similar to: * newbie: overhead paging and nbsd

I've rummaged through the archives and documentation and have yet to find references to nbsd or mention of how to implement overhead paging using chan_oss as mentioned in the list previously. I suspect that one would use a soundcard in the PBX system and feed the output to speakers and/or PA system. Would someone please point me to some procedures or documentation to acomplish overhead paging?

I recently upgraded a server to Asterisk 1.4.22 with OpenR2. Previously I was using 1.4.18. It seems that 1.4.22 has a big bug using chan_alsa.so for overhead paging. After rebooting the server it would work once or twice and then I just got an error on the CLI: [Jan 7 10:35:14] ERROR[26164]: chan_alsa.c:693 alsa_read: Read error: Resource temporarily unavailable I had to switch to chan_oss

> I have a customer that has two buildings that are connected with a > fiber link. We have a single Asterisk server to cover both buildings. > Now the customer went and bought an overhead paging system for the > remote building and they want to integrate it with Asterisk. Is there a > device that can connect over IP or an ATA that has an audio output port? > The buildings

I wasn't thinking of using the conference system as the basis. I was thinking more along the lines of: 1) Setup a second extension on the Cisco phone named "INTERCOM" enabled for auto-answer 2) Create a call group on asterisk to dial that "INTERCOM" extension on every phone that will participate 3) Add a feature code that would dial the intercom extension and connect

Overhead paging is when somebody picks up an extension, speaks into the handset, and their voice is broadcast throughout a building or zone. > What is overhead paging and how is it done with asterisk? > You can do overhead paging and even stream mp3's using nbsd

We have purchased an Asterisk based PBX solution that is completely setup with one exception, overhead paging. We have a powered Paging System and want to find a way to set an extension (i.e. 999) to use the sound card, which would in turn go out the paging system speakers. I've seen several references to this being possible, but no examples how to do it. The system we purchased does not

Hey everybody, I've been trying to recreate a paging unit that we have in house that basically, when a user calls extension 44, it records their message. When they hang-up, it plays a notification tone and then plays back the message. I thought this should be easy, I have a sound card in the Asterisk box, I have chan_oss loaded and working, I planned on hooking the sound card up to the

Hey guys, I have run into one last issue before I do my full * conversion this evening. I can't seem to get paging to work. I have the chan_oss module loaded as per the wiki, and I have the following in my dial plan ;here is our intercom exten => 6000,1,Dial,console/dsp when I dial it here is the output from the console -- Executing Dial("SIP/3062-4f07",

I'm trying to set up the console as an extension (so I can set up overhead paging), but I can't seem to get it to work. When I call my paging extension, I get an error that it can't open the device: -- Executing Ringing("Zap/9-1", "") in new stack -- Executing Dial("Zap/9-1", "Console/dsp0|18|A(new/whistle)") in new stack << Call

Hey all, I know you all saw the topic and let out a groan. However, I understand how to get an overhead paging system to work with respect *, however I am now looking for a small(?) paging amp, that I could hook 3 or 4 horns to. I would like to just have the * extention be routed to a soundcard and out an output, so I would like an amp that is voice signal activated. Has anyone found anything

Hi, I am trying to setup an overhead paging system with asterisk. I have followed some of the advice from the list and have oss.conf set for autoanswer. The sound card and speakers work because they can play mp3s just fine. When I call the extension, the asterisk console looks like everything is working, but I get no sound. Here is what I get on the console: -- Executing

Hello, For the very first time, I tried the command bellow on a newly build Debian Buster box on which I successfully built Asterisk 17.2.0 before. I got : # contrib/scripts/install_prereq install-unpackaged *** Installing NBS (Network Broadcast Sound) *** A nbs-trunk/LICENSE A nbs-trunk/nbsclient.c A nbs-trunk/nbsd.c A nbs-trunk/nbs.h A nbs-trunk/nbscat.c A nbs-trunk/Makefile

I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we

Does anyone have any recommendations for overhead paging systems for use with Asterisk? Thanks, Randy Johnson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031113/82ae09ec/attachment.htm

I spoke the other day about my preliminary tests with office-wide paging with Cisco phones using the new SIP 6.1 image which supports auto-answer. I've got a small and crude recipe for those of you who want to experiment and hopefully create some better and more complete examples than the one I've thrown together below. Create a new line on each of the Cisco phones, and put the

Thank you for the feedback. This is cross-compiling for mingw-w64-x86_64 using gcc-5.3.0 and mingw-w64-4.0.4 on GNU/Linux. Upon attempting to compile now, a large number of errors occur in flac/decode.c which I have placed at the end of this email. They are eased by adding this to decode.c: #if _WIN32 #include <windows.h> #include <shlobj.h> #endif ...among the headers. Then, this

I've recently had an Asterisk 1.4.x install system crash, luckily I was well into configuring a replacement system. Initially, the system was running: Debian 6 64BIT Asteirsk 11.3 USB sound card (to paging amp) Audio file is sent to the console via /dev/dsp1 on a 3 second time delayed call file. Initial testing shows this working well, but the day before I was to deliver and install, I

Installing DevkitLite hardware (Very similar to John Lange's post on Tue Oct 08 2002) I cannot get anything to work on the phone connected to the s100u. I dont know what to do. Can someone please help me? I used the sample configuration files from digium documentaion that was supposed to be "sane" defaults for the kit. Very similar to John Lange's post on Tue Oct 08 2002 Here

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"