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-rw-r--r--net/asterisk10/files/dtmf_debug.diff225
1 files changed, 0 insertions, 225 deletions
diff --git a/net/asterisk10/files/dtmf_debug.diff b/net/asterisk10/files/dtmf_debug.diff
deleted file mode 100644
index 5179d42225fd..000000000000
--- a/net/asterisk10/files/dtmf_debug.diff
+++ /dev/null
@@ -1,225 +0,0 @@
---- include/asterisk/rtp.h.orig 2008-03-18 13:35:42.000000000 +0200
-+++ include/asterisk/rtp.h 2008-03-18 13:35:58.000000000 +0200
-@@ -251,6 +251,9 @@
-
- int ast_rtp_codec_getformat(int pt);
-
-+void ast_rtp_set_chan_name(struct ast_rtp *, const char *);
-+void ast_rtp_set_chan_id(struct ast_rtp *, const char *);
-+
- /*! \brief Set rtp timeout */
- void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout);
- /*! \brief Set rtp hold timeout */
---- main/rtp.c.orig 2008-04-08 14:53:18.000000000 +0300
-+++ main/rtp.c 2008-04-08 14:54:14.000000000 +0300
-@@ -81,6 +81,7 @@
- static int rtpstart; /*!< First port for RTP sessions (set in rtp.conf) */
- static int rtpend; /*!< Last port for RTP sessions (set in rtp.conf) */
- static int rtpdebug; /*!< Are we debugging? */
-+static int rtpdebugdtmf; /*!< Are we debugging DTMFs? */
- static int rtcpdebug; /*!< Are we debugging RTCP? */
- static int rtcpstats; /*!< Are we debugging RTCP? */
- static int rtcpinterval = RTCP_DEFAULT_INTERVALMS; /*!< Time between rtcp reports in millisecs */
-@@ -168,6 +169,8 @@
- struct ast_codec_pref pref;
- struct ast_rtp *bridged; /*!< Who we are Packet bridged to */
- int set_marker_bit:1; /*!< Whether to set the marker bit or not */
-+ char chan_name[100];
-+ char chan_id[100];
- };
-
- /* Forward declarations */
-@@ -669,8 +672,8 @@
- struct ast_frame *f = NULL;
- event = ntohl(*((unsigned int *)(data)));
- event &= 0x001F;
-- if (option_debug > 2 || rtpdebug)
-- ast_log(LOG_DEBUG, "Cisco DTMF Digit: %08x (len = %d)\n", event, len);
-+ if (option_debug > 2 || rtpdebug || rtpdebugdtmf)
-+ ast_log(LOG_DEBUG, "Channel: %s %s Cisco DTMF packet: %08x (len = %d)\n", rtp->chan_name, rtp->chan_id, event, len);
- if (event < 10) {
- resp = '0' + event;
- } else if (event < 11) {
-@@ -684,12 +687,24 @@
- }
- if (rtp->resp && (rtp->resp != resp)) {
- f = send_dtmf(rtp, AST_FRAME_DTMF_END);
-+ ast_log(LOG_DEBUG, "Channel: %s %s Cisco DTMF event: %c\n", rtp->chan_name, rtp->chan_id, rtp->resp);
- }
- rtp->resp = resp;
- rtp->dtmfcount = dtmftimeout;
- return f;
- }
-
-+void ast_rtp_set_chan_id(struct ast_rtp *rtp, const char *chan_id) {
-+ if (rtp == NULL || chan_id == NULL)
-+ return;
-+ snprintf(rtp->chan_id, sizeof(rtp->chan_id), "%s", chan_id);
-+}
-+
-+void ast_rtp_set_chan_name(struct ast_rtp *rtp, const char *chan_name) {
-+ if (rtp == NULL || chan_name == NULL)
-+ return;
-+ snprintf(rtp->chan_name, sizeof(rtp->chan_name), "%s", chan_name);
-+}
- /*!
- * \brief Process RTP DTMF and events according to RFC 2833.
- *
-@@ -1051,6 +1066,10 @@
- struct rtpPayloadType rtpPT;
- int reconstruct = ntohl(rtpheader[0]);
-
-+ /* If we are listening for DTMF - then feed all packets into the core to keep the RTP stream consistent when relaying DTMFs */
-+ if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF))
-+ return -1;
-+
- /* Get fields from packet */
- payload = (reconstruct & 0x7f0000) >> 16;
- mark = (((reconstruct & 0x800000) >> 23) != 0);
-@@ -1062,10 +1081,6 @@
- if (!bridged->current_RTP_PT[payload].code)
- return -1;
-
-- /* If the payload is DTMF, and we are listening for DTMF - then feed it into the core */
-- if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF) && !rtpPT.isAstFormat && rtpPT.code == AST_RTP_DTMF)
-- return -1;
--
- /* Otherwise adjust bridged payload to match */
- bridged_payload = ast_rtp_lookup_code(bridged, rtpPT.isAstFormat, rtpPT.code);
-
-@@ -1254,11 +1269,12 @@
- /* This is special in-band data that's not one of our codecs */
- if (rtpPT.code == AST_RTP_DTMF) {
- /* It's special -- rfc2833 process it */
-- if (rtp_debug_test_addr(&sin)) {
-+ if (rtp_debug_test_addr(&sin) || rtpdebugdtmf) {
- unsigned char *data;
- unsigned int event;
- unsigned int event_end;
- unsigned int duration;
-+
- data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen;
- event = ntohl(*((unsigned int *)(data)));
- event >>= 24;
-@@ -1267,9 +1283,12 @@
- event_end >>= 24;
- duration = ntohl(*((unsigned int *)(data)));
- duration &= 0xFFFF;
-- ast_verbose("Got RTP RFC2833 from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d) \n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration);
-+
-+ ast_verbose("Channel: %s %s Got RTP RFC2833 from %s:%u to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d)\n", rtp->chan_name, rtp->chan_id, ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), ast_inet_ntoa(rtp->us.sin_addr), ntohs(rtp->us.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration);
- }
- f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp);
-+ if (rtpdebugdtmf && f)
-+ ast_verbose("Channel: %s %s Got RFC2833 DTMF event %c of type %s\n", rtp->chan_name, rtp->chan_id, f->subclass, (f->frametype == AST_FRAME_DTMF_BEGIN ? "DTMF BEGIN" : (f->frametype == AST_FRAME_DTMF_END ? "DTMF_END" : "UNKNOWN")));
- } else if (rtpPT.code == AST_RTP_CISCO_DTMF) {
- /* It's really special -- process it the Cisco way */
- if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) {
-@@ -2198,8 +2217,9 @@
- ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n",
- ast_inet_ntoa(rtp->them.sin_addr),
- ntohs(rtp->them.sin_port), strerror(errno));
-- if (rtp_debug_test_addr(&rtp->them))
-- ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
-+ if (rtp_debug_test_addr(&rtp->them) || rtpdebugdtmf)
-+ ast_verbose("Channel: %s %s Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
-+ rtp->chan_name, rtp->chan_id,
- ast_inet_ntoa(rtp->them.sin_addr),
- ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
- /* Increment sequence number */
-@@ -2242,8 +2262,9 @@
- ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n",
- ast_inet_ntoa(rtp->them.sin_addr),
- ntohs(rtp->them.sin_port), strerror(errno));
-- if (rtp_debug_test_addr(&rtp->them))
-- ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
-+ if (rtp_debug_test_addr(&rtp->them) || rtpdebugdtmf)
-+ ast_verbose("Channel: %s %s Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
-+ rtp->chan_name, rtp->chan_id,
- ast_inet_ntoa(rtp->them.sin_addr),
- ntohs(rtp->them.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
-
-@@ -3481,6 +3502,16 @@
- return RESULT_SUCCESS;
- }
-
-+static int rtp_do_debug_dtmf(int fd, int argc, char *argv[])
-+{
-+ if (argc != 3)
-+ return RESULT_SHOWUSAGE;
-+
-+ rtpdebugdtmf = 1;
-+ ast_cli(fd, "RTP DTMF debugging enabled\n");
-+ return RESULT_SUCCESS;
-+}
-+
- static int rtp_do_debug(int fd, int argc, char *argv[])
- {
- if (argc != 2) {
-@@ -3541,6 +3572,7 @@
- if (argc != 3)
- return RESULT_SHOWUSAGE;
- rtpdebug = 0;
-+ rtpdebugdtmf = 0;
- ast_cli(fd,"RTP Debugging Disabled\n");
- return RESULT_SUCCESS;
- }
-@@ -3601,7 +3633,7 @@
- }
-
- static char debug_usage[] =
-- "Usage: rtp debug [ip host[:port]]\n"
-+ "Usage: rtp debug [ip host[:port] | dtmf]\n"
- " Enable dumping of all RTP packets to and from host.\n";
-
- static char no_debug_usage[] =
-@@ -3676,6 +3708,10 @@
- rtp_do_debug, "Enable RTP debugging",
- debug_usage },
-
-+ { { "rtp", "debug", "dtmf", NULL },
-+ rtp_do_debug_dtmf, "Enable RTP debugging on DTMFs",
-+ debug_usage },
-+
- { { "rtp", "debug", "off", NULL },
- rtp_no_debug, "Disable RTP debugging",
- no_debug_usage, NULL, &cli_rtp_no_debug_deprecated },
---- channels/chan_sip.c.orig 2008-06-10 00:29:41.000000000 -0700
-+++ channels/chan_sip.c 2008-06-10 00:42:00.000000000 -0700
-@@ -3813,6 +3813,7 @@
- ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, p->owner);
- else {
- p->owner = newchan;
-+ ast_rtp_set_chan_name(p->rtp, newchan->name);
- /* Re-invite RTP back to Asterisk. Needed if channel is masqueraded out of a native
- RTP bridge (i.e., RTP not going through Asterisk): RTP bridge code might not be
- able to do this if the masquerade happens before the bridge breaks (e.g., AMI
-@@ -4085,6 +4086,7 @@
- if (i->rtp) {
- tmp->fds[0] = ast_rtp_fd(i->rtp);
- tmp->fds[1] = ast_rtcp_fd(i->rtp);
-+ ast_rtp_set_chan_id(i->rtp, i->callid);
- }
- if (needvideo && i->vrtp) {
- tmp->fds[2] = ast_rtp_fd(i->vrtp);
-@@ -4112,6 +4114,8 @@
- if (!ast_strlen_zero(i->language))
- ast_string_field_set(tmp, language, i->language);
- i->owner = tmp;
-+ ast_rtp_set_chan_name(i->rtp, tmp->name);
-+
- ast_module_ref(ast_module_info->self);
- ast_copy_string(tmp->context, i->context, sizeof(tmp->context));
- /*Since it is valid to have extensions in the dialplan that have unescaped characters in them
-@@ -4531,8 +4535,10 @@
- build_via(p);
- if (!callid)
- build_callid_pvt(p);
-- else
-+ else {
- ast_string_field_set(p, callid, callid);
-+ ast_rtp_set_chan_id(p->rtp, p->callid);
-+ }
- /* Assign default music on hold class */
- ast_string_field_set(p, mohinterpret, default_mohinterpret);
- ast_string_field_set(p, mohsuggest, default_mohsuggest);
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-23 17:31:07 +0000