Hi Asterisk hackers,<br>
I have this situation. I have a music player
application X, with is writing to my /dev/dsp. I wanted to send this
music from /dev/dsp to a caller who dials to asterisk. <br>
So to get the voice data from /dev/dsp. I have
written my own device hacking functions and loaded with LD_PRELOAD
before starting application X. Now I could get the data which was
written to /dev/dsp. The native format of this data is AFMT_S16_LE
(Signed 16 bit Little Endian) at 48000 Hz sampling speed.<br>
In order for asterisk to communicate with this, I
have created a new channel called MYCHAN and when ever an asterisk user
is calling to a particular extension, i am reaquesting a Dial to my new
channel. This also happens fine. In my scenario the external caller is
a SIP caller with his native codec as ulaw.<br>
But my problem is, I have chosen AST_FORMAT_SLINEAR
as my native format for my new channel MYCHAN. This takes care of
AFMT_S16_LE to ulaw conversion and vice versa automatically since
asterisk core checks for the native types of channels before bridging
them. But inside asterisk AST_FORMAT_SLINEAR is assumed to be of 8000Hz
sampled voice. But in my case it is 48000Hz sampled.<br>
<br>
Is there any native format for my channel with
will take care of this. some format like signed 16 bit little endian
but at 48000 Hz ?<br>
or<br>
Will i be able to take care of this during conversion to ulaw, by hacking some code ?<br>
or <br>
is there any other way to send my AFMT_S16_LE at 48KHz over ulaw from asterisk ?<br>
<br>
Thanks for you help in advance.<br>
<br>
Regards,<br>
voipwala<br>