diff options
| author | Russell Bryant <russell@russellbryant.com> | 2007-01-30 19:36:28 +0000 |
|---|---|---|
| committer | Russell Bryant <russell@russellbryant.com> | 2007-01-30 19:36:28 +0000 |
| commit | d11f8b7ccd97a6c7cb8ac32a4ebd0a49a059086b (patch) | |
| tree | 850a76120d60ceaf5f59f3e3778b786c4c478023 | |
| parent | 83e2ed50e6f0ddc1a8204870df2578c5d8c5fe3f (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: https://origsvn.digium.com/svn/asterisk/trunk@52953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
| -rw-r--r-- | channels/chan_sip.c | 6 |
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)); } |