From 9b58d1a3daa400ce4b0003e812449695acd8c56f Mon Sep 17 00:00:00 2001 From: kmoore Date: Wed, 20 Jul 2011 19:03:17 +0000 Subject: [PATCH] Merged revisions 328936 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/2.0 ................ r328936 | kmoore | 2011-07-20 14:01:37 -0500 (Wed, 20 Jul 2011) | 15 lines Merged revisions 328935 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) | 8 lines Inband DTMF regression The functionality of inband DTMF in chan_sip relied upon ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF, never inband. This fixes the regression introduced in revision 328823. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328937 f38db490-d61c-443f-a65b-d21fe96a405b --- channels/chan_sip.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 0e5d27f79..a0b9cb037 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -6536,11 +6536,7 @@ static int sip_senddigit_begin(struct ast_channel *ast, char digit) sip_pvt_lock(p); switch (ast_test_flag(&p->flags[0], SIP_DTMF)) { case SIP_DTMF_INBAND: - if (p->rtp && ast_rtp_instance_dtmf_mode_get(p->rtp) == AST_RTP_DTMF_MODE_INBAND) { - ast_rtp_instance_dtmf_begin(p->rtp, digit); - } else { - res = -1; /* Tell Asterisk to generate inband indications */ - } + res = -1; /* Tell Asterisk to generate inband indications */ break; case SIP_DTMF_RFC2833: if (p->rtp) @@ -6572,11 +6568,7 @@ static int sip_senddigit_end(struct ast_channel *ast, char digit, unsigned int d ast_rtp_instance_dtmf_end_with_duration(p->rtp, digit, duration); break; case SIP_DTMF_INBAND: - if (p->rtp && ast_rtp_instance_dtmf_mode_get(p->rtp) == AST_RTP_DTMF_MODE_INBAND) { - ast_rtp_instance_dtmf_end(p->rtp, digit); - } else { - res = -1; /* Tell Asterisk to stop inband indications */ - } + res = -1; /* Tell Asterisk to stop inband indications */ break; } sip_pvt_unlock(p);