similar to: anyone seen this error when running asterisk!

hi michael, here are the core dumps. only kphone works when 0.5.5 and * cvs. audiocodes and msn messenger all cause seg faults when calling ccm thru * (or vice-versa) ~kelvin [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper

Hi, compiled 1.0 on macosx latest (10.3.5). compiled fine. when running, complains about voicemail2 module. Any hints? Marc. == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk CVS-HEAD-09/23/04-09:20:48, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster@digium.com>

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in

Hi, I'm running asterisk-1.0.2-2mdk. When I tried to start it with /usr/sbin/asterisk -vvvvvvvvvvvvvvvvvvvvgc, I get [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) Ouch ... error while writing audio data: : Broken pipe # ps aux | grep mpg123 root 5237 0.1 0.4 5816 4444 pts/0 S 18:45 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.mp3

I upgraded my server from Fedora Core 4 to Fedora Core 5. I was wondering if anybody else has run into the problem and know's the fix? I recompiled asterisk and if I don't have the /usr/lib/asterisk/modules/codec_g729a.so file in place it works. I use or used to use the licensed G729 Codec from Digium. This is the error message from asterisk -vvg: [app_playback.so] => (Sound File

Asterisk Users: I've been working hard on app_voicemail2 which is an enhanced scalability version of app_voicemail. Specifically, its features are: * Highly improved internal architecture (maybe someone else can actually code on it) * Foot print for getting mailboxes from DB (for Vonage) * Segmentable mailboxes, allowing you to truly multihost voicemail for multiple companies

Does anybody have any reason why I should *not* permenantly replace app_voicemail with app_voicemail2? If so, speak now or forever cvs update -D "8/18/2003". Mark

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give

Voicemail.conf in MySQL is not functioning where I get the following error from Asterisk messages log file: CLI debug output is as follows: Executing VoiceMailMain2("SIP/2205-3df0", "") in new stack -- Playing 'vm-login' -- Playing 'vm-password' -- Incorrect password '1234' for user '0' (context = <any>) -- Playing

Hi, Every time I dial into my asterisk box i hear nothing but asterisk jittering. The following is an example of what I get on the asterisk CLI Thanks *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user

Hi, after cvs upgrading my * installation yesterday, the prompts in both VoiceMail2 and VoiceMailMain2 have become silent. All I get is the initial "blip" followed by the Voice taking breath and being cut off before she has a chance to say "Comedian Mail". All other prompts (ie the Playback application) seem to work fine. I can still login to VoiceMailMain2, however, each

Hi all, I have done some minutes ago a full CVS update, like that: cvs checkout zaptel zapata libpri asterisk cd zaptel make clean ; make install cd ../zapata make clean ; make install cd ../libpri make clean ; make install cd ../asterisk make clean ; make install When I try to start astersik with asterisk -vvvvvvc I get the following error and the program stops:

Hi, Yesterday evening I have done a full update of Asterisk on a test system. The version is CVS-08/25/03-15:55:51 After this operation I get some big problems: - the Voicemail2 application does not work anymore. I must disable it in modules.conf file in order to be able to start * without crashing. The following settings: noload => app_voicemail2.so noload => app_sayunixtime.so If

Hello all - This is my situation! I have a PC with no PCI slot and no modem! But I would like to install asterisk on it. I want to use SIP based software ONLY with the asterisk PBX system. I already have the asterisk running on a regular PC. BUt when I run assterisk it fails at this point when it parse modules.conf file. [Wait] == Registered application 'Wait' Asterisk Dynamic

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing

Hi everyone, I am running asterisk on red hat linux 9 box. The sound card is Intel 82801db AC' 97 audio and the module is i810_audio. It runs well with other applications like xmms and the standard tests deliver a sound . I have also tried to record voice and that works well too. 1-)Now when i run asterisk and i dial out an extension to play any sound there is none. The same thing

I've added an important new application: app_transfer. This application is designed to allow Asterisk to request the transfer of an incoming call to a different extension. Consider the following diagram: Caller -> [ PBX1 ] -> SIP or IAX2 -> [PBX2] -> Transfer App A caller calls an extension on PBX1 which forwards to PBX2. PBX2 executes app_transfer, which requests that hte

Hi, I've been having a little problem with my asterisk servers, I have 4 identical asterisk servers setup (same hardware, same OS, same config). Once in a while (once or twice a day) one of the server crashes on the cron job reload. But I realized this only happens on 3 of the 4 servers. Tried to spot the difference between that one server that wasn't crashing. The difference I found was