similar to: FW: No dial tone after installation

I'm a newbie, after installing asterisk software and the Digium Lite card I do not get a dial tone. The telephone push buttons makes tones but no dial tone. Is this a hardware problem ? Can some one point me in the right direction. Sincerely, Don Moskaluk don@moskaluk.com www.moskaluk.com --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com).

I need to find some basic configuration files. Is there a place I can check out how to set up an office using sip telephone and Digium FXO and FXS ports? Don Moskaluk don@moskaluk.com www.moskaluk.com 416 737-8230 Cell 416 614-8230 Home --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date:

Yeah, I'm a newbie and am having problems with missing .conf files in /etc/asterisk/ I get notices when I try to run asterisk like: parking.conf is depreciated in favour of features.conf Please rename it. (I'm not getting rename parking .conf to features.conf ???) I am also missing: mgcp.conf enum.conf cdr.conf I using dev-lite Digium board that has 1 fxo and 1fxs port. So far I was

I currently working on a project using Locustworld's wireless mesh network and running SIP through the network. I want to install Asterisk as server running sip and Digium FXO FXS dev lite card. Unfortunately I having the time of my life to install Asterisk. The astinstaller does everything perfectly but when I try to configure an analogue telephone to setup voice mail I do not get any dial

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

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Hi, Has anyone made a successfull instalation of cdr_odbc?? I've install unixODBC-2.2.8 (made my own RPM) and then built the module. I'm trying to send the cdrs to a M$ SQL server. The sql connection works because I can do any query via isql. When I do the calls I get the following output on the asterisk console: -- Executing Hangup("SIP/test1-a5e1", "") in new

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

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

Hi I tried to install the * in my Radhat 7.3 machine. The "gmake" and "gmake install" went fine, but "gmake sample" gave me errors of "configs/*.adsi", "sounds/demo-*" and "sounds/*.mp3" errors. I created a dummy.adsi in the configs folder and commented out the "sounds/demo-*" and "sounds/*.mp3" in the Makefile to

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

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:

Here are the linker errors: lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: __data_start__ lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: __data_end__ lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: __bss_start__ lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: __bss_end__ lld: warning:

I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc. I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs pwlib openh323 gnugk for h.323 gatekeeper

COFF lld doesn't support the linker script at the moment, and I'm sad to say that it is very unlikely to support that in the future. Linker script support is so huge that I can't imagine we really want it for COFF. GNU BFD linker supports it because the linker is built as an interpreter for the built-in linker script (and that's one of the reasons why GNU linker is by far more

Hello, I have a user who is trying to get LLD to link for the cygwin target: https://github.com/zig-lang/zig/issues/751 Currently the issue they are running into is needing to define a linker script, but the COFF driver (or MinGW driver) does not have support for that. Is there documentation or advice for how to use LLD to link for cygwin? As a starting point, which driver to use? Regards,

Is that the only problem to use lld to link cygwin programs? On Thu, Feb 8, 2018 at 8:19 AM, Andrew Kelley <superjoe30 at gmail.com> wrote: > Here are the linker errors: > > lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: > __data_start__ > lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: > __data_end__ > lld: warning:

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

On 16-08-16 17:45, George Joseph wrote: > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > 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

hi i updated asterisk today via svn no i can'T start asterisk i get core dumps. i have to comment some modules then i can start: noload => format_au.so noload => format_mp3.so noload => format_pcm_alaw.so.so noload => format_pcm_alaw.so compiling was fine just some warnings somebody has any idea? -------------- next part -------------- An HTML attachment was scrubbed...