similar to: buffer-issue when piping live-streams into musiconhold

I'm getting a FLOOD of these types of messages on my MAC OS X box: Sep 9 14:46:31 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:37 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:37 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:43

Does anyone have a good solution to stream hold music to the Asterisk/FreePBX server? I currently have setup WinAMP using ShoutCast but it appears to a very touch and go stream, being that it seems to periodically drop the connection. Asterisk 1.6, FreePBX 2.8, WinAMP v5.6 Chris Davis Enterprise Administrator Barkley Court Reporters | Barkley Trial Technologies Global Deposition Services (310)

Hello, I need to play some musiconhold content starting at a random duration from the start. Thanks to mode=custom option and either madplay or mpg123 programs, I could successfully get what I was after on a Debian Wheezy system. Now I realized sox version on my target system (Debian Squeeze) cannot convert to MP3 format. So I'm looking after workarounds. 0. I've read many mpg123 or

Hi, I follow the web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf - mohstream.sh , to configure music on hold to play using mms but failed. Anyone can play using mms? ango

Hi, I don't seem to be able to get music on hold to play normally. The sound gets often interrupted with a few seconds of silence then starts playing again. I'm using mpg123-0.59r and tried mp3 files with different sample rates with no luck. If that matters, endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes

Hi everybody, I'm receiving the message "res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?!" in asterisk console when I try to put a call on hold. I don't the reason and I'm sure the relative module is loaded. In musiconhold.conf I put these lines, trying something I found in some previous post: ; ; Music on hold class definitions ; [classes]

On Wed, Mar 25, 2015 at 09:43:20AM -0400, Steve Matzura wrote: > I ridded myself of the 32-bit and install the 64-bit version of > EZStream on Fedora 20 (Heizenbug). The RPM doesn't offer 0.6.0, but no > matter, I'm sorry, the previous ezstream RPM package maintainer did not registered the package for monitoring new releases, so Fedora still delivers older version. However I

Hello all, This is my first post to this list that I have just discovered. I am using icecast2 with ices2 and really it is great. I am using it also with ezstream. Now I would like to use icecast with asterisk for the music on hold. I read all I found about this but without success. Is there anyway to use icecast and asterisk in the way that this works? Is there someone used asterisk and

Hi. I am having the following warning when using music on hold. It works from X-Lite to Grandstream. I get a lot of errors and warnings. 1.Warning, flexibel rate not heavily tested! 2. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread: Request to schedule in the past?!?! Thanks for any help. Full Output below: Jan 30 10:24:55 WARNING[1133718080]: chan_sip.c:486

So here's the story: I thought I would explore further features of the ezstream, including its ability to reencode various input types (e.g. ogg, mp3, flac) on the fly into one type of stream (ezstream_reencode_mp3.xml config). It works wonderfully except for one problem: the decoding of mp3s results in a steady stream of noise (i.e. the source is not distinguishable among the noise). Using

Trying to make the fax detection work. My current setup (with no fax) is done without Answer(), so the call is answered only when someone actually picks-up the phone. But when the incoming call is fax, I can her the tone and call is never forwarded to "Fax" extension. But... Strange thing happens when I (mistakenly) put a call on hold: -- Executing [youngandson-test at

Hi all, A long time ago I built an Asterisk system that plays IceCast streams via moh. extensions.conf: Exten => moh,1,Set(SIP_CODEC=ulaw) Exten => moh,2,Answer Exten => moh,3,MusicONHold(test_new) Exten => moh,4,Hangup musiconhold.conf ; test_new [test_new] mode=custom application=/etc/mystreams/test_new.sh test_new.sh #!/bin/bash wget -q -T 120 -O -

I am using asterisk-oh323-0.7.2-pre and CVS Head of Asterisk. Oh323 Module compiled without errors. But When I try to stary Asterisk with the Oh323.so file in the modules folder, Asterisk is dying with the following error. [chan_oh323.so]Aug 2 14:08:14 NOTICE[18873]: res_musiconhold.c:490 monmp3thread: Request to schedule in the past?!?! => (InAccess Networks OpenH323 Channel Driver) ==

On Tue, 9 Feb 2016, Larry Turnbull wrote: > One of the streams is an old time radio stream and I use ezstream to run the > prerecorded shows. > > Is there a package or some sort of way I can apply dynamic compression to > the stream? Ezstream really isn't designed for this. It's primarily meant for streaming files as they are, which is a very light-weight operation, with

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

On Wed, 25 Mar 2015, Steve Matzura wrote: > I've encountered something new having to do with something > called 'madplay'. EZStream is what it sounds like. It's a very basic streamer. You can use it to stream files as they are, at the bitrate they are, etc. For anything more complicated, you will need decoders and encoders. This includes being able to stream multiple

Thanks Geoff, I will give it a try. I am reencoding all files to a 128K mp3, 44100 sample rate stereo stream for consistency. So yes I have the decode and encode lines in the xml file as you outlined. I will check out sox and see how it does. I am also exploring mp3gain to run across the entire library and see if getting all files to the same level will help. Larry -----Original

Hey guys. Im trying to setup Music on Hold. If I transfer a call (with dial) I like to put the call on Music on hold.. Here's what I've tried so far: On my I extensions.conf exten =>1,1,WaitMusicOnHold(30) exten =>1,2,Dial(SIP/mateo,18) exten =>1,3,VoiceMail(1001) I have also added this line to [context].. So it looks like that: ;[context] musiconhold=default Additinaly,

Hello! I'm using Asterisk for our office telephony, but we have some problems that still we can't resolve about it. Here they are: 1) merge in/out call recording files I also tried to use a script I found on the internet, called 2wav2mp3 In extensions.conf I added the following lines ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 exten

Hi all, I am running Asterisk in conjunction with a Sip proxy. Asterisk is registered to an external SIP carrier (sip.uni.it) If a call reachs Asterisk through the SIP carrier, then it is forwarded to the external SIP proxy extension (530 at weboffice.dyndns.org), when the extension 530 that has answered the call tries to transfer the call to another extension (513 at