similar to: Problem with chan_dahdi and conferencing

How did you install these packages -- make sure you do ./configure and if needed make menuselect in each one of these before the make and make install. This is the only thing I can think of -- check whether there are any built-in modules as well. on Monday 03/12/2007 Asterisk Asterisk(asteriskbunnies@yahoo.com) wrote > Hey! > > Thanks for your interest, i checked the modules and i

Oh, I forgot to mention that Asterisk 15 went End-Of-Life last Thursday. :) You should use Asterisk 16. On Mon, Oct 7, 2019 at 5:58 AM George Joseph <gjoseph at digium.com> wrote: > > > On Fri, Oct 4, 2019 at 1:19 PM John Covici <covici at ccs.covici.com> wrote: > >> Hi. I am trying to install asterisk 15.7.4 from git onto a Debian 10 >> system and I am

Hi, I don't have a zaptel device for conferencing. I read from the lists, that ztdummy and zaprtc need to be installed to get conferencing. I could able to compile successfully with ztdummy and when i receive the call it says, -- Goto (13732,s,1) -- Executing MeetMe("SIP/-08118800", "1234") in new stack == Parsing

I originally built my Asterisk server without installing the Zaptel package as it was going to be a purely SIP based system. However when I went to setup conferencing using meetme I found out that app_meetme is dependant on the ztdummy for timing. I have now installed the zaptel package and I believe the ztdummy module is loading ok [root@astro asterisk-1.4.0-beta2]# lsmod Module

We recently updated our Asterisk (1.4.24) box from Zaptel (1.4.12.1) to Dahdi (2.1.0.4). Everything seemed to go smooth with the exception of meetme. Meetme seems to not be able to find a zap channel for conferencing. We use voice introductions in our conference bridge and it seems to break that feature. The error from the console is.... # app_meetme.c:2593 find_conf: No Zap channel available for

Hi all, I'm trying to get the MeetMe system to take a caller and announce to them they've joined the conference in addition to the other members of the conference assuming previous members of the conference >= 1. I can see where the meetme.c app actually processes it using the ast_pthread_create_background(&conf->announcethread, NULL, announce_thread, conf); function. The

Hello ! In file app_meetme.c we can read A ZAPTEL INTERFACE MUST BE\n" "INSTALLED FOR CONFERENCING FUNCTIONALITY.\n" I receive message, when I try conference WARNING[28686]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' Does it means that I cannot establish conference without any hardware zaptel interface ??? What

Hi all, I downloaded a copy of bristuff 0.4.0-RC11 on my Ubuntu 10.04.2 LTS server machine with 2.6.32-24-generic-pae kernel. The prereq.sh script executes without complaints (BTW on my system, libncurses-dev evaluates to libncurses5-dev and libz-dev evaluates to zlib1g-dev). With the asterisk 1.4.41 package that is installed, a make menuselect operation indicates that all dependencies are met

Some recent issues I had with hardware seem to come back to not understanding two very similarly named files: /etc/asterisk/dahdi-channels.conf /etc/asterisk/chan_dahdi.conf I've modified the chan_dahdi.conf to work now, but it would appear all I needed to do was include dahdi-channels.conf in chan_dahdi.conf and the problem would not have persisted? Is it me or is that a bit Monty Python?

Hi guys, I'm currently playing around with optimizing an Asterisk install (trying to remove as many possible configuration files as possible) for testing and debugging purposes. I've been able to remove most of the files and maintain an error-less Asterisk full file, with a single exception: I can't seem to remove `chan_dahdi.conf` without having Asterisk complain in the full logs:

I am really a beginner of PRI ISDN board, I am wondering if there is a "quick start" chan_dahdi.conf configuration I could use. I tried to install two "FreePBX" boxes follow the instructions from "http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html" connected them between PRIs, It worked. And now if I refer the FreePBX "chan_dahdi.conf" it looks

Hi, I am trying to build asterisk with xtensa compiler on embedded platform. I am trying to integrate my driver code to asterisk. For this tying to call driver code IOCTLs from chan_dahdi instead of dahdi IOCTLs. While compiling asterisk with xtensa, Observed chan_dahdi is not compiling [chan_dahdi.so is not creating].But with default settings [/usrc/src] it was creating. Please let me

Hi, It seems BRI signalling settings are missing from http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conf I would like to add those parameters : bri_cpe_ptmp bri_cpe bri_net Is this http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conftied to a specific Asterisk version ? Can I edit this ? Regards -------------- next part -------------- An HTML attachment was

Hi, these are my first steps with DAHDI and I finally managed to get asterisk to load chan_dahdi (after I found out, that I need libpri). But how do I tell chan_dahdi on which isdn numbers it should react? I haven't found a parameter like "incomingmsn" for chan_capi in the documentation. Thanks for your help, Stefan

Hi I am experiencing Asterisk Crash. Log got stopped when asterisk crashed. Please help me to identify the reason and fix this issue. Asterisk: 1.8.20 I am using AMI and fastAGI to control the call. Some part of dial plan is also defined in extensions.conf I am experiencing this crash when app_meetme conference functionality is used with more than 3 parties. I faced this issue with

This Friday's edition of the weekly VoIP Users Conference call is all about wideband audio (aka HD Voice) and conferencing. The guest for this call is David Frankel, CEO of ZipDX a commercial service that specializes in wideband conferencing. We expect an interesting call touching on many aspects of VoIP going beyond the traditional phone service, conference bridges, technical standards,

We've started to use Asterisk for conferencing and have been getting some complaints. Our configuration is that some people call in from home, but we have a physical conference room with a Polycom. When somebody was giving a presentation in the physical conference room, we were told that the remote people kept hearing him cut in and our. To me, this sounds like the talking optimization was

Hi it seems like chan_dahdi.so is missing in debian asterisk 1.4.21.... so I have upgraded to 1.6 and no I can load chan_dahdi.so Command 'module load chan_dahdi.so' failed. [Jun 16 21:22:30] WARNING[4360]: loader.c:417 load_dynamic_module: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref [Jun 16 21:22:30]

Hi. I want to use the new asterisk 1.4 with dahdi, but I would like to know the svn branches for the dahdi, so I can use them that way -- much easier to keep up with bug fixes, etc. Thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com

HI, I'm using asterisk-1.4.24, dahdi-linux-complete-2.4.0+2.4.0 and libpri-1.4.11.4 When dial, when 492131 answer, in console appear some error messages -- AGI Script Executing Application: (DIAL) Options: (DAHDI/g1/492131|60) -- Requested transfer capability: 0x00 - SPEECH -- Called g1/492131 [Dec 4 11:15:59] WARNING[7669]: chan_dahdi.c:1776 dahdi_enable_ec: Unable to enable