diff options
Diffstat (limited to 'net/asterisk10/files/rtp_force_dtmf-nocodecnego.diff')
| -rw-r--r-- | net/asterisk10/files/rtp_force_dtmf-nocodecnego.diff | 69 |
1 files changed, 69 insertions, 0 deletions
diff --git a/net/asterisk10/files/rtp_force_dtmf-nocodecnego.diff b/net/asterisk10/files/rtp_force_dtmf-nocodecnego.diff new file mode 100644 index 000000000000..9dcb7d8b134f --- /dev/null +++ b/net/asterisk10/files/rtp_force_dtmf-nocodecnego.diff @@ -0,0 +1,69 @@ +--- channels/chan_sip.c.orig 2009-11-23 17:28:47.000000000 +0200 ++++ channels/chan_sip.c 2009-11-23 17:29:29.000000000 +0200 +@@ -565,6 +565,9 @@ + 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 */ +@@ -5593,6 +5596,12 @@ + + /* 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); ++ 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); +@@ -18366,6 +18375,9 @@ + + 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)); + +@@ -18428,6 +18440,18 @@ + } + } 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, "%30d", &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 2008-08-16 01:33:42.000000000 +0300 ++++ configs/sip.conf.sample 2008-12-12 17:03:11.000000000 +0200 +@@ -49,6 +49,12 @@ + ; 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/ip-tos.txt for a description of these parameters. + ;tos_sip=cs3 ; Sets TOS for SIP packets. + ;tos_audio=ef ; Sets TOS for RTP audio packets. |