We are having issues with the quality of our custom IVR recordings and the system recording included with asterisk.
Real time conversations have excellent voice quality, but the system recordings sound a little robotic and choppy. It isn’t sever enough to not be able to understand the message but it doesn’t sound nearly as good as the real time voice. I should also mention that the quality is better when listening via an IP phone but still not perfect. When using a cell or landline phone there is a noticable degradation in quality.
Is there any adjustments that can be made to improve the playback quality of the system recordings?
are you using the default system recordings, or custom (ie self-made) recordings? if you recorded them yourself, what format did you record in?
what codecs are you using locally? if you’re connecting to the PSTN via SIP or IAX (and not direct connect T1’s), what codecs are you using there?
I have similiar problems with both. I used the freepbx interface to setup an extension to record the custom messages. The extension was a CIsco 7940 using SIP and 711
what codecs are you using locally? if you’re connecting to the PSTN via SIP or IAX (and not direct connect T1’s), what codecs are you using there?[/quote]
I am connecting the the pstn using an IAX trunk with 711 codec.
i’m not sure how freepbx records stuff, but i’d guess either GSM or RAW.
RAW would be good, GSM, not so much.
if you’re interested, Kris over at AstLinux has re-recorded all of the main prompts in native formats, so there isn’t any transcoding required for most of your codecs - you MIGHT try dropping some of those in and see if it helps - they’re supposed to make a pretty big difference.
thanks, I’ll give them a try and report back
Here’s a stupid question.
Which format should I be using. I am in the US, using 711 codecs exclusively. I assume I should try ulaw?
Is that correct?