Merged revisions 53138 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r53138 | file | 2007-02-03 15:05:02 -0600 (Sat, 03 Feb 2007) | 2 lines

Make SIPDtmfMode application work with recent capability changes, and also fix an RTP stack issue when the auto option was used. (issue #8972 reported by mdu113)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53139 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index d0d3dd0d27df3f6645124f9f564ae5c852b39c2a..2360f7a2a102503a83bca18f2a2232ec74c0e2e7 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5263,6 +5263,9 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
                if (newnoncodeccapability & AST_RTP_DTMF) {
                        /* XXX Would it be reasonable to drop the DSP at this point? XXX */
                        ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
+                       /* Since RFC2833 is now negotiated we need to change some properties of the RTP stream */
+                       ast_rtp_setdtmf(p->rtp, 1);
+                       ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
                } else {
                        ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
                }
@@ -17120,14 +17123,19 @@ static int sip_dtmfmode(struct ast_channel *chan, void *data)
        if (!strcasecmp(mode,"info")) {
                ast_clear_flag(&p->flags[0], SIP_DTMF);
                ast_set_flag(&p->flags[0], SIP_DTMF_INFO);
+               p->jointnoncodeccapability &= ~AST_RTP_DTMF;
        } else if (!strcasecmp(mode,"rfc2833")) {
                ast_clear_flag(&p->flags[0], SIP_DTMF);
                ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
+               p->jointnoncodeccapability |= AST_RTP_DTMF;
        } else if (!strcasecmp(mode,"inband")) { 
                ast_clear_flag(&p->flags[0], SIP_DTMF);
                ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
+               p->jointnoncodeccapability &= ~AST_RTP_DTMF;
        } else
                ast_log(LOG_WARNING, "I don't know about this dtmf mode: %s\n",mode);
+       if (p->rtp)
+               ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
        if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) {
                if (!p->vad) {
                        p->vad = ast_dsp_new();