similar to: Paging (I know, AGAIN)

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?

We currently have an Asterisk system that is hooked up to our old paging speakers via sound card, plugged into two amps. Each amp drives up to 8 analog speakers in each warehouse (we have 2). Both warehouses are around 30k square feet. Both have a large number of printing presses. The computer system is that is running Asterisk is around 10 years old and starting to fail. I'm looking to

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've looked through the lists and archives for overhead paging and I've seen responses to use nbsd or chan_oss. Does someone have a recipe to follow that shows how to implement overhead paging? best regards, erik

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

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 have a production asterisk-1.2.8 system with FreePBX & PRI Digium card. I am looking for a paging system to an external speaker. I can page to internal Polycom 501 VoIP. But, what hardware or system do I need to integrate with the asterisk to have this acheived. -- Deepak Linux your Life, Don't Window it [[]] { All for the best }

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

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

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

Let post this question.. Because I must be real slow... The following is my config on this... group=1 context=default signalling=fxs_ks channel => 1 context=local signalling=fxo_ks callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no musiconhold=default usecallerid=yes

I see a lot of chatter in the archives about intercom and paging, but has anyone addressed zone paging? Each of the 50 telephones in a large clinic would be members of one or more paging zones. Someone could then page Dr. X in zone #1. Would this be possible with analog phones? SIP? Thanks, Mike

Stutter tone when voicemail in boxHi, I'm having a problem getting Asterisk to make a stutter tone when a voicemail is received. It used to work, but since I rebuilt my server I can't get it going. I've set put a line in the sip.conf file: mailbox=100 to check it, but it's not doing the stutter tone ! I am using the stable version of asterisk from the CVS tree. Any help /

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

Has anyone seen the following problem? Until recently, I couldn't understand why some extensions on my * system would have a "congestion tone" as soon as I picked up the handset. A little sleuthing through the logs and the source code led me to understand that * thought it had seen the extension go off-hook, send some DTMF tones, and then wait. * treated this situation as a

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\"

A while back I converted back to static conf files from a database setup. However I decided to tackle it again. The problem that I was experiencing, was, there was no stutter tone on my sipura 2000 or 3000 when there was a voicemail left at either extension when I was using mysql setup for peers and voicemail. I have 2 contexts... home, office in my voicemail configuration I now use

Hi I have asterisk 1.8.18 with freepbx 2.10.1.9. I get an asterisk crash occasionally with the followingerror. It always seems to happen while paging. 16 spa508g phones 1 snom pa1 paging amp Kelly == Extension Changed 1101[ext-paging] new state Idle for Notify User 101 pbx*CLI> *** glibc detected *** /usr/sbin/asterisk: malloc(): smallbin

== Manager 'sendcron' logged off from 127.0.0.1 vicidialnow*CLI> dial 919545090201 -- Executing AGI("OSS/dsp", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("OSS/dsp", "SIP/19545090201 at sip203||tTor") in new stack -- Called 19545090201 at sip203 Feb 2 13:36:38