similar to: No music on hold?

Hi, Does any one have any pointers on improving moh quality?? Symptoms are crackling and hissing as the sound comes and goes.. I installed mpg123 this morning.. I have tried various MP3's sampled at 160k, 128k, 32k and 8k and they all sounded terrible... The PC is a P4 so its got plenty of processing power.. I have tried a few different types of classical music (Piano, Violin and full

Hi Guys, I am having trouble trying to get format_mp3 working to play music on hold. I have followed the instructions in the read-me and the wiki however it seems after un-installing mpg123, asterisk is not even attempting to play MOH. My musiconhold.conf is ; Music on hold class definitions ; [classes] [moh_files] default = >/var/lib/asterisk/moh-native ;default =>

Hello list, after the last time that mpg123 wen ballistic on our production system, we decided to skip mp3 playback altogether and to go for raw files. After half an hour playing with mpg123 and sox parameters in order to translate a mp3 file to a wav file that can be streamed back through * with no need for an mp3 decoder, I thought I'd post the result to the list to avoid wasting

Hello, Using asterisk-1.2.1 I am trying to convert my music-on-hold files from .wav to alaw: % sox moh.wav -r 8000 -c 1 moh.al resample -ql The file sounds fine when listened with: % sox mox.al -t ossdsp /dev/dsp But when listened through asterisk with an alaw SIP phone the sound is clicky and too fast. Did I forget something in my conversion command? -- ldm@apartia.fr

We're setting up asterisk at the office (really doing some testing right now) and it is going to be hosted on a dual G5 XServe running OS X. We're an apple certified solutions provider, etc. so we want to build all our stuff on apple hardware and software. Anyway, the last sticking point is moh and meetme. Is there any solution to get moh and meetme working on OS X? Meetme

I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines connected via TE405P. Everything works great, except MOH. I added an exten with MusicOnHold(30), and it plays just fine. Conferences have music when no one is in. I have SIP phones. When I place a call on hold, the CLI give no indication the call is on hold. I have set musiconhold(default) everywhere, removed it from everywhere,

Does anybody have a good source for hold music? I can see a number of companies on the web that sell royalty-free MOH, but they don't all provide samples. The customer service desk has requested "calming, not sleeping, but calming" and "this is a high-tech company, so make it 'techie' [sic]". Thanks, --Ernest

I swear I read somewhere at sometime a command line that someone put forth that used mpg123 and sox to normalize the MP3 for MoH to a constant bitrate, etc... Through my search again for this information I can't find it. Can someone tell me if I am crazy and if so... does anyone have an "off the top of their head" solution to getting the bitrate constant for MoH? Thanks, Mike

I've noticed that native music on hold volume seems to be very loud sometimes. Is there anyway to turn this down? I know when using mpg123 I can set "quietmp3" but what about when using native?

Dear List, I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable attended transfer feature. but i just can't do it work. I've already set "atxfer = *" (and many other combinations) and all extensions on extensions.conf have the t and T option. But when I'm going to test, it doesn't work. Is there any other file that i have to configure in order to

Type 'mpg123' at the Linux CL. (no quotes) If the version is anything other than 59r, you problem is solved. Go to the Wiki and search for Music On Hold. Do the install of version 59r ONLY as described in the docs. Cheers, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Neil A. Hillard

Thanks Joshua and everyone, Joshua's solution seems a lot simpler and works well. Only one thing now - The reason I named the classes as I did, was so that I could select the class based on callerID plus extension. Unless I've misread it, I'm limited to 9 switchable classes via the "digit=#" option, is that correct? Or is there a clever hack around this? extensions.conf

I have followed the instructions provided at: http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.conf including installing asterisk-addons-1.2. I have left musiconhold.conf as is, calm-river et al are fine for now. However, no sound is heard and I get this message from the CLI when accessing MOH: -- Started music on hold, class 'default', on channel

I am having a problem getting music on hold working one of my servers. I have had this working on a PII 400 just fine but decided to upgrade my Asterisk server to a PIV 1.5ghz. I have installed mpg123 which seems to be working fine but when I start *, I get the following error message at the CLI prompt when I start *: WARNING[81931]: File chan_oss.c, Line 232 (sound_thread): Read error on

Good day all I added the music on hold entry in vpb.conf and commented out default line in musiconhold.conf. Asterisk starts up with the default mp3 but as soon as I remove it and add my mp3 it just doenst start up and gives a broken pipe error? Please Help or advice Thanks ALtus

Hello, I am having problems with music on hold on grandstream phones. When I press Hold button on grandstream phone this is the debug of sip. But nothing happens, no music. Is it problem of asterisk or grandstream budget phone? Sip read: INVITE sip:1105@192.168.1.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK7fcd3a44e7721b41 From:

OK. So I did a test last night. All of asterisk's threads where using 0.0% CPU. I made 1 call to our call queue. CPU jumped to average of 9% and stayed around that for the 2 minutes I was in the queue just listening to music on hold. MOH is in MP3 format and I'm using format_mp3. Phone was linksys PAP2-NA using G729. Can I reasonably assume that the 9% was decoding the MP3, then

Hi there; I didn't see any "G" option in the example above, and the usage for the option parameters is entirely undocumented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial The G options are as below G - If the call is answered, transfer the calling party to the specified priority and the called party to the specified priority plus one. context exten

Been using mpg123 for moh for the last two years or so. However, when I have * config errors, often times get a endless stream of console messages and need to kill the two mpg123 processes. Is there an alternative to mpg123 that eliminates that issue? I see references in musiconhold.conf relative to madplay, native file format, asterisk-addons, etc. Not sure why the asterisk-addon approach

All, Is there a way of rejecting a call using SIP in the Asterisk Dialplan? Essentially, I want to look at the called number and if it matches something I don't like I want to send back a SIP response which will not cause the other end to 'hunt'. The response codes that will achieve this are: 401 Unauthorized 403 Forbidden Is there a way of getting Asterisk to send back such a