- Remote UNIX connection 81949 [May 6 10:07:51] VERBOSE[29627] logger.c:

Hello, I have a strange h323 issue. After executing command "Dial("SIP/333-0d1dfe00", "H323/361737052390920 at ccg|5|tT")" at Oct 18 22:32:23. Meanwile I have sniffing traffic on port 1720. The call was established just at Oct 18 22:33:03 (New H.323 Connection created.) and also packet sniffer grabs the h323 invites at this time also. So my question is what