69 lines
3.4 KiB
Diff
69 lines
3.4 KiB
Diff
--- channels/chan_sip.c.orig 2009-11-23 17:28:47.000000000 +0200
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+++ channels/chan_sip.c 2009-11-23 17:29:29.000000000 +0200
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@@ -565,6 +565,9 @@
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static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */
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static unsigned int global_tos_audio; /*!< IP type of service for audio RTP packets */
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static unsigned int global_tos_video; /*!< IP type of service for video RTP packets */
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+static int global_force_dtmf_relay = 0;
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+static int global_force_dtmf_relay_pt = 101;
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+
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static int compactheaders; /*!< send compact sip headers */
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static int recordhistory; /*!< Record SIP history. Off by default */
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static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
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@@ -5593,6 +5596,12 @@
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/* Now gather all of the codecs that we are asked for: */
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ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
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+ /* Add telephone-event */
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+ if (global_force_dtmf_relay && !(peernoncodeccapability & AST_RTP_DTMF)) {
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+ ast_rtp_set_m_type(newaudiortp, global_force_dtmf_relay_pt);
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+ ast_rtp_set_rtpmap_type(newaudiortp, global_force_dtmf_relay_pt, "audio", "telephone-event", 0);
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+ peernoncodeccapability |= AST_RTP_DTMF;
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+ }
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ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);
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newjointcapability = p->capability & (peercapability | vpeercapability);
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@@ -18366,6 +18375,9 @@
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global_matchexterniplocally = FALSE;
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+ global_force_dtmf_relay = 0;
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+ global_force_dtmf_relay_pt = 101;
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+
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/* Copy the default jb config over global_jbconf */
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memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
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@@ -18428,6 +18440,18 @@
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}
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} else if (!strcasecmp(v->name, "vmexten")) {
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ast_copy_string(default_vmexten, v->value, sizeof(default_vmexten));
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+ } else if (!strcasecmp(v->name, "rtp_force_dtmf_relay")) {
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+ if ((global_force_dtmf_relay = ast_true(v->value)))
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+ ast_verbose("RTP DTMF relaying will be enforced\n");
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+ else
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+ ast_verbose("RTP DTMF relaying will not be enforced\n");
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+ } else if (!strcasecmp(v->name, "rtp_force_dtmf_relay_pt")) {
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+ sscanf(v->value, "%d", &global_force_dtmf_relay_pt);
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+ if (global_force_dtmf_relay_pt < 96 || global_force_dtmf_relay_pt > 255) {
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+ ast_verbose("RTP forced DTMF relay payload type is not valid: %d. Using default (101)\n", global_force_dtmf_relay_pt);
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+ global_force_dtmf_relay_pt = 101;
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+ } else
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+ ast_log(LOG_WARNING, "RTP forced DTMF relay payload type is %d\n", global_force_dtmf_relay_pt);
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} else if (!strcasecmp(v->name, "rtptimeout")) {
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if ((sscanf(v->value, "%30d", &global_rtptimeout) != 1) || (global_rtptimeout < 0)) {
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ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno);
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--- configs/sip.conf.sample.orig 2008-08-16 01:33:42.000000000 +0300
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+++ configs/sip.conf.sample 2008-12-12 17:03:11.000000000 +0200
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@@ -49,6 +49,12 @@
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; and multiline formatted headers for strict
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; SIP compatibility (defaults to "no")
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+;rtp_force_dtmf_relay=no ; Enable RFC2833 DTMFs to be sent even if peer
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+ ; hasn't announced support for it. Default: no
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+
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+;rtp_force_dtmf_relay_pt=101 ; RTP payload type value for enforced RFC2833
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+ ; DTMFs. Default: 101
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+
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; See doc/ip-tos.txt for a description of these parameters.
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;tos_sip=cs3 ; Sets TOS for SIP packets.
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;tos_audio=ef ; Sets TOS for RTP audio packets.
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