Specifically, I need to be able to take 2 H.323 channels, each running a G.723.1 codec, and bridge them together.
However, before I do that, I need to play a message and then listen to one of the channels to determine how to route the call.
For example, it may play a menu asking the user to select one of technical support, sales, or accounting. Or it might ask them to press 1 for Mandarin and route the call to Singapore, 2 for Khmer and and route to Phenom Phen, etc. Since I can program the gateways which will be interfacing to the PSTN to send the DTMF tones via H.245 out of band, there should be no technical reason why this won't work...in other words, I never have to actually decode the G.723.1 stream. The messages can be stored in G.723,1 format already, so I never have to encode either.
However, I have not been able to discern whether Asterisk will work in this mode or not. Can someone who has actually implemented an Asterisk system using G.723.1 in pass thru enlighten me? The documentation on this is very confusing. Will it use the out of band H.245 messages to detect a DTMF tone?
Since there is no technical reason it should not work, if the answer is "it doesn't work" can someone give me a hint on what must be changed to make it functional? I would certainly be willing to put in a little effort to make this functional if it isn't already and someone can give me some instructions on where to begin.
Thanks in advance for any assistance,
Chris
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