similar to: Music-on-Hold radio input?

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

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 have searching the forums here on how to get Music On Hold working and I have been able to get * to accept a command for MusicOnHold and for Meetme after loading the ztdummy module. I used the default config for /etc/zaptel.conf since I saw no guidance on this. My problem now is that when I activate MusicOnHold, the sample music file sounds very slow and distorted. My best guess is that it is

I'm curious if anyone has used a radio for MOH? If so, how did you set it up? I have a client who is interested in using a radio for the music on hold, since that is what they did with their old phone system. Thanks, Leif Madsen.

Does anyone know how to maximize music on hold quality on calls inbound from PSTN? I know that it is common to have choppy and static sounding music on hold when connecting via PSTN but how can that be minimized? I assume that the bitrates, type of music, etc can minimize the effects. Does anyone have any experience in this area? Do you know where I should look for more information?

Thanks for the reply. We are currently using AMBE (4Kbps) for our Traffic/Weather Channels. If you have ever had a chance to hear the service, you will know that AMBE does not do us well. I understand that 2Kbps is low quality, but any poorer than AMBE? If can get a decent quality for other low bandwidth talk channels, such as about 10-16Kbps and have it sound rather clean, then I would be

OS: CentOS 5.0 x86. Hi, I have some old music tapes and a walkman, and want to move the music to mp3 format. So far I am thinking to use Audacity to record the music to wavs, and then do not know, somehow I will compress the wavs to mp3s as vbr 192 Kbps with something. Has anyone done this? What app did you use for recording the music from the tapes and what app did you use to make it mp3?

Hello folks. I understand this is the development email address, but I don't have any others to use, maybe you can help. I'm an engineer at XM Satellite Radio in Washington, DC. I'm scouting for other low bandwidth algorithms for some voice content. Speex looks pretty interesting. My goal is to find a promising codec for 2Kbps. I see the download links on the pages offer

I have just setup music on hold by downloading and installing mpg123 r Now I have music on hold but it sounds terrible - clipping, buzzing, digital distortion, and its too loud (which probably isn't helping) and I'm just running it thru the 'default' line in music onhold.conf line default => quietmp3:/var/lib/asterisk/mohmp3, with the default mp3s. This is a standard 1.0.3

Hi, all. I'm running the Asterisk 1.0.7 package that comes with Debian Sarge. I'm new to Asterisk, but so far almost everything is going well. However, I'm having a Hell of a time with Music On Hold. I've got an extension set up to test Music On Hold, as follows: exten => 3000,1,WaitMusicOnHold(300) I've tried the following in musiconhold.conf: default =>

07.03.2015 0:24, Kevin Harwell ?????: > On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com > <mailto:serov.d.p at gmail.com>> wrote: > > Hello. > > Asterisk 13.2. > I transfer configs from chan_sip to res_pjsip. > In chan_sip i have "match_auth_username=yes" and have nothing in > pjsip. > > I have a

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

Hello, I am attempting to use the LARTC traffic control to simulate a radio link that has variable bandwidth and availability. The basic bandwidth could be as low as 500 bits/sec but will generally be about 4000 bits/sec. If the simulated radio link is unavailable (zero bandwidth) then packets should be queued until a link is re-established. i.e. Initial bandwidth is 5000bits/sec then to 0

Hi, I have a bunch of old radio programs (mystery/drama shows, not music) encoded at 32 kbit (and some 48kbit) mp3 (mono). I want to reencode them in ogg and make them available over gnutella. My question is this. What is the best quality level (-q) for transcoding them. I want to preserve quality, but I want to be sensitive to the many modem based gnutella users. I also want to to

Hi, I've recently installed Asterisk and have got the majority of it configured (what an excellent piece of software it is, too), but I'm having a couple of problems. The first one is with music on hold! I've downloaded and installed mpg123 as specified: ># whereis mpg123 >mpg123: /usr/local/bin/mpg123 It's the correct version: >#

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

This mail depends upon the fact that I don't have a couple of good earphones ;-) I read in the site that q=6 is a very high quality, but does it contain perceivable differencies from the original? (for 95% of people, of course). I also found q=6 to produce files slightly bigger (1/10 bigger) than those produced with lame VBR q=2 (about 192 bps on average). I always thought LAME VBR q=2

Hi, guys. I''m so happy to join this group. I''m new to ROR. And I haven''t learned or used any other program langueage before. I''m intrest in develop web-based database-driven prgramming. My company need to develop a bussiness management system recently. I wanna use ROR for a try. So, where should I start? In another words, what books or resources should I read?

Hi, > You can try the Ogg DirectShow filter to get Windows support for Speex: > http://www.illiminable.com/ogg/ if you wait a little longer, I am trying to provide speex support for the ffmpeg project (www.ffmpeg.org). It does encode/decode to/from avi and wav, a support for ogg has not been done, and my question is, should it be necessary? I don't really like ogg in first place..

Hello. Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have "match_auth_username=yes" and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk]