I’m running asterisk-10.3.0 on a centos 6.2 server.
I’ve managed to create custom voice prompts for voicemenu’s but I wondered if better quality was available. The best I’ve done so far is record hq audio and convert it down to 16bit pcm wave files at 8k. The recordings are clean but a bit woolly.
Would v10.3.0 benefit from the asterisk add-on’s? (ie mp3’s) or are the add-ons only there for earlier versions? Are there other ways to get higher quality files to play?
I’ve searched on google but can’t find any way yet to improve on the sound files I’ve made so fer. Thanks for any pointers.
For G.711, GSM and many other common codecs, you should either record directly at 16 bit 8kHz, or drastically oversample and use a good interpolator. I believe recent versions of Windows tend to do the latter, anyway, although I don’t know how good their intepolation is.
G.711, which is the standard PSTN codec, can be converted losslessly to 16 bit, 8kHz, so there is no point in storing in any better format for the PSTN and for most direct IP phones.
GSM is considerably lower quality than G.711.
I don’t believe that normal rate G.729 is intended to any better than 16 bit, 8kHz, for speech (and will be considerably worse for music).
Basically, telephony codecs are not intended for high quality audio; they are intended for technically efficient transmission of human speech.
I would have thought this was a Support, not a General, topic.
Asterisk 10 can deal with all kinds of 16-bit signed linear audio files: 8, 12, 16, 24, 32, 44.1, 48, 96, and 192kHz sampling rates.
The “best” sound you’re going to get out of Asterisk right now, for transcoded stuffs, is 32kHz Speex, 32kHz Siren14, or 24kHz SILK. Find a client that supports one of those. 