[asterisk-commits] russell: trunk r52953 - in /trunk: ./ channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: russell
Date: Tue Jan 30 13:36:28 2007
New Revision: 52953

URL: http://svn.digium.com/view/asterisk?view=rev&rev=52953
Log:
Merged revisions 52952 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r52952 | russell | 2007-01-30 13:33:12 -0600 (Tue, 30 Jan 2007) | 5 lines

Only set the DTMF flag on the rtp structure if the DTMF mode is actually
RFC2833, not just that it is not INFO.  This makes it get set for inband DTMF
as well, which is not valid.
(issue #8936)

........

Modified:
    trunk/   (props changed)
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=52953&r1=52952&r2=52953
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Jan 30 13:36:28 2007
@@ -2756,7 +2756,7 @@
 	do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE );
 
 	if (dialog->rtp) {
-		ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
+		ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
 		ast_rtp_setdtmfcompensate(dialog->rtp, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
 		ast_rtp_set_rtptimeout(dialog->rtp, peer->rtptimeout);
 		ast_rtp_set_rtpholdtimeout(dialog->rtp, peer->rtpholdtimeout);
@@ -4341,7 +4341,7 @@
 			free(p);
 			return NULL;
 		}
-		ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
+		ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
 		ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
 		ast_rtp_settos(p->rtp, global_tos_audio);
 		ast_rtp_set_rtptimeout(p->rtp, global_rtptimeout);
@@ -13556,7 +13556,7 @@
 		build_contact(p);			/* Build our contact header */
 
 		if (p->rtp) {
-			ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
+			ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
 			ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
 		}