sysmocom
/
asterisk
11
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity

Increase amount of time we send tones on RTP to 100ms 

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer 2004-03-14 23:12:22 +00:00
parent 0852f5a930
commit f997f9f139
1 changed files with 2 additions and 2 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

4
rtp.c
View File

@ -919,8 +919,8 @@ int ast_rtp_senddigit(struct ast_rtp *rtp, char digit)
if (x ==0) {
/* Clear marker bit and increment seqno */
rtpheader[0] = htonl((2 << 30) | (101 << 16) | (rtp->seqno++));
/* Make duration 240 */
rtpheader[3] |= htonl((240));
/* Make duration 800 (100ms) */
rtpheader[3] |= htonl((800));
/* Set the End bit for the last 3 */
rtpheader[3] |= htonl((1 << 23));
}