similar to: How to minimally configure modules.conf loading?

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'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

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

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

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

Compiled Asterisk from FreeBSD port (0.9.0_2) When I start asterisk it uses 100% cpu. Searches on Google say to comment the noload => chan_oss.so in modules.conf But this is already commented. Make.conf contains some optimizations. modules.conf: ; Asterisk configuration file ; ; Module Loader configuration file ; [modules] autoload=yes ; ; If you want, load the GTK console right away. ;

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 *, 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

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

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 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

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

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! 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 >

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, 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

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:

Afternoon all, I'm trying to load Asterisk, however I am getting the following error: [skipping res_musiconhold.so] [chan_sip.so]May 22 18:42:24 WARNING[16384]: loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop May 22 18:42:24 WARNING[16384]: loader.c:421 load_modules: Loading module chan_sip.so failed! I've tried doing

This is a new install so it's a fresh checkout of everything. Does anyone know what might cause this error? [chan_sip.so]WARNING[16384]: File loader.c, Line 239 (ast_load_resource): /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop -- BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643

Hi, I have upgraded my PBX to Asterisk 1.2.5 , previously I was using Asterisk 1.0.9, and Every thing was working fine ,But now voice mail is not working. The error I am receiving in log files is like following, WARNING[2413] app_voicemail.c: No entry in voicemail config file for '12' I have searched for solution a lot can Any one of you let me know how can I solve this issue