similar to: FreeBSD 100% cpu

Hi, I'm new to asterisk and try to get it work with capi.so. When I try to start asterisk with "asterisk -vvvvc" I get the following errors. I couldn't find any hint on the net what may be wrong in my configs. Has anybody got a hint? Here is the error output: [capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Jun 23

Currently we have problems with our asterisk server connect with an Eicon Diva 4BRI Card to a PBX. Asterisk is running in version 1.0.0 on RedHat Enterprise Linux 3AS with kernel 2.4.21-4.EL. The Diva server is running in version 7.5 Can someone help us in reolving the errors with incoming calls? When we try to call to an extension we get these messages in the CLI: -- creating pipe for

Hi *, I have set up a very simple asterisk configuration where I intend to be redirected to the voicemail whenever I dial 100 with my kphone SIP client. The problem is that asterisk can not find the stream 'vm-theperson'. I have made a non-standard installation (since I am just testing), and that file is located in /mnt/tr2/fake_root/installed/usr/local/var/lib/asterisk/sounds. 1. How

Hi Guys, I've attempted to get this moh-native thing to work with no success. I've reviewed wiki, mantis and e-mail postings and I'm confused. The latest I've read is native moh should be in asterisk-addons in format_mp3, but what version will it work with? I've tried asterisk 1.0.1, 1.0.5, addons 1.0.1, 1.0.4 and also -r stable CVS. I followed the wiki example with

i'm having this error message when trying to run the agents-feature Sep 2 17:37:40 WARNING[10445]: pbx.c:1645 pbx_extension_helper: No application 'AgentsLogin' for extension (from-internal, 28, 1) while chan_agent.so is beeing loaded i still don't seem to have access to the commands like agentlogin or agentcallbacklogin. my agents.conf and queues.conf are configured correctly

When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to

Hi, I have read the wiki entries on IAX(2), but I'm afraid, it still have some questions: I have a working connection between two Asterisk-Servers (Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on Debian 3.1) via IAX. Does this connection work with IAX or IAX2? When I try to load chan_iax2.so, I get the error message chan_iax2.so]Oct 11 10:09:52 WARNING[2288]: loader.c:258 ast_load_resource:

I have some problems starting asterisk with a hfc card using zaphfc: [chan_zap.so]Mar 8 17:53:06 WARNING[2447]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_retrieve_call_to_death Mar 8 17:53:06 WARNING[2447]: loader.c:391 load_modules: Loading module chan_zap.so failed! I'm using asterisk from debian/sid: Asterisk

Hi everybody. I'm trying to setting up a V4 BRI EICON card on ASTERISK 1.0.7 My linux is a debian. It was working during a few days an suddenly (after a lot of reboot) I've got this error message that seems to be very popular but I couldn't find any answer on the net : ==============================================. Asterisk Dynamic Loader Starting: == Parsing

Hi :) I've had all this working before, but I'm revisiting it, and in short, I currently have huge problems receiving incoming calls. I've been trying with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel and libpri as of yesterday afternoon. Would someone mind helping? :) My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set as the 'DMZ

Hello List! I finally got asterisk with capi working, and its already answering my call as well! :) Now i would like to call a number from my shoft phone (kphone). This is my extentions.conf: --- [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password

Hi! Please is it possible for Cisco 5XX phones to use asterisk/FreePBX phone directories, and if so, how? Thanks in advance! On Feb 17, 2013 6:40 PM, <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Hi, I'm trying to setup a small BRI ISDN <-> voip gateway. The ISDN card is based on Cologne chipset, so I try set it up with zaphfc. The versions i'm running: kernel-2.4.27 Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e zaptel modules 1.0.7 zaphfc is from bristuff-0.2.0-RC8e When I'm doing the insmod on zaptel, zaphfc, zaprtc: Zapata Telephony Interface Registered on major 196 PCI:

On 16-08-16 04:38, George Joseph wrote: > > > On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > using pjproject 2.5.5 > using asterisk-certified-13.8-cert1 > > > IIRC there were API changes in pjproject 2.5 that aren't accounted for > in

Hello all! My basic problem is that we haven't been able to get realtime to use ODBC to store configuration data. Here are the details: We've installed Asterisk on a CentOS machine as follows: 1. Downloaded, compiled, and installed FreeTDS 0.63 2. Downloaded, compiled, and installed unixODBC 2.2.11 3. Downloaded, compiled, and installed Asterisk, Asterisk-Addons, and Zaptel from CVS

Hi, i've downloaded/compiled/installed the bristuffed asterisk Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a and i'm using it with an hfc card. It runs on a debian 3.1 sarge machine with kernel 2.6.11. Asterisk works well if i configure the card using isdn4linux. I'm having problems dialing out (not tried the input yet). This is the output from asterisk: -- Accepting AUTHENTICATED call from

I'm trying to somewhat reduce the security risk of Asterisk, by loading less modules. In my installation I use SIP and IAX2 for incoming calls, and that's it. No voicemail, no call parking, it just plays back voice clips. I can remove /etc/asterisk/modules.conf modules one by one: ------------------------------------ [modules] autoload=yes noload => pbx_gtkconsole.so noload =>

A little off-topic: I have the following hardware: 2610 XM NM-2V VIC-2BRI NT/TE IOS loaded: flash:c2600-ipvoice-mz.123-5d.bin" I get the following error while booting: %C542-1-UNKNOWN_VIC: VNM(1), vic daughter card has an unknown id of FF Is the VIC-2BRI compatible with the 2610XM? What IOS needs to be loaded? http://www.cisco.com/en/US/products/hw/modules/ps2641/products_tech_note0918

Does Asterisk 1.0.2 support 302 redirects? With 1.0.1 I get: Got SIP response 302 "Moved Temporarily" When forwarding the call to other SIP server. This is a "bug": http://lists.digium.com/pipermail/asterisk-users/2004-May/045774.html --- Jan Baggen - jbaggen@ip2.nl IP2 Internet BV / http://www.ip2.nl

Hi list, Now that I've got my Asterisk server to recognize my HFC-PCI card, I've run into some serious problems. The first thing I noticed was this message that would show up every five seconds on the CLI: Dec 27 15:46:42 WARNING[12484]: chan_zap.c:2512 pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway! == Primary D-Channel on span 1 down