diff options
Diffstat (limited to 'net/asterisk14/files/rtp_force_dtmf-nocodecnego.diff')
| -rw-r--r-- | net/asterisk14/files/rtp_force_dtmf-nocodecnego.diff | 70 |
1 files changed, 0 insertions, 70 deletions
diff --git a/net/asterisk14/files/rtp_force_dtmf-nocodecnego.diff b/net/asterisk14/files/rtp_force_dtmf-nocodecnego.diff deleted file mode 100644 index 776422b34645..000000000000 --- a/net/asterisk14/files/rtp_force_dtmf-nocodecnego.diff +++ /dev/null @@ -1,70 +0,0 @@ ---- 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. |