diff options
| -rw-r--r-- | apps/app_disa.c | 7 | ||||
| -rw-r--r-- | apps/app_playback.c | 6 | ||||
| -rw-r--r-- | channels/chan_sip.c | 13 | ||||
| -rw-r--r-- | configs/sip.conf.sample | 6 | ||||
| -rw-r--r-- | main/pbx.c | 2 |
5 files changed, 29 insertions, 5 deletions
diff --git a/apps/app_disa.c b/apps/app_disa.c index 3c8bad9d4..038274057 100644 --- a/apps/app_disa.c +++ b/apps/app_disa.c @@ -176,7 +176,12 @@ static int disa_exec(struct ast_channel *chan, void *data) /* answer */ ast_answer(chan); } - } else special_noanswer = 1; + } else { + special_noanswer = 1; + if (chan->_state != AST_STATE_UP) { + ast_indicate(chan, AST_CONTROL_PROGRESS); + } + } i = k = x = 0; /* k is 0 for pswd entry, 1 for ext entry */ did_ignore = 0; exten[0] = 0; diff --git a/apps/app_playback.c b/apps/app_playback.c index 6763bd917..972435f18 100644 --- a/apps/app_playback.c +++ b/apps/app_playback.c @@ -418,9 +418,13 @@ static int playback_exec(struct ast_channel *chan, void *data) if (option_skip) { /* At the user's option, skip if the line is not up */ goto done; - } else if (!option_noanswer) + } else if (!option_noanswer) { /* Otherwise answer unless we're supposed to send this while on-hook */ res = ast_answer(chan); + } else { + ast_indicate(chan, AST_CONTROL_PROGRESS); + } + } if (!res) { char *back = args.filenames; diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 684bff098..30243e45a 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -539,6 +539,7 @@ static struct ast_codec_pref default_prefs; /*!< Default codec prefs */ /* Global settings only apply to the channel */ static int global_directrtpsetup; /*!< Enable support for Direct RTP setup (no re-invites) */ +static int global_prematuremediafilter; /*!< Enable/disable premature frames in a call (causing 183 early media) */ static int global_limitonpeers; /*!< Match call limit on peers only */ static int global_rtautoclear; static int global_notifyringing; /*!< Send notifications on ringing */ @@ -3894,9 +3895,11 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame) !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) { ast_rtp_new_source(p->rtp); - p->invitestate = INV_EARLY_MEDIA; - transmit_provisional_response(p, "183 Session Progress", &p->initreq, 1); - ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT); + if (!global_prematuremediafilter) { + p->invitestate = INV_EARLY_MEDIA; + transmit_provisional_response(p, "183 Session Progress", &p->initreq, 1); + ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT); + } } else if (p->t38.state == T38_ENABLED && !p->t38.direct) { p->t38.state = T38_DISABLED; transmit_reinvite_with_sdp(p); @@ -11317,6 +11320,7 @@ static int sip_show_settings(int fd, int argc, char *argv[]) print_codec_to_cli(fd, &default_prefs); ast_cli(fd, "\n"); ast_cli(fd, " T1 minimum: %d\n", global_t1min); + ast_cli(fd, " No premature media: %s\n", global_prematuremediafilter ? "Yes" : "No"); ast_cli(fd, " Relax DTMF: %s\n", global_relaxdtmf ? "Yes" : "No"); ast_cli(fd, " Compact SIP headers: %s\n", compactheaders ? "Yes" : "No"); ast_cli(fd, " RTP Keepalive: %d %s\n", global_rtpkeepalive, global_rtpkeepalive ? "" : "(Disabled)" ); @@ -18127,6 +18131,7 @@ static int reload_config(enum channelreloadreason reason) expiry = DEFAULT_EXPIRY; global_notifyringing = DEFAULT_NOTIFYRINGING; global_limitonpeers = FALSE; + global_prematuremediafilter = FALSE; global_directrtpsetup = FALSE; /* Experimental feature, disabled by default */ global_notifyhold = FALSE; global_alwaysauthreject = 0; @@ -18232,6 +18237,8 @@ static int reload_config(enum channelreloadreason reason) ast_set2_flag(&global_flags[1], i || ast_true(v->value), SIP_PAGE2_RTAUTOCLEAR); } else if (!strcasecmp(v->name, "usereqphone")) { ast_set2_flag(&global_flags[0], ast_true(v->value), SIP_USEREQPHONE); + } else if (!strcasecmp(v->name, "prematuremedia")) { + global_prematuremediafilter = ast_true(v->value); } else if (!strcasecmp(v->name, "relaxdtmf")) { global_relaxdtmf = ast_true(v->value); } else if (!strcasecmp(v->name, "checkmwi")) { diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 966d9abf5..741ae7570 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -115,6 +115,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; use 'never' to never use in-band signalling, even in cases ; where some buggy devices might not render it ; Valid values: yes, no, never Default: never +;prematuremedia=no ; Some ISDN links send empty media frames before + ; the call is in ringing or progress state. The SIP + ; channel will then send 183 indicating early media + ; which will be empty - thus users get no ring signal. + ; Setting this to "no" will stop any media before we have + ; call progress. Default is "yes". ;useragent=Asterisk PBX ; Allows you to change the user agent string ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address ; Note that promiscredir when redirects are made to the diff --git a/main/pbx.c b/main/pbx.c index 1672b5332..acf2da9cf 100644 --- a/main/pbx.c +++ b/main/pbx.c @@ -5710,6 +5710,8 @@ static int pbx_builtin_background(struct ast_channel *chan, void *data) } else if (!ast_test_flag(&flags, BACKGROUND_NOANSWER)) { res = ast_answer(chan); } + /* Send progress control frame to start early media */ + ast_indicate(chan, AST_CONTROL_PROGRESS); } if (!res) { |