Overhead Page with OSS/ALSA

General Help
1

Hello all, fine tuning our system, we currently have it set up to do overhead Pages using OSS modules, it works fairly fine and dandy however I do have a few things I’d like to tweak, and was wondering if anyone had some things to add…

But first I’d like to throw some info on the forum regarding how I set it up, there are a lot of pages out there concerning overhead Page and Intercom but the best info I found was at voip-info.org (http://www.voip-info.org/wiki/view/Setting+up+paging+with+a+sound+card) and even that walkthru I had a few complaints about, although they were minor… so here’s what we did…

Firstly, we are using Asterisk 1.2

Step 1-3 from the voip-info.org.

Step4.
I don’t quite understand why asterisk needs to own /dev/dsp. I added asterisk to the audio group, as the file is originally root:audio, but things did not seem to work that way… does asterisk need to lock this file in order to use it? Also, it did not start working even when asterisk did own the file, until we chmod 666 /dev/dsp… strange to me!

Step5.
Alright this makes sense… a config file for the module for chan_oss.so it is oss.conf, however, when I tried configuring it for alsa instead, I simply replicated the file as alsa.conf… this did not work, is there another config file that calls oss.conf and or alsa.conf that I need to edit? I could see this being the case as OSS is default, but I could not find anything relating to it.
also, I don’t think that the file needs to be executable as voip-info says (777) please correct me if I’m wrong.

Step6.
Cool worked good, we don’t use trixbox, but hey, hi five for CentOS fans that do :slight_smile:

step7.
I’m no conf file guru, heck I use the web GUI whenever I can (thanks guys!) But the lines mentioned here are out of date or inaccurate for my installation so I took the easy route!

-Create a CUSTOM Extension (called it INTERCOM myself, if you prefer OPTIMUS PRIME, power to ya!) in the DIAL box put in CONSOLE/dsp. Then submit and update.

-Under paging and intercom, add a paging group, number it whatever you want, for this I used 9.

-Select whatever extension(s) you want to include (ctrl-click for multiple), if you want to page to compatible speakerphones simultaneously, select them as well. But remember if you want to do overhead page you must at least have that extension selected.

-Dial *809 (*80 is default featurecode for paging, 9 being your group)
9 can also be substituted for extension numbers etc etc.

*54 for allow I am unsure of what it does exactly… this may need to be done from your phone prior to trying page.

I hope this helps some people out who struggled with it like I did.

Now for my issue… the sound from my speakers connected to my soundcard seem to be quite distorted (especially durring the BEEP), I am USING OSS however with the alsa module loaded, alasmixer works fine for adjusting volume, but the level seems to change after I exit the program… is it not saving this? I assume that switching to ALSA modules my sound quality might improve, but like I mentioned earlier, I can’t seem to get CONSOLE/dsp to answer when configured with alsa.

Thanks for any help… hope this can help others too

2

oss works okay I suppose, but has nobody set it up to work properly with alsa?

3

I have it working well with Alsa and pretty much followed the steps in Voip-info as well. I did not go the OSS route first though.

So are you saying that you can get it to work using Alsa and the only problem is the level of sound coming out?

From my understanding, you have to use one or the other. I would recommend Alsa.

My issue was that the sound levels would be lost on a reboot. This has been fixed.

M

4

Using asterisk v1.4 and setting up overhead paging is fairly simple. Our situation was to have both overhead paging in the Shop as well as have the speaker ring when the shop phone rang. Here are the steps I followed.

First go into modules.conf and change noload => chan_oss.so to load => chan_oss.so . This is around line 34. Then you have to create the oss.conf file. Here is what mine has

[general]
autoanswer=no
context=from-internal
overridecontext=yes
extension=s
playbackonly=yes

I have intentionally left autoanswer=no to make the speaker ring when the phone rings.

Now using your favorite editor create two simple script files in your /etc/asterisk directory and make sure they are owned by asterisk. The first file is ring.txt and it is like this

#bin/sh

asterisk -rx “console autoanswer off”

and create a second file called ringoff.txt

#bin/sh

asterisk -rx “console autoanswer on”

Again make sure they are owned and exacutable by asterisk. I found that even with autoanswer=no in the oss.conf file I sitll had to have the two scripts to make it work right.

Now edit extensions_custom.conf and below [from-internal-custom] place the following

exten => *51,1,system(/etc/asterisk/ringoff.txt)
exten => *51,2,Dial(console/dsp,20,A(beep))
exten => *51,3,Hangup()

This will allow you to dial *51 and page through the external speaker. You can make it any number you wish. I found *51 was the same as our old nortel phones.

Now in FreePBX create a custom extension and give it a number similar to your shop phone. My shop phone has the extension of 10128 so my external speaker I gave it 11128. You can use any numbers you like. In the custom dial string for the new extension put console/dsp.

Now create a ring group that includes your shop phone and your external speaker. Hit the redbar and now you need to reboot your entire system. Make sure you have your sound card installed or you have turned it on in the bios if it is on the motherboard.

Once the system is back up and running go into your extensions_additional.conf file. You may have to do this through webmin or on the command line as extensions_additional.conf is one of the files you should not muck with. Scroll down till you find [ext-group] and copy everything from [ext-group] down to ; end of [ext-group]. Now paste this into extensions_override_freepbx.conf. So you will have this

[ext-group]
include => ext-group-custom
exten => 600,1,Macro(user-callerid,)
exten => 600,n,GotoIf($[“foo${BLKVM_OVERRIDE}” = “foo”]?skipdb)
exten => 600,n,GotoIf($["${DB(${BLKVM_OVERRIDE})}" = “TRUE”]?skipov)
exten => 600,n(skipdb),Set(__NODEST=)
exten => 600,n,Set(__BLKVM_OVERRIDE=BLKVM/${EXTEN}/${CHANNEL})
exten => 600,n,Set(__BLKVM_BASE=${EXTEN})
exten => 600,n,Set(DB(${BLKVM_OVERRIDE})=TRUE)
exten => 600,n(skipov),Set(RRNODEST=${NODEST})
exten => 600,n(skipvmblk),Set(__NODEST=${EXTEN})
exten => 600,n,Set(RecordMethod=Group)
exten => 600,n,Macro(record-enable,8000,${RecordMethod})
exten => 600,n,Set(RingGroupMethod=ringall)
exten => 600,n(DIALGRP),Macro(dial,60,${DIAL_OPTIONS},10128-11128)
exten => 600,n,Set(RingGroupMethod=)
exten => 600,n,GotoIf($[“foo${RRNODEST}” != “foo”]?nodest)
exten => 600,n,Set(__NODEST=)
exten => 600,n,dbDel(${BLKVM_OVERRIDE})
exten => 600,n,Hangup
exten => 600,n(nodest),Noop(SKIPPING DEST, CALL CAME FROM Q/RG: ${RRNODEST})

; end of [ext-group]

Now just above this line

exten => 600,n(DIALGRP),Macro(dial,60,${DIAL_OPTIONS},10128-11128)

stick in this line

exten => 600,n,system(/etc/asterisk/ringoff.txt)

Now save and reload asterisk to pickup the changes.

Now plug a set of speakers into your box and dial *51 you should hear a beep then you should hear yourself speaking. Hang up and dial your ring group 600 and you should hear the speaker ringing along with your shop phone.

Most sound cards need some resistance to them and I have found that the external speakers will not work unless you cut one speaker off of your $10.00 set of speakers and then hook that upto the external speaker. Looks cheesy but it works. I know you can purchase amplifiers that go with most phone systems. I have found them to be quite expensive. As well if they are not close to your asterisk box a direct line from the sound card to the amplifier will not work as there is not enough load. I do think that you can purchase a small amp from the source for about 20 bucks and it is the size of a 10 pack of cds. It should work and look a bit more professional. I’m going out today to see if I can find one and will post if it works.

So that sums it up. You should now be able to page and ring over the same speaker. If two people try at the same time it should busy out for the second person.

Rob