mirror of
https://git.freebsd.org/ports.git
synced 2025-06-21 20:50:31 -04:00
285e2c278a
o move all additional functionality into separate patches and make it opt in. It has been concern of asterisks devs that the FreeBSD pacakage adds functionalty not present in the original version, whch could be confusing.
70 lines
3.4 KiB
Diff
70 lines
3.4 KiB
Diff
--- channels/chan_sip.c.orig 2008-01-31 21:52:49.000000000 +0200
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+++ channels/chan_sip.c 2008-03-14 17:50:57.000000000 +0200
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@@ -556,6 +556,9 @@
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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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@@ -5392,6 +5395,13 @@
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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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+ found_rtpmap_codecs[last_rtpmap_codec++] = 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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@@ -16833,6 +16843,9 @@
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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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@@ -16889,6 +16902,18 @@
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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, "%d", &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 2008-03-12 17:57:19.000000000 +0200
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+++ configs/sip.conf.sample 2008-03-12 18:13:03.000000000 +0200
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@@ -53,6 +53,12 @@
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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/README.tos 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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