diff options
| author | kmoore <kmoore@f38db490-d61c-443f-a65b-d21fe96a405b> | 2011-07-19 18:07:22 +0000 |
|---|---|---|
| committer | kmoore <kmoore@f38db490-d61c-443f-a65b-d21fe96a405b> | 2011-07-19 18:07:22 +0000 |
| commit | 4b03897207848035c13ad20f8f69b3c264f0c448 (patch) | |
| tree | 6b0c0bf9d0548cfd085d9a8e3741f4de71fb4f79 /res | |
| parent | b7854890e0bf54421cc02e811299f85a6cd27143 (diff) | |
Merged revisions 328824 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10
................
r328824 | kmoore | 2011-07-19 13:05:21 -0500 (Tue, 19 Jul 2011) | 18 lines
Merged revisions 328823 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | 11 lines
RTP bridge away with inband DTMF and feature detection
When deciding whether Asterisk was allowed to bridge the call away from the
core, chan_sip did not take into account the usage of features on dialed
channels that require monitoring of DTMF on channels utilizing inband DTMF.
This would cause Asterisk to allow the call to be locally or remotely bridged,
preventing access to the data required to detect activations of such features.
(closes 17237)
Review: https://reviewboard.asterisk.org/r/1302/
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328825 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'res')
| -rw-r--r-- | res/res_rtp_asterisk.c | 18 |
1 files changed, 18 insertions, 0 deletions
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c index a520cdb16..06af4832d 100644 --- a/res/res_rtp_asterisk.c +++ b/res/res_rtp_asterisk.c @@ -147,6 +147,7 @@ struct ast_rtp { unsigned int dtmf_duration; /*!< Total duration in samples since the digit start event */ unsigned int dtmf_timeout; /*!< When this timestamp is reached we consider END frame lost and forcibly abort digit */ unsigned int dtmfsamples; + enum ast_rtp_dtmf_mode dtmfmode;/*!< The current DTMF mode of the RTP stream */ /* DTMF Transmission Variables */ unsigned int lastdigitts; char sending_digit; /*!< boolean - are we sending digits */ @@ -260,6 +261,8 @@ static int ast_rtp_destroy(struct ast_rtp_instance *instance); static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit); static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit); static int ast_rtp_dtmf_end_with_duration(struct ast_rtp_instance *instance, char digit, unsigned int duration); +static int ast_rtp_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode); +static enum ast_rtp_dtmf_mode ast_rtp_dtmf_mode_get(struct ast_rtp_instance *instance); static void ast_rtp_update_source(struct ast_rtp_instance *instance); static void ast_rtp_change_source(struct ast_rtp_instance *instance); static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame); @@ -286,6 +289,8 @@ static struct ast_rtp_engine asterisk_rtp_engine = { .dtmf_begin = ast_rtp_dtmf_begin, .dtmf_end = ast_rtp_dtmf_end, .dtmf_end_with_duration = ast_rtp_dtmf_end_with_duration, + .dtmf_mode_set = ast_rtp_dtmf_mode_set, + .dtmf_mode_get = ast_rtp_dtmf_mode_get, .update_source = ast_rtp_update_source, .change_source = ast_rtp_change_source, .write = ast_rtp_write, @@ -534,6 +539,19 @@ static int ast_rtp_destroy(struct ast_rtp_instance *instance) return 0; } +static int ast_rtp_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode) +{ + struct ast_rtp *rtp = ast_rtp_instance_get_data(instance); + rtp->dtmfmode = dtmf_mode; + return 0; +} + +static enum ast_rtp_dtmf_mode ast_rtp_dtmf_mode_get(struct ast_rtp_instance *instance) +{ + struct ast_rtp *rtp = ast_rtp_instance_get_data(instance); + return rtp->dtmfmode; +} + static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit) { struct ast_rtp *rtp = ast_rtp_instance_get_data(instance); |