[svn-commits] mmichelson: branch mmichelson/trunk-digiumphones r363686 - in /team/mmichelso...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Apr 25 14:12:35 CDT 2012
Author: mmichelson
Date: Wed Apr 25 14:12:33 2012
New Revision: 363686
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=363686
Log:
Remove AST_CONTROL_CUSTOM frame type and add a public function to send a SIP INFO request.
In 1.8-digiumphones and 10-digiumphones, an AST_CONTROL_CUSTOM frame type was created in order
to send a channel driver-specific indication. In addition, functions were added in order to
encode SIP INFO data into a frame and decode data from a frame.
The code in question had an odd mix of generic behavior and very SIP-specific behavior. Since this
was added with the primary purpose of being able to send SIP INFO requests to a phone, it seemed
like the generic elements should be eliminated in preference of a more direct method of sending
a SIP INFO.
With this change, a new file, sip_api.h has been added with a single function, ast_sipinfo_send().
The function is defined in chan_sip.c. This means that any modules that use ast_sipinfo_send() will
need to load after chan_sip.so has been loaded.
Several files have been removed since they are no longer useful.
Added:
team/mmichelson/trunk-digiumphones/channels/chan_sip.exports.in (with props)
team/mmichelson/trunk-digiumphones/include/asterisk/sip_api.h (with props)
Removed:
team/mmichelson/trunk-digiumphones/include/asterisk/custom_control_frame.h
team/mmichelson/trunk-digiumphones/main/custom_control_frame.c
team/mmichelson/trunk-digiumphones/tests/test_custom_control.c
Modified:
team/mmichelson/trunk-digiumphones/channels/chan_sip.c
team/mmichelson/trunk-digiumphones/funcs/func_frame_trace.c
team/mmichelson/trunk-digiumphones/include/asterisk/frame.h
team/mmichelson/trunk-digiumphones/main/channel.c
Modified: team/mmichelson/trunk-digiumphones/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/trunk-digiumphones/channels/chan_sip.c?view=diff&rev=363686&r1=363685&r2=363686
==============================================================================
--- team/mmichelson/trunk-digiumphones/channels/chan_sip.c (original)
+++ team/mmichelson/trunk-digiumphones/channels/chan_sip.c Wed Apr 25 14:12:33 2012
@@ -230,6 +230,7 @@
affect the speed of the program at all. They can be considered to be documentation.
*/
/* #define REF_DEBUG 1 */
+
#include "asterisk/lock.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
@@ -278,7 +279,7 @@
#include "sip/include/dialog.h"
#include "sip/include/dialplan_functions.h"
#include "sip/include/security_events.h"
-
+#include "asterisk/sip_api.h"
/*** DOCUMENTATION
<application name="SIPDtmfMode" language="en_US">
@@ -6944,34 +6945,37 @@
return 0;
}
-/* XXX Candidate for moving into its own file */
-/*!
- * \brief Sends AST_CUSTOM_FRAME of type sip info.
- *
- * \note pvt is expected to be locked before entering this function.
- */
-static int sip_handle_custom_info(struct sip_pvt *pvt, struct ast_custom_payload *pl)
-{
- struct ast_variable *headers = NULL;
- char *content_type = NULL;
- char *content = NULL;
- char *useragent_filter = NULL;
+int ast_sipinfo_send(
+ struct ast_channel *chan,
+ struct ast_variable *headers,
+ const char *content_type,
+ const char *content,
+ const char *useragent_filter)
+{
+ struct sip_pvt *p;
struct ast_variable *var;
struct sip_request req;
int res = -1;
- if (ast_custom_payload_sipinfo_decode(pl, &headers, &content_type, &content, &useragent_filter)) {
- goto custom_info_cleanup;
- }
+ ast_channel_lock(chan);
+
+ if (ast_channel_tech(chan) != &sip_tech) {
+ ast_log(LOG_WARNING, "Attempted to send a custom INFO on a non-SIP channel %s\n", ast_channel_name(chan));
+ ast_channel_unlock(chan);
+ return res;
+ }
+
+ p = ast_channel_tech_pvt(chan);
+ sip_pvt_lock(p);
if (!(ast_strlen_zero(useragent_filter))) {
- int match = (strstr(pvt->useragent, useragent_filter)) ? 1 : 0;
+ int match = (strstr(p->useragent, useragent_filter)) ? 1 : 0;
if (!match) {
- goto custom_info_cleanup;
- }
- }
-
- reqprep(&req, pvt, SIP_INFO, 0, 1);
+ goto cleanup;
+ }
+ }
+
+ reqprep(&req, p, SIP_INFO, 0, 1);
for (var = headers; var; var = var->next) {
add_header(&req, var->name, var->value);
}
@@ -6980,18 +6984,13 @@
add_content(&req, content);
}
- res = send_request(pvt, &req, XMIT_RELIABLE, pvt->ocseq);
-
-custom_info_cleanup:
-
- ast_free(content);
- ast_free(content_type);
- ast_free(useragent_filter);
- ast_variables_destroy(headers);
-
+ res = send_request(p, &req, XMIT_RELIABLE, p->ocseq);
+
+cleanup:
+ sip_pvt_unlock(p);
+ ast_channel_unlock(chan);
return res;
}
-
/*! \brief Play indication to user
* With SIP a lot of indications is sent as messages, letting the device play
the indication - busy signal, congestion etc
@@ -7120,11 +7119,6 @@
break;
case AST_CONTROL_REDIRECTING:
update_redirecting(p, data, datalen);
- break;
- case AST_CONTROL_CUSTOM:
- if (datalen && ast_custom_payload_type((struct ast_custom_payload *) data) == AST_CUSTOM_SIP_INFO) {
- sip_handle_custom_info(p, (struct ast_custom_payload *) data);
- }
break;
case AST_CONTROL_AOC:
{
@@ -30779,7 +30773,6 @@
static int sip_sendcustominfo(struct ast_channel *chan, const char *data)
{
char *info_data, *useragent;
- struct ast_custom_payload *pl = NULL;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "You must provide data to be sent\n");
@@ -30789,13 +30782,10 @@
useragent = ast_strdupa(data);
info_data = strsep(&useragent, ",");
- if (!(pl = ast_custom_payload_sipinfo_encode(NULL, "text/plain", info_data, useragent))) {
+ if (ast_sipinfo_send(chan, NULL, "text/plain", info_data, useragent)) {
ast_log(LOG_WARNING, "Failed to create payload for custom SIP INFO\n");
return 0;
}
-
- ast_indicate_data(chan, AST_CONTROL_CUSTOM, pl, ast_custom_payload_len(pl));
- ast_free(pl);
return 0;
}
#endif
@@ -32005,7 +31995,7 @@
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Session Initiation Protocol (SIP)",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Session Initiation Protocol (SIP)",
.load = load_module,
.unload = unload_module,
.reload = reload,
Added: team/mmichelson/trunk-digiumphones/channels/chan_sip.exports.in
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/trunk-digiumphones/channels/chan_sip.exports.in?view=auto&rev=363686
==============================================================================
--- team/mmichelson/trunk-digiumphones/channels/chan_sip.exports.in (added)
+++ team/mmichelson/trunk-digiumphones/channels/chan_sip.exports.in Wed Apr 25 14:12:33 2012
@@ -1,0 +1,6 @@
+{
+ global:
+ LINKER_SYMBOL_PREFIX*ast_sipinfo_send;
+ local:
+ *;
+};
Propchange: team/mmichelson/trunk-digiumphones/channels/chan_sip.exports.in
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: team/mmichelson/trunk-digiumphones/channels/chan_sip.exports.in
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: team/mmichelson/trunk-digiumphones/channels/chan_sip.exports.in
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: team/mmichelson/trunk-digiumphones/funcs/func_frame_trace.c
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/trunk-digiumphones/funcs/func_frame_trace.c?view=diff&rev=363686&r1=363685&r2=363686
==============================================================================
--- team/mmichelson/trunk-digiumphones/funcs/func_frame_trace.c (original)
+++ team/mmichelson/trunk-digiumphones/funcs/func_frame_trace.c Wed Apr 25 14:12:33 2012
@@ -321,9 +321,6 @@
case AST_CONTROL_END_OF_Q:
ast_verbose("SubClass: END_OF_Q\n");
break;
- case AST_CONTROL_CUSTOM:
- ast_verbose("Subclass: Custom");
- break;
case AST_CONTROL_UPDATE_RTP_PEER:
ast_verbose("SubClass: UPDATE_RTP_PEER\n");
break;
Modified: team/mmichelson/trunk-digiumphones/include/asterisk/frame.h
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/trunk-digiumphones/include/asterisk/frame.h?view=diff&rev=363686&r1=363685&r2=363686
==============================================================================
--- team/mmichelson/trunk-digiumphones/include/asterisk/frame.h (original)
+++ team/mmichelson/trunk-digiumphones/include/asterisk/frame.h Wed Apr 25 14:12:33 2012
@@ -263,8 +263,6 @@
AST_CONTROL_READ_ACTION = 27, /*!< Tell ast_read to take a specific action */
AST_CONTROL_AOC = 28, /*!< Advice of Charge with encoded generic AOC payload */
AST_CONTROL_END_OF_Q = 29, /*!< Indicate that this position was the end of the channel queue for a softhangup. */
- /* XXX WTF? Why 200? XXX */
- AST_CONTROL_CUSTOM = 200, /*!< Indicate a custom channel driver specific payload. Look in custom_control_frame.h for how to define and use this frame. */
AST_CONTROL_INCOMPLETE = 30, /*!< Indication that the extension dialed is incomplete */
AST_CONTROL_MCID = 31, /*!< Indicate that the caller is being malicious. */
AST_CONTROL_UPDATE_RTP_PEER = 32, /*!< Interrupt the bridge and have it update the peer */
Added: team/mmichelson/trunk-digiumphones/include/asterisk/sip_api.h
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/trunk-digiumphones/include/asterisk/sip_api.h?view=auto&rev=363686
==============================================================================
--- team/mmichelson/trunk-digiumphones/include/asterisk/sip_api.h (added)
+++ team/mmichelson/trunk-digiumphones/include/asterisk/sip_api.h Wed Apr 25 14:12:33 2012
@@ -1,0 +1,51 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2012, Digium, Inc.
+ *
+ * Mark Michelson <mmichelson at digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+#ifndef __ASTERISK_SIP_H
+#define __ASTERISK_SIP_H
+
+#if defined(__cplusplus) || defined(c_plusplus)
+extern "C" {
+#endif
+
+#include "asterisk/optional_api.h"
+#include "asterisk/config.h"
+
+/*!
+ * \brief Send a customized SIP INFO request
+ *
+ * \param headers The headers to add to the INFO request
+ * \param content_type The content type header to add
+ * \param conten The body of the INFO request
+ * \param useragent_filter If non-NULL, only send the INFO if the
+ * recipient's User-Agent contains useragent_filter as a substring
+ *
+ * \retval 0 Success
+ * \retval non-zero Failure
+ */
+int ast_sipinfo_send(struct ast_channel *chan,
+ struct ast_variable *headers,
+ const char *content_type,
+ const char *content,
+ const char *useragent_filter);
+
+#if defined(__cplusplus) || defined(c_plusplus)
+}
+#endif
+
+#endif /* __ASTERISK_SIP_H */
Propchange: team/mmichelson/trunk-digiumphones/include/asterisk/sip_api.h
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: team/mmichelson/trunk-digiumphones/include/asterisk/sip_api.h
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: team/mmichelson/trunk-digiumphones/include/asterisk/sip_api.h
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: team/mmichelson/trunk-digiumphones/main/channel.c
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/trunk-digiumphones/main/channel.c?view=diff&rev=363686&r1=363685&r2=363686
==============================================================================
--- team/mmichelson/trunk-digiumphones/main/channel.c (original)
+++ team/mmichelson/trunk-digiumphones/main/channel.c Wed Apr 25 14:12:33 2012
@@ -4165,7 +4165,6 @@
case AST_CONTROL_CC:
case AST_CONTROL_READ_ACTION:
case AST_CONTROL_AOC:
- case AST_CONTROL_CUSTOM:
case AST_CONTROL_END_OF_Q:
case AST_CONTROL_MCID:
case AST_CONTROL_UPDATE_RTP_PEER:
@@ -4355,7 +4354,6 @@
case AST_CONTROL_CC:
case AST_CONTROL_READ_ACTION:
case AST_CONTROL_AOC:
- case AST_CONTROL_CUSTOM:
case AST_CONTROL_END_OF_Q:
case AST_CONTROL_MCID:
case AST_CONTROL_UPDATE_RTP_PEER:
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