[asterisk-commits] rizzo: branch 1.4 r44109 -
/branches/1.4/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Sun Oct 1 08:01:29 MST 2006
Author: rizzo
Date: Sun Oct 1 10:01:28 2006
New Revision: 44109
URL: http://svn.digium.com/view/asterisk?rev=44109&view=rev
Log:
sync with trunk - move variable declarations to the beginning of a block.
Modified:
branches/1.4/channels/chan_sip.c
Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?rev=44109&r1=44108&r2=44109&view=diff
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Sun Oct 1 10:01:28 2006
@@ -4541,6 +4541,9 @@
int numberofmediastreams = 0;
int debug = sip_debug_test_pvt(p);
+ int found_rtpmap_codecs[32];
+ int last_rtpmap_codec=0;
+
if (!p->rtp) {
ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
return -1;
@@ -4712,8 +4715,6 @@
*/
/* XXX This needs to be done per media stream, since it's media stream specific */
iterator = req->sdp_start;
- int found_rtpmap_codecs[32];
- int last_rtpmap_codec=0;
while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */
if (option_debug > 1) {
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