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--- channels/chan_sip.c.orig 2008-01-31 21:52:49.000000000 +0200
+++ channels/chan_sip.c 2008-03-14 17:50:57.000000000 +0200
@@ -556,6 +556,9 @@
static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */
static unsigned int global_tos_audio; /*!< IP type of service for audio RTP packets */
static unsigned int global_tos_video; /*!< IP type of service for video RTP packets */
+static int global_force_dtmf_relay = 0;
+static int global_force_dtmf_relay_pt = 101;
+
static int compactheaders; /*!< send compact sip headers */
static int recordhistory; /*!< Record SIP history. Off by default */
static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
@@ -5392,6 +5395,13 @@
/* Now gather all of the codecs that we are asked for: */
ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
+ /* Add telephone-event */
+ if (global_force_dtmf_relay && !(peernoncodeccapability & AST_RTP_DTMF)) {
+ ast_rtp_set_m_type(newaudiortp, global_force_dtmf_relay_pt);
+ found_rtpmap_codecs[last_rtpmap_codec++] = global_force_dtmf_relay_pt;
+ ast_rtp_set_rtpmap_type(newaudiortp, global_force_dtmf_relay_pt, "audio", "telephone-event", 0);
+ peernoncodeccapability |= AST_RTP_DTMF;
+ }
ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);
newjointcapability = p->capability & (peercapability | vpeercapability);
@@ -16833,6 +16843,9 @@
global_matchexterniplocally = FALSE;
+ global_force_dtmf_relay = 0;
+ global_force_dtmf_relay_pt = 101;
+
/* Copy the default jb config over global_jbconf */
memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
@@ -16889,6 +16902,18 @@
}
} else if (!strcasecmp(v->name, "vmexten")) {
ast_copy_string(default_vmexten, v->value, sizeof(default_vmexten));
+ } else if (!strcasecmp(v->name, "rtp_force_dtmf_relay")) {
+ if ((global_force_dtmf_relay = ast_true(v->value)))
+ ast_verbose("RTP DTMF relaying will be enforced\n");
+ else
+ ast_verbose("RTP DTMF relaying will not be enforced\n");
+ } else if (!strcasecmp(v->name, "rtp_force_dtmf_relay_pt")) {
+ sscanf(v->value, "%d", &global_force_dtmf_relay_pt);
+ if (global_force_dtmf_relay_pt < 96 || global_force_dtmf_relay_pt > 255) {
+ ast_verbose("RTP forced DTMF relay payload type is not valid: %d. Using default (101)\n", global_force_dtmf_relay_pt);
+ global_force_dtmf_relay_pt = 101;
+ } else
+ ast_log(LOG_WARNING, "RTP forced DTMF relay payload type is %d\n", global_force_dtmf_relay_pt);
} else if (!strcasecmp(v->name, "rtptimeout")) {
if ((sscanf(v->value, "%d", &global_rtptimeout) != 1) || (global_rtptimeout < 0)) {
ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno);
--- configs/sip.conf.sample.orig 2008-03-12 17:57:19.000000000 +0200
+++ configs/sip.conf.sample 2008-03-12 18:13:03.000000000 +0200
@@ -53,6 +53,12 @@
; and multiline formatted headers for strict
; SIP compatibility (defaults to "no")
+;rtp_force_dtmf_relay=no ; Enable RFC2833 DTMFs to be sent even if peer
+ ; hasn't announced support for it. Default: no
+
+;rtp_force_dtmf_relay_pt=101 ; RTP payload type value for enforced RFC2833
+ ; DTMFs. Default: 101
+
; See doc/README.tos for a description of these parameters.
;tos_sip=cs3 ; Sets TOS for SIP packets.
;tos_audio=ef ; Sets TOS for RTP audio packets.
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