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authorrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2007-01-30 19:36:28 +0000
committerrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2007-01-30 19:36:28 +0000
commit6776231953a6d1e4791a326220a71ac94fc949b6 (patch)
tree850a76120d60ceaf5f59f3e3778b786c4c478023
parentfb998cea26d3302b134dd109b5e324f22b19b66a (diff)
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) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@52953 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--channels/chan_sip.c6
1 files changed, 3 insertions, 3 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index d3c611df8..c1478a53c 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -2756,7 +2756,7 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
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 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
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 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
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));
}