[asterisk-commits] oej: branch oej/videocaps r116233 - in /team/oej/videocaps: ./ channels/ main...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed May 14 08:03:52 CDT 2008
Author: oej
Date: Wed May 14 08:03:52 2008
New Revision: 116233
URL: http://svn.digium.com/view/asterisk?view=rev&rev=116233
Log:
Update to trunk again, try to turn on automerge
Modified:
team/oej/videocaps/ (props changed)
team/oej/videocaps/CHANGES
team/oej/videocaps/channels/chan_sip.c
team/oej/videocaps/main/manager.c
team/oej/videocaps/pbx/pbx_spool.c
team/oej/videocaps/res/res_agi.c
team/oej/videocaps/sample.call
Propchange: team/oej/videocaps/
------------------------------------------------------------------------------
automerge = http://www.codename-pineapple.org/
Propchange: team/oej/videocaps/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Wed May 14 08:03:52 2008
@@ -1,1 +1,1 @@
-/trunk:1-116179
+/trunk:1-116231
Modified: team/oej/videocaps/CHANGES
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/CHANGES?view=diff&rev=116233&r1=116232&r2=116233
==============================================================================
--- team/oej/videocaps/CHANGES (original)
+++ team/oej/videocaps/CHANGES Wed May 14 08:03:52 2008
@@ -166,6 +166,7 @@
* Originate now requires the Originate privilege and, if you want to call out
to a subshell, it requires the System privilege, as well. This was done to
enhance manager security.
+ * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
* New command: Atxfer. See doc/manager_1_1.txt for more details or
manager show command Atxfer from the CLI
Modified: team/oej/videocaps/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/channels/chan_sip.c?view=diff&rev=116233&r1=116232&r2=116233
==============================================================================
--- team/oej/videocaps/channels/chan_sip.c (original)
+++ team/oej/videocaps/channels/chan_sip.c Wed May 14 08:03:52 2008
@@ -25,15 +25,21 @@
* See Also:
* \arg \ref AstCREDITS
*
- * Implementation of RFC 3261 - without S/MIME, TCP and TLS support
+ * Implementation of RFC 3261 - without S/MIME, and experimental TCP and TLS support
* Configuration file \link Config_sip sip.conf \endlink
*
+ * ********** IMPORTANT *
+ * \note TCP/TLS support is EXPERIMENTAL and WILL CHANGE. This applies to configuration
+ * settings, dialplan commands and dialplans apps/functions
+ *
*
+ * TODO:s
* \todo Better support of forking
* \todo VIA branch tag transaction checking
* \todo Transaction support
* \todo We need to test TCP sessions with SIP proxies and in regards
* to the SIP outbound specs.
+ * \todo Fix TCP/TLS handling in dialplan, SRV records, transfers and much more
*
* \ingroup channel_drivers
*
@@ -12643,11 +12649,12 @@
Reference: RFC 3428 */
static void receive_message(struct sip_pvt *p, struct sip_request *req)
{
- char buf[1024];
+ char buf[1400];
struct ast_frame f;
const char *content_type = get_header(req, "Content-Type");
- if (strcmp(content_type, "text/plain")) { /* No text/plain attachment */
+ if (strncmp(content_type, "text/plain", strlen("text/plain"))) {
+ //if (strcmp(content_type, "text/plain")) { /* No text/plain attachment */
transmit_response(p, "415 Unsupported Media Type", req);
if (!p->owner)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
@@ -12664,7 +12671,7 @@
if (p->owner) {
if (sip_debug_test_pvt(p))
- ast_verbose("Message received: '%s'\n", buf);
+ ast_verbose("SIP Text message received: '%s'\n", buf);
memset(&f, 0, sizeof(f));
f.frametype = AST_FRAME_TEXT;
f.subclass = 0;
@@ -12673,11 +12680,13 @@
f.datalen = strlen(buf);
ast_queue_frame(p->owner, &f);
transmit_response(p, "202 Accepted", req); /* We respond 202 accepted, since we relay the message */
- } else { /* Message outside of a call, we do not support that */
- ast_log(LOG_WARNING, "Received message to %s from %s, dropped it...\n Content-Type:%s\n Message: %s\n", get_header(req, "To"), get_header(req, "From"), content_type, buf);
- transmit_response(p, "405 Method Not Allowed", req);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- }
+ return;
+ }
+
+ /* Message outside of a call, we do not support that */
+ ast_log(LOG_WARNING, "Received message to %s from %s, dropped it...\n Content-Type:%s\n Message: %s\n", get_header(req, "To"), get_header(req, "From"), content_type, buf);
+ transmit_response(p, "405 Method Not Allowed", req);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return;
}
Modified: team/oej/videocaps/main/manager.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/main/manager.c?view=diff&rev=116233&r1=116232&r2=116233
==============================================================================
--- team/oej/videocaps/main/manager.c (original)
+++ team/oej/videocaps/main/manager.c Wed May 14 08:03:52 2008
@@ -2106,11 +2106,12 @@
return 0;
}
-/* helper function for originate */
+/*! \brief helper function for originate */
struct fast_originate_helper {
char tech[AST_MAX_EXTENSION];
char data[AST_MAX_EXTENSION];
int timeout;
+ int format; /*!< Codecs used for a call */
char app[AST_MAX_APP];
char appdata[AST_MAX_EXTENSION];
char cid_name[AST_MAX_EXTENSION];
@@ -2132,12 +2133,12 @@
char requested_channel[AST_CHANNEL_NAME];
if (!ast_strlen_zero(in->app)) {
- res = ast_pbx_outgoing_app(in->tech, AST_FORMAT_SLINEAR, in->data, in->timeout, in->app, in->appdata, &reason, 1,
+ res = ast_pbx_outgoing_app(in->tech, in->format, in->data, in->timeout, in->app, in->appdata, &reason, 1,
S_OR(in->cid_num, NULL),
S_OR(in->cid_name, NULL),
in->vars, in->account, &chan);
} else {
- res = ast_pbx_outgoing_exten(in->tech, AST_FORMAT_SLINEAR, in->data, in->timeout, in->context, in->exten, in->priority, &reason, 1,
+ res = ast_pbx_outgoing_exten(in->tech, in->format, in->data, in->timeout, in->context, in->exten, in->priority, &reason, 1,
S_OR(in->cid_num, NULL),
S_OR(in->cid_name, NULL),
in->vars, in->account, &chan);
@@ -2198,6 +2199,7 @@
const char *appdata = astman_get_header(m, "Data");
const char *async = astman_get_header(m, "Async");
const char *id = astman_get_header(m, "ActionID");
+ const char *codecs = astman_get_header(m, "Codecs");
struct ast_variable *vars = astman_get_variables(m);
char *tech, *data;
char *l = NULL, *n = NULL;
@@ -2207,6 +2209,7 @@
int reason = 0;
char tmp[256];
char tmp2[256];
+ int format = AST_FORMAT_SLINEAR;
pthread_t th;
if (!name) {
@@ -2241,6 +2244,10 @@
ast_shrink_phone_number(l);
if (ast_strlen_zero(l))
l = NULL;
+ }
+ if (!ast_strlen_zero(codecs)) {
+ format = 0;
+ ast_parse_allow_disallow(NULL, &format, codecs, 1);
}
if (ast_true(async)) {
struct fast_originate_helper *fast = ast_calloc(1, sizeof(*fast));
@@ -2261,6 +2268,7 @@
ast_copy_string(fast->context, context, sizeof(fast->context));
ast_copy_string(fast->exten, exten, sizeof(fast->exten));
ast_copy_string(fast->account, account, sizeof(fast->account));
+ fast->format = format;
fast->timeout = to;
fast->priority = pi;
if (ast_pthread_create_detached(&th, NULL, fast_originate, fast)) {
@@ -2285,10 +2293,10 @@
astman_send_error(s, m, "Originate with certain 'Application' arguments requires the additional System privilege, which you do not have.");
return 0;
}
- res = ast_pbx_outgoing_app(tech, AST_FORMAT_SLINEAR, data, to, app, appdata, &reason, 1, l, n, vars, account, NULL);
+ res = ast_pbx_outgoing_app(tech, format, data, to, app, appdata, &reason, 1, l, n, vars, account, NULL);
} else {
if (exten && context && pi)
- res = ast_pbx_outgoing_exten(tech, AST_FORMAT_SLINEAR, data, to, context, exten, pi, &reason, 1, l, n, vars, account, NULL);
+ res = ast_pbx_outgoing_exten(tech, format, data, to, context, exten, pi, &reason, 1, l, n, vars, account, NULL);
else {
astman_send_error(s, m, "Originate with 'Exten' requires 'Context' and 'Priority'");
return 0;
Modified: team/oej/videocaps/pbx/pbx_spool.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/pbx/pbx_spool.c?view=diff&rev=116233&r1=116232&r2=116233
==============================================================================
--- team/oej/videocaps/pbx/pbx_spool.c (original)
+++ team/oej/videocaps/pbx/pbx_spool.c Wed May 14 08:03:52 2008
@@ -66,6 +66,7 @@
int retrytime; /*!< How long to wait between retries (in seconds) */
int waittime; /*!< How long to wait for an answer */
long callingpid; /*!< PID which is currently calling */
+ int format; /*!< Formats (codecs) for this call */
char tech[256]; /*!< Which channel driver to use for outgoing call */
char dest[256]; /*!< Which device/line to use for outgoing call */
@@ -94,6 +95,7 @@
o->priority = 1;
o->retrytime = 300;
o->waittime = 45;
+ o->format = AST_FORMAT_SLINEAR;
ast_set_flag(&o->options, SPOOL_FLAG_ALWAYS_DELETE);
}
@@ -165,6 +167,8 @@
ast_log(LOG_WARNING, "Invalid max retries at line %d of %s\n", lineno, fn);
o->maxretries = 0;
}
+ } else if (!strcasecmp(buf, "codecs")) {
+ ast_parse_allow_disallow(NULL, &o->format, c, 1);
} else if (!strcasecmp(buf, "context")) {
ast_copy_string(o->context, c, sizeof(o->context));
} else if (!strcasecmp(buf, "extension")) {
@@ -310,10 +314,10 @@
int res, reason;
if (!ast_strlen_zero(o->app)) {
ast_verb(3, "Attempting call on %s/%s for application %s(%s) (Retry %d)\n", o->tech, o->dest, o->app, o->data, o->retries);
- res = ast_pbx_outgoing_app(o->tech, AST_FORMAT_SLINEAR, o->dest, o->waittime * 1000, o->app, o->data, &reason, 2 /* wait to finish */, o->cid_num, o->cid_name, o->vars, o->account, NULL);
+ res = ast_pbx_outgoing_app(o->tech, o->format, o->dest, o->waittime * 1000, o->app, o->data, &reason, 2 /* wait to finish */, o->cid_num, o->cid_name, o->vars, o->account, NULL);
} else {
ast_verb(3, "Attempting call on %s/%s for %s@%s:%d (Retry %d)\n", o->tech, o->dest, o->exten, o->context,o->priority, o->retries);
- res = ast_pbx_outgoing_exten(o->tech, AST_FORMAT_SLINEAR, o->dest, o->waittime * 1000, o->context, o->exten, o->priority, &reason, 2 /* wait to finish */, o->cid_num, o->cid_name, o->vars, o->account, NULL);
+ res = ast_pbx_outgoing_exten(o->tech, o->format, o->dest, o->waittime * 1000, o->context, o->exten, o->priority, &reason, 2 /* wait to finish */, o->cid_num, o->cid_name, o->vars, o->account, NULL);
}
if (res) {
ast_log(LOG_NOTICE, "Call failed to go through, reason (%d) %s\n", reason, ast_channel_reason2str(reason));
Modified: team/oej/videocaps/res/res_agi.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/res/res_agi.c?view=diff&rev=116233&r1=116232&r2=116233
==============================================================================
--- team/oej/videocaps/res/res_agi.c (original)
+++ team/oej/videocaps/res/res_agi.c Wed May 14 08:03:52 2008
@@ -811,15 +811,13 @@
if (res == 0) {
ast_agi_fdprintf(chan, agi->fd, "200 result=%d (timeout)\n", res);
return RESULT_SUCCESS;
- }
+ }
if (res > 0) {
ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", res);
return RESULT_SUCCESS;
}
- else {
- ast_agi_fdprintf(chan, agi->fd, "200 result=%d (hangup)\n", res);
- return RESULT_FAILURE;
- }
+ ast_agi_fdprintf(chan, agi->fd, "200 result=%d (hangup)\n", res);
+ return RESULT_FAILURE;
}
static int handle_recvtext(struct ast_channel *chan, AGI *agi, int argc, char *argv[])
@@ -829,7 +827,7 @@
if (argc != 3)
return RESULT_SHOWUSAGE;
- buf = ast_recvtext(chan,atoi(argv[2]));
+ buf = ast_recvtext(chan, atoi(argv[2]));
if (buf) {
ast_agi_fdprintf(chan, agi->fd, "200 result=1 (%s)\n", buf);
ast_free(buf);
@@ -846,19 +844,23 @@
if (argc != 3)
return RESULT_SHOWUSAGE;
- if (!strncasecmp(argv[2],"on",2))
+ if (!strncasecmp(argv[2],"on",2)) {
x = 1;
- else
+ } else {
x = 0;
- if (!strncasecmp(argv[2],"mate",4))
+ }
+ if (!strncasecmp(argv[2],"mate",4)) {
x = 2;
- if (!strncasecmp(argv[2],"tdd",3))
+ }
+ if (!strncasecmp(argv[2],"tdd",3)) {
x = 1;
+ }
res = ast_channel_setoption(chan, AST_OPTION_TDD, &x, sizeof(char), 0);
- if (res != RESULT_SUCCESS)
+ if (res != RESULT_SUCCESS) {
ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
- else
+ } else {
ast_agi_fdprintf(chan, agi->fd, "200 result=1\n");
+ }
return RESULT_SUCCESS;
}
@@ -866,12 +868,14 @@
{
int res;
- if (argc != 3)
- return RESULT_SHOWUSAGE;
+ if (argc != 3) {
+ return RESULT_SHOWUSAGE;
+ }
res = ast_send_image(chan, argv[2]);
- if (!ast_check_hangup(chan))
+ if (!ast_check_hangup(chan)) {
res = 0;
+ }
ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", res);
return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
}
@@ -879,33 +883,31 @@
static int handle_controlstreamfile(struct ast_channel *chan, AGI *agi, int argc, char *argv[])
{
int res = 0, skipms = 3000;
- char *fwd = NULL, *rev = NULL, *pause = NULL, *stop = NULL;
-
- if (argc < 5 || argc > 9)
- return RESULT_SHOWUSAGE;
-
- if (!ast_strlen_zero(argv[4]))
+ char *fwd = "#", *rev = "*", *pause = NULL, *stop = NULL; /* Default values */
+
+ if (argc < 5 || argc > 9) {
+ return RESULT_SHOWUSAGE;
+ }
+
+ if (!ast_strlen_zero(argv[4])) {
stop = argv[4];
- else
- stop = NULL;
-
- if ((argc > 5) && (sscanf(argv[5], "%d", &skipms) != 1))
- return RESULT_SHOWUSAGE;
-
- if (argc > 6 && !ast_strlen_zero(argv[6]))
+ }
+
+ if ((argc > 5) && (sscanf(argv[5], "%d", &skipms) != 1)) {
+ return RESULT_SHOWUSAGE;
+ }
+
+ if (argc > 6 && !ast_strlen_zero(argv[6])) {
fwd = argv[6];
- else
- fwd = "#";
-
- if (argc > 7 && !ast_strlen_zero(argv[7]))
+ }
+
+ if (argc > 7 && !ast_strlen_zero(argv[7])) {
rev = argv[7];
- else
- rev = "*";
-
- if (argc > 8 && !ast_strlen_zero(argv[8]))
+ }
+
+ if (argc > 8 && !ast_strlen_zero(argv[8])) {
pause = argv[8];
- else
- pause = NULL;
+ }
res = ast_control_streamfile(chan, argv[3], fwd, rev, stop, pause, NULL, skipms, NULL);
Modified: team/oej/videocaps/sample.call
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/sample.call?view=diff&rev=116233&r1=116232&r2=116233
==============================================================================
--- team/oej/videocaps/sample.call (original)
+++ team/oej/videocaps/sample.call Wed May 14 08:03:52 2008
@@ -13,6 +13,9 @@
# would for the "Dial" application. Only one channel name is permitted.
#
Channel: Zap/1
+#
+# You can specify codecs for the call
+Codecs: alaw, speex, h264
#
# You may also specify a wait time (default is 45 seconds) for how long to
# wait for the channel to be answered, a retry time (default is 5 mins)
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