[asterisk-commits] pcadach: trunk r43597 - in /trunk: channels/
channels/h323/ configs/
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Mon Sep 25 02:03:15 MST 2006
Author: pcadach
Date: Mon Sep 25 04:03:14 2006
New Revision: 43597
URL: http://svn.digium.com/view/asterisk?rev=43597&view=rev
Log:
Support for negotiation and receiption of Cisco's RTP DTMF
Modified:
trunk/channels/chan_h323.c
trunk/channels/h323/ast_h323.cxx
trunk/channels/h323/ast_h323.h
trunk/channels/h323/caps_h323.cxx
trunk/channels/h323/chan_h323.h
trunk/configs/h323.conf.sample
Modified: trunk/channels/chan_h323.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_h323.c?rev=43597&r1=43596&r2=43597&view=diff
==============================================================================
--- trunk/channels/chan_h323.c (original)
+++ trunk/channels/chan_h323.c Mon Sep 25 04:03:14 2006
@@ -184,7 +184,7 @@
int peercapability; /* Capabilities learned from peer */
int jointcapability; /* Common capabilities for local and remote side */
struct ast_codec_pref peer_prefs; /* Preferenced list of codecs which remote side supports */
- int dtmf_pt; /* Payload code used for RFC2833 messages */
+ int dtmf_pt[2]; /* Payload code used for RFC2833/CISCO messages */
int curDTMF; /* DTMF tone being generated to Asterisk side */
int DTMFsched; /* Scheduler descriptor for DTMF */
int update_rtp_info; /* Configuration of fd's array is pending */
@@ -515,7 +515,9 @@
return -1;
}
ast_mutex_lock(&pvt->lock);
- if (pvt->rtp && (pvt->options.dtmfmode & H323_DTMF_RFC2833) && (pvt->dtmf_pt > 0)) {
+ if (pvt->rtp &&
+ (((pvt->options.dtmfmode & H323_DTMF_RFC2833) && pvt->dtmf_pt[0])
+ /*|| ((pvt->options.dtmfmode & H323_DTMF_CISCO) && pvt->dtmf_pt[1]))*/)) {
/* out-of-band DTMF */
if (h323debug) {
ast_log(LOG_DTMF, "Begin sending out-of-band digit %c on %s\n", digit, c->name);
@@ -554,7 +556,7 @@
return -1;
}
ast_mutex_lock(&pvt->lock);
- if (pvt->rtp && (pvt->options.dtmfmode & H323_DTMF_RFC2833) && (pvt->dtmf_pt > 0)) {
+ if (pvt->rtp && (pvt->options.dtmfmode & H323_DTMF_RFC2833) && ((pvt->dtmf_pt[0] > 0) || (pvt->dtmf_pt[0] > 0))) {
/* out-of-band DTMF */
if (h323debug) {
ast_log(LOG_DTMF, "End sending out-of-band digit %c on %s\n", digit, c->name);
@@ -644,7 +646,7 @@
pvt->outgoing = 1;
if (h323debug)
- ast_log(LOG_DEBUG, "Placing outgoing call to %s, %d\n", called_addr, pvt->options.dtmfcodec);
+ ast_log(LOG_DEBUG, "Placing outgoing call to %s, %d/%d\n", called_addr, pvt->options.dtmfcodec[0], pvt->options.dtmfcodec[1]);
ast_mutex_unlock(&pvt->lock);
res = h323_make_call(called_addr, &(pvt->cd), &pvt->options);
if (res) {
@@ -757,7 +759,7 @@
f = ast_rtp_read(pvt->rtp);
/* Don't send RFC2833 if we're not supposed to */
- if (f && (f->frametype == AST_FRAME_DTMF) && !(pvt->options.dtmfmode & H323_DTMF_RFC2833)) {
+ if (f && (f->frametype == AST_FRAME_DTMF) && !(pvt->options.dtmfmode & (H323_DTMF_RFC2833 | H323_DTMF_CISCO))) {
return &ast_null_frame;
}
if (pvt->owner) {
@@ -979,8 +981,10 @@
ast_log(LOG_DEBUG, "Setting NAT on RTP to %d\n", pvt->options.nat);
ast_rtp_setnat(pvt->rtp, pvt->options.nat);
- if (pvt->dtmf_pt > 0)
- ast_rtp_set_rtpmap_type(pvt->rtp, pvt->dtmf_pt, "audio", "telephone-event", 0);
+ if (pvt->dtmf_pt[0] > 0)
+ ast_rtp_set_rtpmap_type(pvt->rtp, pvt->dtmf_pt[0], "audio", "telephone-event", 0);
+ if (pvt->dtmf_pt[1] > 0)
+ ast_rtp_set_rtpmap_type(pvt->rtp, pvt->dtmf_pt[1], "audio", "cisco-telephone-event", 0);
if (pvt->peercapability)
ast_rtp_codec_setpref(pvt->rtp, &pvt->peer_prefs);
@@ -1121,7 +1125,7 @@
}
memcpy(&pvt->options, &global_options, sizeof(pvt->options));
pvt->jointcapability = pvt->options.capability;
- if (pvt->options.dtmfmode & H323_DTMF_RFC2833) {
+ if (pvt->options.dtmfmode & (H323_DTMF_RFC2833 | H323_DTMF_CISCO)) {
pvt->nonCodecCapability |= AST_RTP_DTMF;
} else {
pvt->nonCodecCapability &= ~AST_RTP_DTMF;
@@ -1263,18 +1267,28 @@
options->dtmfmode |= H323_DTMF_INBAND;
} else if (!strcasecmp(val, "rfc2833")) {
options->dtmfmode |= H323_DTMF_RFC2833;
- if (!opt)
- options->dtmfcodec = H323_DTMF_RFC2833_PT;
- else if ((tmp >= 96) && (tmp < 128))
- options->dtmfcodec = tmp;
- else {
- options->dtmfcodec = H323_DTMF_RFC2833_PT;
- ast_log(LOG_WARNING, "Unknown rfc2833 payload %s specified at line %d, using default %d\n", opt, v->lineno, options->dtmfcodec);
- }
+ if (!opt) {
+ options->dtmfcodec[0] = H323_DTMF_RFC2833_PT;
+ } else if ((tmp >= 96) && (tmp < 128)) {
+ options->dtmfcodec[0] = tmp;
+ } else {
+ options->dtmfcodec[0] = H323_DTMF_RFC2833_PT;
+ ast_log(LOG_WARNING, "Unknown rfc2833 payload %s specified at line %d, using default %d\n", opt, v->lineno, options->dtmfcodec[0]);
+ }
+ } else if (!strcasecmp(val, "cisco")) {
+ options->dtmfmode |= H323_DTMF_CISCO;
+ if (!opt) {
+ options->dtmfcodec[1] = H323_DTMF_CISCO_PT;
+ } else if ((tmp >= 96) && (tmp < 128)) {
+ options->dtmfcodec[1] = tmp;
+ } else {
+ options->dtmfcodec[1] = H323_DTMF_CISCO_PT;
+ ast_log(LOG_WARNING, "Unknown Cisco DTMF payload %s specified at line %d, using default %d\n", opt, v->lineno, options->dtmfcodec[1]);
+ }
+ } else if (!strcasecmp(v->value, "h245-signal")) {
+ options->dtmfmode |= H323_DTMF_SIGNAL;
} else {
- ast_log(LOG_WARNING, "Unknown dtmf mode '%s', using rfc2833\n", v->value);
- options->dtmfmode |= H323_DTMF_RFC2833;
- options->dtmfcodec = H323_DTMF_RFC2833_PT;
+ ast_log(LOG_WARNING, "Unknown dtmf mode '%s' at line %d\n", v->value, v->lineno);
}
} else if (!strcasecmp(v->name, "dtmfcodec")) {
ast_log(LOG_NOTICE, "Option %s at line %d is deprecated. Use dtmfmode=rfc2833[:<payload>] instead.\n", v->name, v->lineno);
@@ -1282,7 +1296,7 @@
if (tmp < 96)
ast_log(LOG_WARNING, "Invalid %s value %s at line %d\n", v->name, v->value, v->lineno);
else
- options->dtmfcodec = tmp;
+ options->dtmfcodec[0] = tmp;
} else if (!strcasecmp(v->name, "bridge")) {
options->bridge = ast_true(v->value);
} else if (!strcasecmp(v->name, "nat")) {
@@ -2367,21 +2381,21 @@
ast_mutex_unlock(&pvt->lock);
}
-static void set_dtmf_payload(unsigned call_reference, const char *token, int payload)
+static void set_dtmf_payload(unsigned call_reference, const char *token, int payload, int is_cisco)
{
struct oh323_pvt *pvt;
if (h323debug)
- ast_log(LOG_DEBUG, "Setting DTMF payload to %d on %s\n", payload, token);
+ ast_log(LOG_DEBUG, "Setting %s DTMF payload to %d on %s\n", (is_cisco ? "Cisco" : "RFC2833"), payload, token);
pvt = find_call_locked(call_reference, token);
if (!pvt) {
return;
}
if (pvt->rtp) {
- ast_rtp_set_rtpmap_type(pvt->rtp, payload, "audio", "telephone-event", 0);
- }
- pvt->dtmf_pt = payload;
+ ast_rtp_set_rtpmap_type(pvt->rtp, payload, "audio", (is_cisco ? "cisco-telephone-event" : "telephone-event"), 0);
+ }
+ pvt->dtmf_pt[is_cisco ? 1 : 0] = payload;
ast_mutex_unlock(&pvt->lock);
if (h323debug)
ast_log(LOG_DEBUG, "DTMF payload on %s set to %d\n", token, payload);
@@ -2750,7 +2764,8 @@
memset(&global_options, 0, sizeof(global_options));
global_options.fastStart = 1;
global_options.h245Tunneling = 1;
- global_options.dtmfcodec = H323_DTMF_RFC2833_PT;
+ global_options.dtmfcodec[0] = H323_DTMF_RFC2833_PT;
+ global_options.dtmfcodec[1] = H323_DTMF_CISCO_PT;
global_options.dtmfmode = 0;
global_options.capability = GLOBAL_CAPABILITY;
global_options.bridge = 1; /* Do native bridging by default */
Modified: trunk/channels/h323/ast_h323.cxx
URL: http://svn.digium.com/view/asterisk/trunk/channels/h323/ast_h323.cxx?rev=43597&r1=43596&r2=43597&view=diff
==============================================================================
--- trunk/channels/h323/ast_h323.cxx (original)
+++ trunk/channels/h323/ast_h323.cxx Mon Sep 25 04:03:14 2006
@@ -533,7 +533,7 @@
bridging = FALSE;
progressSetup = progressAlert = 0;
dtmfMode = 0;
- dtmfCodec = (RTP_DataFrame::PayloadTypes)0;
+ dtmfCodec[0] = dtmfCodec[1] = (RTP_DataFrame::PayloadTypes)0;
redirect_reason = -1;
#ifdef TUNNELLING
tunnelOptions = remoteTunnelOptions = 0;
@@ -664,7 +664,8 @@
progressSetup = opts->progress_setup;
progressAlert = opts->progress_alert;
- dtmfCodec = (RTP_DataFrame::PayloadTypes)opts->dtmfcodec;
+ dtmfCodec[0] = (RTP_DataFrame::PayloadTypes)opts->dtmfcodec[0];
+ dtmfCodec[1] = (RTP_DataFrame::PayloadTypes)opts->dtmfcodec[1];
dtmfMode = opts->dtmfmode;
if (isIncoming) {
@@ -1213,7 +1214,8 @@
void MyH323Connection::OnUserInputTone(char tone, unsigned duration, unsigned logicalChannel, unsigned rtpTimestamp)
{
- if ((dtmfMode & H323_DTMF_RFC2833)) {
+ /* Why we should check this? */
+ if ((dtmfMode & (H323_DTMF_CISCO | H323_DTMF_RFC2833 | H323_DTMF_SIGNAL)) != 0) {
if (h323debug) {
cout << "\t-- Received user input tone (" << tone << ") from remote" << endl;
}
@@ -1243,10 +1245,10 @@
H245_Capability & cap = entry.m_capability;
if (cap.GetTag() == H245_Capability::e_receiveRTPAudioTelephonyEventCapability) {
H245_AudioTelephonyEventCapability & atec = cap;
- atec.m_dynamicRTPPayloadType = dtmfCodec;
-// on_set_rfc2833_payload(GetCallReference(), (const char *)GetCallToken(), (int)dtmfCodec);
+ atec.m_dynamicRTPPayloadType = dtmfCodec[0];
+// on_set_rfc2833_payload(GetCallReference(), (const char *)GetCallToken(), (int)dtmfCodec[0]);
if (h323debug) {
- cout << "\t-- Transmitting RFC2833 on payload " <<
+ cout << "\t-- Receiving RFC2833 on payload " <<
atec.m_dynamicRTPPayloadType << endl;
}
}
@@ -1299,21 +1301,12 @@
};
#endif
struct ast_codec_pref prefs;
+ RTP_DataFrame::PayloadTypes pt;
if (!H323Connection::OnReceivedCapabilitySet(remoteCaps, muxCap, reject)) {
return FALSE;
}
- const H323Capability * cap = remoteCaps.FindCapability(H323_UserInputCapability::SubTypeNames[H323_UserInputCapability::SignalToneRFC2833]);
- if (cap != NULL) {
- RTP_DataFrame::PayloadTypes pt = ((H323_UserInputCapability*)cap)->GetPayloadType();
- on_set_rfc2833_payload(GetCallReference(), (const char *)GetCallToken(), (int)pt);
- if ((dtmfMode & H323_DTMF_RFC2833) && (sendUserInputMode == SendUserInputAsTone))
- sendUserInputMode = SendUserInputAsInlineRFC2833;
- if (h323debug) {
- cout << "\t-- Inbound RFC2833 on payload " << pt << endl;
- }
- }
memset(&prefs, 0, sizeof(prefs));
int peer_capabilities = 0;
for (int i = 0; i < remoteCapabilities.GetSize(); ++i) {
@@ -1346,6 +1339,32 @@
}
}
break;
+ case H323Capability::e_Data:
+ if (!strcmp((const char *)remoteCapabilities[i].GetFormatName(), CISCO_DTMF_RELAY)) {
+ pt = remoteCapabilities[i].GetPayloadType();
+ if ((dtmfMode & H323_DTMF_CISCO) != 0) {
+ on_set_rfc2833_payload(GetCallReference(), (const char *)GetCallToken(), (int)pt, 1);
+// if (sendUserInputMode == SendUserInputAsTone)
+// sendUserInputMode = SendUserInputAsInlineRFC2833;
+ }
+ if (h323debug) {
+ cout << "\t-- Outbound Cisco RTP DTMF on payload " << pt << endl;
+ }
+ }
+ break;
+ case H323Capability::e_UserInput:
+ if (!strcmp((const char *)remoteCapabilities[i].GetFormatName(), H323_UserInputCapability::SubTypeNames[H323_UserInputCapability::SignalToneRFC2833])) {
+ pt = remoteCapabilities[i].GetPayloadType();
+ if ((dtmfMode & H323_DTMF_RFC2833) != 0) {
+ on_set_rfc2833_payload(GetCallReference(), (const char *)GetCallToken(), (int)pt, 0);
+// if (sendUserInputMode == SendUserInputAsTone)
+// sendUserInputMode = SendUserInputAsInlineRFC2833;
+ }
+ if (h323debug) {
+ cout << "\t-- Outbound RFC2833 on payload " << pt << endl;
+ }
+ }
+ break;
#if 0
case H323Capability::e_Video:
for (int x = 0; vcodecs[x].asterisk_codec > 0; ++x) {
@@ -1416,7 +1435,7 @@
return connectionState != ShuttingDownConnection;
}
-void MyH323Connection::SetCapabilities(int cap, int dtmf_mode, void *_prefs, int pref_codec)
+void MyH323Connection::SetCapabilities(int caps, int dtmf_mode, void *_prefs, int pref_codec)
{
PINDEX lastcap = -1; /* last common capability index */
int alreadysent = 0;
@@ -1427,11 +1446,12 @@
struct ast_format_list format;
int frames_per_packet;
int max_frames_per_packet;
+ H323Capability *cap;
localCapabilities.RemoveAll();
if (h323debug) {
- cout << "Setting capabilities to " << ast_getformatname_multiple(caps_str, sizeof(caps_str), cap) << endl;
+ cout << "Setting capabilities to " << ast_getformatname_multiple(caps_str, sizeof(caps_str), caps) << endl;
ast_codec_pref_string(prefs, caps_str, sizeof(caps_str));
cout << "Capabilities in preference order is " << caps_str << endl;
}
@@ -1449,7 +1469,7 @@
y <<= 1;
codec = y;
}
- if (!(cap & codec) || (alreadysent & codec) || !(codec & AST_FORMAT_AUDIO_MASK))
+ if (!(caps & codec) || (alreadysent & codec) || !(codec & AST_FORMAT_AUDIO_MASK))
continue;
alreadysent |= codec;
format = ast_codec_pref_getsize(prefs, codec);
@@ -1518,23 +1538,64 @@
}
}
- lastcap++;
- lastcap = localCapabilities.SetCapability(0, lastcap, new H323_UserInputCapability(H323_UserInputCapability::HookFlashH245));
-
- lastcap++;
+ cap = new H323_UserInputCapability(H323_UserInputCapability::HookFlashH245);
+ if (cap && cap->IsUsable(*this)) {
+ lastcap++;
+ lastcap = localCapabilities.SetCapability(0, lastcap, cap);
+ } else if (cap)
+ delete cap; /* Capability is not usable */
+
dtmfMode = dtmf_mode;
- if ((dtmfMode & H323_DTMF_INBAND)) {
- localCapabilities.SetCapability(0, lastcap, new H323_UserInputCapability(H323_UserInputCapability::BasicString));
- sendUserInputMode = SendUserInputAsString;
- } else {
- lastcap = localCapabilities.SetCapability(0, lastcap, new H323_UserInputCapability(H323_UserInputCapability::SignalToneRFC2833));
- /* Cisco sends DTMF only through h245-alphanumeric or h245-signal, no support for RFC2833 */
- lastcap = localCapabilities.SetCapability(0, lastcap, new H323_UserInputCapability(H323_UserInputCapability::SignalToneH245));
- sendUserInputMode = SendUserInputAsTone; /* RFC2833 transmission handled at Asterisk level */
- }
-
- if (h323debug) {
- cout << "Allowed Codecs:\n\t" << setprecision(2) << localCapabilities << endl;
+ if (h323debug) {
+ cout << "DTMF mode is " << (int)dtmfMode << endl;
+ }
+ if (dtmfMode) {
+ lastcap++;
+ if (dtmfMode == H323_DTMF_INBAND) {
+ cap = new H323_UserInputCapability(H323_UserInputCapability::BasicString);
+ if (cap && cap->IsUsable(*this)) {
+ lastcap = localCapabilities.SetCapability(0, lastcap, cap);
+ } else if (cap)
+ delete cap; /* Capability is not usable */
+ sendUserInputMode = SendUserInputAsString;
+ } else {
+ if ((dtmfMode & H323_DTMF_RFC2833) != 0) {
+ cap = new H323_UserInputCapability(H323_UserInputCapability::SignalToneRFC2833);
+ if (cap && cap->IsUsable(*this))
+ lastcap = localCapabilities.SetCapability(0, lastcap, cap);
+ else {
+ dtmfMode |= H323_DTMF_SIGNAL;
+ if (cap)
+ delete cap; /* Capability is not usable */
+ }
+ }
+ if ((dtmfMode & H323_DTMF_CISCO) != 0) {
+ /* Try Cisco's RTP DTMF relay too, but prefer RFC2833 or h245-signal */
+ cap = new AST_CiscoDtmfCapability();
+ if (cap && cap->IsUsable(*this)) {
+ lastcap = localCapabilities.SetCapability(0, lastcap, cap);
+ /* We cannot send Cisco RTP DTMFs, use h245-signal instead */
+ dtmfMode |= H323_DTMF_SIGNAL;
+ } else {
+ dtmfMode |= H323_DTMF_SIGNAL;
+ if (cap)
+ delete cap; /* Capability is not usable */
+ }
+ }
+ if ((dtmfMode & H323_DTMF_SIGNAL) != 0) {
+ /* Cisco usually sends DTMF correctly only through h245-alphanumeric or h245-signal */
+ cap = new H323_UserInputCapability(H323_UserInputCapability::SignalToneH245);
+ if (cap && cap->IsUsable(*this))
+ lastcap = localCapabilities.SetCapability(0, lastcap, cap);
+ else if (cap)
+ delete cap; /* Capability is not usable */
+ }
+ sendUserInputMode = SendUserInputAsTone; /* RFC2833 transmission handled at Asterisk level */
+ }
+ }
+
+ if (h323debug) {
+ cout << "Allowed Codecs for " << GetCallToken() << " (" << GetSignallingChannel()->GetLocalAddress() << "):\n\t" << setprecision(2) << localCapabilities << endl;
}
}
Modified: trunk/channels/h323/ast_h323.h
URL: http://svn.digium.com/view/asterisk/trunk/channels/h323/ast_h323.h?rev=43597&r1=43596&r2=43597&view=diff
==============================================================================
--- trunk/channels/h323/ast_h323.h (original)
+++ trunk/channels/h323/ast_h323.h Mon Sep 25 04:03:14 2006
@@ -112,7 +112,7 @@
unsigned progressAlert;
int cause;
- RTP_DataFrame::PayloadTypes dtmfCodec;
+ RTP_DataFrame::PayloadTypes dtmfCodec[2];
int dtmfMode;
};
Modified: trunk/channels/h323/caps_h323.cxx
URL: http://svn.digium.com/view/asterisk/trunk/channels/h323/caps_h323.cxx?rev=43597&r1=43596&r2=43597&view=diff
==============================================================================
--- trunk/channels/h323/caps_h323.cxx (original)
+++ trunk/channels/h323/caps_h323.cxx Mon Sep 25 04:03:14 2006
@@ -18,6 +18,7 @@
H323_REGISTER_CAPABILITY(AST_G729ACapability, OPAL_G729A);
H323_REGISTER_CAPABILITY(AST_GSM0610Capability, OPAL_GSM0610);
H323_REGISTER_CAPABILITY(AST_CiscoG726Capability, CISCO_G726r32);
+H323_REGISTER_CAPABILITY(AST_CiscoDtmfCapability, CISCO_DTMF_RELAY);
OPAL_MEDIA_FORMAT_DECLARE(OpalG711ALaw64kFormat,
OPAL_G711_ALAW_64K,
@@ -99,6 +100,18 @@
8, // 1 millisecond
OpalMediaFormat::AudioTimeUnits,
0);
+#if 0
+OPAL_MEDIA_FORMAT_DECLARE(OpalCiscoDTMFRelayFormat,
+ CISCO_DTMF_RELAY,
+ OpalMediaFormat::DefaultAudioSessionID,
+ (RTP_DataFrame::PayloadTypes)121, // Choose this for Cisco IOS compatibility
+ TRUE, // Needs jitter
+ 100, // bits/sec
+ 4, // bytes/frame
+ 8*150, // 150 millisecond
+ OpalMediaFormat::AudioTimeUnits,
+ 0);
+#endif
/*
* Capability: G.711
Modified: trunk/channels/h323/chan_h323.h
URL: http://svn.digium.com/view/asterisk/trunk/channels/h323/chan_h323.h?rev=43597&r1=43596&r2=43597&view=diff
==============================================================================
--- trunk/channels/h323/chan_h323.h (original)
+++ trunk/channels/h323/chan_h323.h Mon Sep 25 04:03:14 2006
@@ -52,7 +52,7 @@
int progress_setup;
int progress_alert;
int progress_audio;
- int dtmfcodec;
+ int dtmfcodec[2];
int dtmfmode;
int capability;
int bridge;
@@ -172,7 +172,7 @@
/* This is a callback prototype function, called when
we know which RTP payload type RFC2833 will be
transmitted */
-typedef void (*rfc2833_cb)(unsigned, const char *, int);
+typedef void (*rfc2833_cb)(unsigned, const char *, int, int);
extern rfc2833_cb on_set_rfc2833_payload;
typedef void (*hangup_cb)(unsigned, const char *, int);
@@ -188,7 +188,9 @@
extern int h323debug;
#define H323_DTMF_RFC2833 (1 << 0)
-#define H323_DTMF_INBAND (1 << 1)
+#define H323_DTMF_CISCO (1 << 1)
+#define H323_DTMF_SIGNAL (1 << 2)
+#define H323_DTMF_INBAND (1 << 3)
#define H323_DTMF_RFC2833_PT 101
#define H323_DTMF_CISCO_PT 121
Modified: trunk/configs/h323.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/h323.conf.sample?rev=43597&r1=43596&r2=43597&view=diff
==============================================================================
--- trunk/configs/h323.conf.sample (original)
+++ trunk/configs/h323.conf.sample Mon Sep 25 04:03:14 2006
@@ -28,7 +28,7 @@
;
; User-Input Mode (DTMF)
;
-; valid entries are: rfc2833, inband
+; valid entries are: rfc2833, inband, cisco, h245-signal
; default is rfc2833
;dtmfmode=rfc2833
;
@@ -38,6 +38,8 @@
; To specify required payload type, put it after colon in dtmfmode
; option like
;dtmfmode=rfc2833:101
+; or
+;dtmfmode=cisco:121
;
; Set the gatekeeper
; DISCOVER - Find the Gk address using multicast
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