[asterisk-commits] file: branch 1.6.1 r230883 - in /branches/1.6.1: ./ channels/ configs/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Nov 23 09:47:56 CST 2009
Author: file
Date: Mon Nov 23 09:47:54 2009
New Revision: 230883
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=230883
Log:
Merged revisions 230881 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r230881 | file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines
Change fax detection in chan_sip so it behaves as one would expect.
Internally the way T.38 is negotiated has changed and the option no longer
reflects a behavior that is valid. It will now look for a CNG tone on
received calls and if present send the call to the 'fax' extension. It is
then up to the application or channel to request the switch over to T.38.
........
Modified:
branches/1.6.1/ (props changed)
branches/1.6.1/channels/chan_sip.c
branches/1.6.1/configs/sip.conf.sample
Propchange: branches/1.6.1/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.1/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.1/channels/chan_sip.c?view=diff&rev=230883&r1=230882&r2=230883
==============================================================================
--- branches/1.6.1/channels/chan_sip.c (original)
+++ branches/1.6.1/channels/chan_sip.c Mon Nov 23 09:47:54 2009
@@ -1091,6 +1091,7 @@
#define SIP_PAGE2_RFC2833_COMPENSATE (1 << 25) /*!< DP: Compensate for buggy RFC2833 implementations */
#define SIP_PAGE2_BUGGY_MWI (1 << 26) /*!< DP: Buggy CISCO MWI fix */
#define SIP_PAGE2_DIALOG_ESTABLISHED (1 << 27) /*!< 29: Has a dialog been established? */
+#define SIP_PAGE2_FAX_DETECT (1 << 28) /*!< DP: Fax Detection support */
#define SIP_PAGE2_REGISTERTRYING (1 << 29) /*!< DP: Send 100 Trying on REGISTER attempts */
#define SIP_PAGE2_UDPTL_DESTINATION (1 << 30) /*!< DP: Use source IP of RTP as destination if NAT is enabled */
#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS (1 << 31) /*!< DP: Always set up video, even if endpoints don't support it */
@@ -1099,7 +1100,7 @@
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \
SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | \
SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | \
- SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_CONSTANT_SSRC)
+ SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_CONSTANT_SSRC | SIP_PAGE2_FAX_DETECT)
/*@}*/
@@ -1364,7 +1365,7 @@
int laststate; /*!< SUBSCRIBE: Last known extension state */
int dialogver; /*!< SUBSCRIBE: Version for subscription dialog-info */
- struct ast_dsp *vad; /*!< Inband DTMF Detection dsp */
+ struct ast_dsp *dsp; /*!< A DSP for inband DTMF and fax CNG tone detection */
struct sip_peer *relatedpeer; /*!< If this dialog is related to a peer, which one
Used in peerpoke, mwi subscriptions */
@@ -5607,8 +5608,8 @@
append_history(p, needcancel ? "Cancel" : "Hangup", "Cause %s", p->owner ? ast_cause2str(p->hangupcause) : "Unknown");
/* Disconnect */
- if (p->vad)
- ast_dsp_free(p->vad);
+ if (p->dsp)
+ ast_dsp_free(p->dsp);
p->owner = NULL;
ast->tech_pvt = dialog_unref(ast->tech_pvt, "unref ast->tech_pvt");
@@ -6250,12 +6251,19 @@
else
ast_debug(3, "This channel will not be able to handle video.\n");
- if ((ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) || (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
- i->vad = ast_dsp_new();
- ast_dsp_set_features(i->vad, DSP_FEATURE_DIGIT_DETECT);
+ if ((ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) || (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) ||
+ (ast_test_flag(&i->flags[1], SIP_PAGE2_FAX_DETECT))) {
+ int features = DSP_FEATURE_DIGIT_DETECT;
+
+ if (ast_test_flag(&i->flags[1], SIP_PAGE2_FAX_DETECT)) {
+ features |= DSP_FEATURE_FAX_DETECT;
+ }
+
+ i->dsp = ast_dsp_new();
+ ast_dsp_set_features(i->dsp, features);
if (global_relaxdtmf)
- ast_dsp_set_digitmode(i->vad, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
- }
+ ast_dsp_set_digitmode(i->dsp, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
+ }
/* Set file descriptors for audio, video, realtime text and UDPTL as needed */
if (i->rtp) {
@@ -6566,12 +6574,20 @@
ast_set_write_format(p->owner, p->owner->writeformat);
}
- if (f && (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) && p->vad) {
- f = ast_dsp_process(p->owner, p->vad, f);
- if (f && f->frametype == AST_FRAME_DTMF) {
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) && f->subclass == 'f') {
- ast_debug(1, "Fax CNG detected on %s\n", ast->name);
- *faxdetect = 1;
+ if (f && ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) || ast_test_flag(&p->flags[1], SIP_PAGE2_FAX_DETECT)) && p->dsp) {
+ f = ast_dsp_process(p->owner, p->dsp, f);
+ if (f && f->frametype == AST_FRAME_DTMF) {
+ if (f->subclass == 'f') {
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Fax CNG detected on %s\n", ast->name);
+ *faxdetect = 1;
+ /* If we only needed this DSP for fax detection purposes we can just drop it now */
+ if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) {
+ ast_dsp_set_features(p->dsp, DSP_FEATURE_DIGIT_DETECT);
+ } else {
+ ast_dsp_free(p->dsp);
+ p->dsp = NULL;
+ }
} else {
ast_debug(1, "* Detected inband DTMF '%c'\n", f->subclass);
}
@@ -6592,20 +6608,26 @@
fr = sip_rtp_read(ast, p, &faxdetected);
p->lastrtprx = time(NULL);
- /* If we are NOT bridged to another channel, and we have detected fax tone we issue T38 re-invite to a peer */
- /* If we are bridged then it is the responsibility of the SIP device to issue T38 re-invite if it detects CNG or fax preamble */
- if (faxdetected && ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) && (p->t38.state == T38_DISABLED) && !(ast_bridged_channel(ast))) {
- if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
- if (!p->pendinginvite) {
- ast_debug(3, "Sending reinvite on SIP (%s) for T.38 negotiation.\n", ast->name);
- change_t38_state(p, T38_LOCAL_REINVITE);
- transmit_reinvite_with_sdp(p, TRUE, FALSE);
- }
- } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
- ast_debug(3, "Deferring reinvite on SIP (%s) - it will be re-negotiated for T.38\n", ast->name);
- ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
- }
- }
+ /* If we detect a CNG tone and fax detection is enabled then send us off to the fax extension */
+ if (faxdetected && ast_test_flag(&p->flags[1], SIP_PAGE2_FAX_DETECT)) {
+ ast_channel_lock(ast);
+ if (strcmp(ast->exten, "fax")) {
+ const char *target_context = S_OR(ast->macrocontext, ast->context);
+ ast_channel_unlock(ast);
+ if (ast_exists_extension(ast, target_context, "fax", 1, ast->cid.cid_num)) {
+ ast_verbose(VERBOSE_PREFIX_2 "Redirecting '%s' to fax extension\n", ast->name);
+ pbx_builtin_setvar_helper(ast, "FAXEXTEN", ast->exten);
+ if (ast_async_goto(ast, target_context, "fax", 1)) {
+ ast_log(LOG_NOTICE, "Failed to async goto '%s' into fax of '%s'\n", ast->name, target_context);
+ }
+ fr = &ast_null_frame;
+ } else {
+ ast_log(LOG_NOTICE, "Fax detected but no fax extension\n");
+ }
+ } else {
+ ast_channel_unlock(ast);
+ }
+ }
/* Only allow audio through if they sent progress with SDP, or if the channel is actually answered */
if (fr && fr->frametype == AST_FRAME_VOICE && p->invitestate != INV_EARLY_MEDIA && ast->_state != AST_STATE_UP) {
@@ -22330,6 +22352,9 @@
} else if (!strcasecmp(v->name, "constantssrc")) {
ast_set_flag(&mask[1], SIP_PAGE2_CONSTANT_SSRC);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_CONSTANT_SSRC);
+ } else if (!strcasecmp(v->name, "faxdetect")) {
+ ast_set_flag(&mask[1], SIP_PAGE2_FAX_DETECT);
+ ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_FAX_DETECT);
} else
res = 0;
@@ -23358,6 +23383,7 @@
ast_clear_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_VIDEOSUPPORT_ALWAYS);
ast_clear_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT);
ast_clear_flag(&global_flags[1], SIP_PAGE2_IGNORESDPVERSION);
+ ast_clear_flag(&global_flags[1], SIP_PAGE2_FAX_DETECT);
/* Read the [general] config section of sip.conf (or from realtime config) */
@@ -24251,14 +24277,14 @@
if (p->rtp)
ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) {
- if (!p->vad) {
- p->vad = ast_dsp_new();
- ast_dsp_set_features(p->vad, DSP_FEATURE_DIGIT_DETECT);
+ if (!p->dsp) {
+ p->dsp = ast_dsp_new();
+ ast_dsp_set_features(p->dsp, DSP_FEATURE_DIGIT_DETECT);
}
} else {
- if (p->vad) {
- ast_dsp_free(p->vad);
- p->vad = NULL;
+ if (p->dsp) {
+ ast_dsp_free(p->dsp);
+ p->dsp = NULL;
}
}
sip_pvt_unlock(p);
Modified: branches/1.6.1/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.1/configs/sip.conf.sample?view=diff&rev=230883&r1=230882&r2=230883
==============================================================================
--- branches/1.6.1/configs/sip.conf.sample (original)
+++ branches/1.6.1/configs/sip.conf.sample Mon Nov 23 09:47:54 2009
@@ -460,7 +460,7 @@
; ; send 400 byte T.38 FAX packets to it.
;
; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists)
-; after T.38 is successfully negotiated.
+; when a CNG tone is detected on an incoming call.
;
; faxdetect = yes ; Default false
;
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