RTP Payload 96, need to modify source code and recompile

Hi:

My itsp uses a payload of 96 for the DTMF, what causes DTMF is not working in the incoming SIP calls, as you know, the providers not always are able (or want) to change this, so ive read that this can ve solved modifying the rtp.c and adding the following

1) Unknown RTP codec 96 received:

If you are getting messages such as the one below, chances are big you are trying to connect to a Cirpack/v4.39a (gw_sip) sip gateway or to a Mediatrix 1124 gateway

Dec 24 10:57:40 NOTICE[5570]: rtp.c:564 ast_rtp_read: Unknown RTP codec 96 received

This is because those 2 devices for some reason decide to send DTMF as a dynamic payload type 96, and asterisk expects the DTMF to have number 101 or 121.

We can easily fix this with a one line patch to rtp.c

Look for:

[34] = {1, AST_FORMAT_H263},
[103] = {1, AST_FORMAT_H263_PLUS},
[97] = {1, AST_FORMAT_ILBC},
[101] = {0, AST_RTP_DTMF},
[110] = {1, AST_FORMAT_SPEEX},

and change this to:

[34] = {1, AST_FORMAT_H263},
[103] = {1, AST_FORMAT_H263_PLUS},
[96] = {0, AST_RTP_DTMF},
[97] = {1, AST_FORMAT_ILBC},
[101] = {0, AST_RTP_DTMF},
[110] = {1, AST_FORMAT_SPEEX},

ive already installed the asterisk-devel package but it only contains the headers, not the source code, can anyone please provide a hint about how to do what i need to ?

Thanks