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
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@ -6536,11 +6536,7 @@ static int sip_senddigit_begin(struct ast_channel *ast, char digit)
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sip_pvt_lock(p);
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switch (ast_test_flag(&p->flags[0], SIP_DTMF)) {
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case SIP_DTMF_INBAND:
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if (p->rtp && ast_rtp_instance_dtmf_mode_get(p->rtp) == AST_RTP_DTMF_MODE_INBAND) {
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ast_rtp_instance_dtmf_begin(p->rtp, digit);
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} else {
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res = -1; /* Tell Asterisk to generate inband indications */
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}
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res = -1; /* Tell Asterisk to generate inband indications */
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break;
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case SIP_DTMF_RFC2833:
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if (p->rtp)
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@ -6572,11 +6568,7 @@ static int sip_senddigit_end(struct ast_channel *ast, char digit, unsigned int d
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ast_rtp_instance_dtmf_end_with_duration(p->rtp, digit, duration);
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break;
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case SIP_DTMF_INBAND:
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if (p->rtp && ast_rtp_instance_dtmf_mode_get(p->rtp) == AST_RTP_DTMF_MODE_INBAND) {
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ast_rtp_instance_dtmf_end(p->rtp, digit);
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} else {
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res = -1; /* Tell Asterisk to stop inband indications */
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}
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res = -1; /* Tell Asterisk to stop inband indications */
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break;
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}
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sip_pvt_unlock(p);
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