Greetings fellow listers,
I have an application where I have
approximately 300 files that I playback individually or in blocks to
simulate "text-to-speech" in a "less mechanical" voice than normal Allison
files provide. These files are presently in GSM format and sound pretty
good when I play them on my computer speakers or on my in-house Polycom
501's over SIP connections. The "problem" I have is that the intended use
of the application is going to be over SIP/DAHDI trunks that will connect to
VM's over IAX trunks. What is your best suggestion for maintaining the
quality of the audio as much as possible?
Best Case presently - SIP phone in-house to IAX
Worst Case presently - Cell phone calls Asterisk 1 on TDM400P which connects
to VM Asterisk 2 via IAX.
Asterisk version is 1.4.30
Thanks in Advance
Danny Nicholas
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