g723 message playback issues

I’m currently using Asterisk as a call router, and using AudioCodes Mediant and FXS to handle calls. The AudioCodes boxes support g723 at various frame sizes (not just the 30 msec standard). I’m using the supposedly open-source g723 codec on Asterisk, solely for encoding and presenting the routing messages to the caller (“please enter the extension you wish to call”, etc).

However, the codec is fixed at 30 msec frame size, as specified by the ITU spec. We need to be able to use the 60 msec and 90 msec frame sizes which are supported by the AudioCodes boxes, without losing all the messages from Asterisk. What options do I have for accomplishing this??

It seems to me that I will have to find some external program that will let me manually record these messages at the desired format, and then somehow tell Ast to just play back the files without doing any encoding; can I do this at all??

If so, how??

I have the TFoT book, so if this is discussed in that somewhere, I would welcome references to that book (I did some preliminary searching and haven’t found anything yet), otherwise, any other pointers would be VERY welcome!!

If I cannot pre-record the messages (and thus avoid requiring the g723 codec in Ast in the first place), what other choices do I have?? Are there any other g723 codecs which do support the larger frame sizes??

As I recall, frame sizes are one of *'s weak points. Using most sip codecs it can only accept 20ms frames as it uses them for its internal timing (although I think there is some work happening to remove this dependency…)

Well, that’s pretty sad. Does anyone know of any programs that would allow me to record my own messages in G.723.1 with different frame sizes??