similar to: "Broadcasting" Calls?

Hi - after hearing others rave about * I thought I'd have a go - extract from a 'make' on a stock debian system as follows... (I tried to post the whole make up to this point but it was too big for the list) make[1]: Leaving directory `/usr/src/asterisk/channels' make[1]: Entering directory `/usr/src/asterisk/pbx' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes

This is cvs version from yesterday. There is small problem with that version. When starting I've got: [...] [skipping pbx_gtkconsole.so] [pbx_spool.so] => (Outgoing Spool Support) /var/spool/asterisk/outgoing [pbx_wilcalu.so] WARNING[1024]: File loader.c, Line 226 (ast_load_resource): /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_select WARNING[1024]: File loader.c,

Good evening, I am just getting started with Asterisk. I have it installed, and I believe I am on the right track, overall, to get it working, but I can't get the linejack to answer any calls. At this point, all I'm trying to do is have Asterisk answer an inbound call on my linejack, /dev/phone0, and play a greeting or tone. I figure, once I am able to get asterisk to actually answer the

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

I downloaded the latest CVS a few days ago. It all compiled nicely on my new AAH platform. However, it won't start up. Investigation of my log files produces this; Jul 26 22:59:18 VERBOSE[31473] logger.c: [pbx_wilcalu.so] Jul 26 22:59:18 VERBOSE[31473] logger.c: [pbx_wilcalu.so] Jul 26 22:59:18 WARNING[31473] loader.c: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol:

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

hi ! i have installed asterisk-1.2.9.1 but am unable to run it i am getting this error "[pbx_wilcalu.so]Jun 11 16:43:00 WARNING[8968]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_pthread_create Jun 11 16:43:00 WARNING[8968]: loader.c:554 load_modules: Loading module pbx_wilcalu.so failed!" can anyone help me i have redhat linux

Hello, I'm trying to call different SIP-accounts to connect them to a conference. This is my call-file : Channel: SIP/test3&SIP/test1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: from-conf Extension: 1000 I get the following in the CLI : [Mar 22 14:40:26] -- Attempting call on SIP/test3&SIP/test1 for 1000 at from-conf:1 (Retry 1) [Mar 22 14:40:26] WARNING[29908]:

Dear LLVM user, We are proud to announce the second European LLVM event on April 12-13 2012 in London, UK, starting at noon on April 12th . This will be a full one-day conference with the intention and aim of exposing new developments and supporting and strengthening the network of LLVM developers around Europe. The format will resemble that of the previous meeting held in London in September

We are in the process of migrating to new database servers. Our current RAC clusters are running OCFS2 1.2.9 on RHEL 4. Our new servers are running OCFS2 1.6 OEL5. If possible, we would like to minimize the amount of data that needs to move as we migrate to the new servers. We have the following questions: 1. Can we mount an existing OCFS2 1.2 volume on a servers running OCFS2 1.6?

Steve, 1) go to /etc/asterisk 2) open modules.conf for editing using vi 3) add this line: noload=pbx_wilcalu.so 4) Save the file 5) Restart asterisk Lightup the candles, open the Cabernet Savignon ( or whatever your prefernce) and call your girlfriend. ;) Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com

I am using outgoing call files. I typically see the "ooh something changed / timeout" on a regular bases every second to be exact. Then it stops until some other call event happens. So I "mv" my call file to the outgoing spool directory, I am listening to that message, another call file is "mv"'ed into the directory and something happens to the timeout that its

I have a confbridge setup that feeds the conference from the ALSA microphone input (this is the conference leader) and then uses app_ices to send the conference audio to icecast. I start the conference leader like this: console dial 1000_admin at conferences I join the ices user to the confbridge with a call file: Channel: Local/1000 at conferences MaxRetries: 2 RetryTime: 60 WaitTime: 30

Le 11 janv. 2012 à 13:00, James Molloy a écrit : > Dear LLVM user, > > We are proud to announce the second European LLVM event on April 12-13 2012 in London, UK, starting at noon on April 12th . > > This will be a full one-day conference with the intention and aim of exposing new developments and supporting and strengthening the network of LLVM developers around Europe. The

On 11/20/2011 4:33 PM, Chris Lattner wrote: > One idea for a hacking session would be a "performance analysis > workshop". People could bring their apps, we could sample them track > down what part of the compiler would need to change and code it up (if > time allowed). > > Given the trade offs involved, it could be helpful to many folks, the > trick is to get

I am not having any luck placing out going calls I dial the number 08 82420173 ( our outside line ) But all I get is engaged signal and log this. Oct 14 08:40:14 DEBUG[16401]: File pbx_wilcalu.c, Line 65 (autodial): Entered Wil-Calu fd=20 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 548

Putting a whole bunch of people into a listen-only/muted Confbridge conference or getting the broadcaster audio into a MOH class and then just having callers attach to that MOH class? Does the the muted side of a Confbridge Room still try to mix in audio from the muted channels or does it just disregard those channels and only run mixes against unmuted channels? Now, if the answer is

HI I am trying to use the outcall going by the wiki.( http://www.voip-info.org/wiki-Asterisk+auto-dial+out) But I keep getting the errors below. Here is a sample of a callout file. What am I doing wrong? ////Begin Outgoing.call//// Channel: sip/2075 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: managers Extension: 2184 Priority: 1 ////End outgoing.call//// Nov 9 20:32:02

I am using Realtime extensions as well, in case that would matter. Following problem arises from time to time, a call will successfully terminate: [May 14 14:31:41] VERBOSE[3274] pbx_realtime.c: -- Executing [t at project_init:1] Hangup("SIP/peer-2-00002f7e", "") [May 14 14:31:41] VERBOSE[3274] pbx.c: == Spawn extension (project_init, t, 1) exited non-zero on

I have a few Samba 3.0.14a servers nicely running with ADS security and winbind on Debian Sarge. The one problem I seem to have is their DNS registration with our WIN2K3 name servers that I have no access to. I see that most WIN2K/XP/03 register automatically (DDNS) or manually with the command ipconfig /registerdns. I can probably ask for records to be manually entered but I thought I would