[asterisk-dev] DTMF Volume

[Konstantin Bokarius]

When using the SendDTMF I send *84085551212 in order to transfer calls on a
SIP channel with RFC2833 DMTF mode.  The issue is that the DTMFs getting
passed are at a very high volume when heard by the transferee before the
transfer actually initiates.  Furthermore I need to space the DTMFs out by
at least 300-400ms in order for it to work consistently.

 

What I am looking to do is decrease or eliminate the volume at which these
DTMFs are heard.  

 

I have narrowed down this function in indications.c:

 

int ast_playtones_start(struct ast_channel *chan, int vol, const char
*playlst, int interruptible)

{

        char *s, *data = ast_strdupa(playlst); /* cute */

        struct playtones_def d = { vol, -1, 0, 1, NULL};

        char *stringp;

        char *separator;

 

        if (vol < 1)

                d.vol = 7219; /* Default to -8db */

 

        d.interruptible = interruptible;

        .

 

Why is vol defaulted to 7219?  What is the significance of that number?

 

When I tried to change that value to a negative, low (1-10), or very high
number I did not notice any change with the DTMF volume.

 

Are there other values that need to be changed - or am I missing something? 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20080702/0123bdad/attachment.htm