So, I’ve got an asterisk 1.2 system in stable production and a development/new production 1.4 system in the testing phases.
As has been previously noted, the format_mp3 native player starts the hold music from the beginning each time MoH is needed. Has anyone found a fix for this? I also would like to get away from mpg123. I do use mp3 files for my MoH.
If anyone could post their working/stable solutions for MoH which does not use mpg123 (or has a very new version of mpg123 working properly), it would be greatly appreciated.
It is possible to run format_mp3 without mpg123, however i would recommend using native formats instead. It would take no additional resources to be used while playing MoH. Just convert your music to all possible formats you use, and put correct files in MoH dir with .ulaw / .alaw / .gsm extensions and asterisk will grab most appropriate.
Thanks for the reponse atis. I may have been unclear in my original post. While yes, format_mp3 works just fine on my 1.4 test system, it starts the music on hold from scratch over and over again. Ie, you hear the first 10 seconds fo a song, hang up, call back, you get the first 10 seconds again.
My real question was if anyone had a fix for this. I’m really just looking for the best quality MoH setup that people have found for 1.4 thus far.
Well, i was answering secondary question. I’m not sure about restarting from beginning, but i think using native formats (using 1.2) don’t have such problems (i can check tomorrow at work). Also the sound quality from statically encoded audio is higher than recoding it on the fly, so you might give it a try.
Thanks for the response! I’d love to re-sample everything and keep using format_mp3/native, but the way the audio continually re-starts is driving me crazy! If I can get that sorted so that it uses the “pause” type functionality of mpg123, I’m golden.
Hey X - I’m having the EXACT same issue with my MOH and 1.4. I call in, listen for a few seconds of my classical music, hang up, call again - same music. I dont know why its a problem, the same files worked just fine with the same configuration on Asterisk 1.2.
It frustrates the crap out of me, but no one seems to know anything about it. If you find a fix, please post it, I would love to get this fixed myself.
(and I really don’t like the idea of re-sampling everything because thats just a pain in my ass)
I honestly don’t mind resampling. One side-effect of my business is that I happen to have a pretty decent recording/radio voice. I’ve found that different codecs for conversions and sampling have amazing effects on sound quality. I’m totally for re-sampling all the music to 8000 using proper conversion techniques. I have no doubt that my callers would have a much better experience this way than the on-the-fly re-coding of the current music on hold.
(check out wavepad, free .wav editing tools, great for re-sampling.)
Anyway, it’s late, I’ve had a few beers. I’d like to move away from mpg123. 0.59r is terribly old and the most recent development still does not seem to work properly with asterisk. There’s got to be up-to-date solutions that actually WORK.