Make all of the various rtpqos parameters in this branch available from the CHANNEL function.

Also includes a test for retrieving rtpqos parameters, including a NULL RTP
driver.  Additionally, some further separation of the SIP internal API into
headers was necessary.

(closes issue #16652)
 Reported by: kkm
 Patches: 
       20100204__issue16652.diff.txt uploaded by tilghman (license 14)
 
Review: https://reviewboard.asterisk.org/r/501/


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247124 f38db490-d61c-443f-a65b-d21fe96a405b

5 files changed, 201 insertions, 3 deletions

diff --git a/channels/sip/include/config_parser.h b/channels/sip/include/config_parser.h
index 0b86188c6..76fefc2c6 100644
--- a/channels/sip/include/config_parser.h
+++ b/channels/sip/include/config_parser.h
@@ -43,7 +43,7 @@ int sip_parse_register_line(struct sip_registry *reg, int default_expiry, const
  */
 int sip_parse_host(char *line, int lineno, char **hostname, int *portnum, enum sip_transport *transport);
 
-/*! 
+/*!
  * \brief register config parsing tests
  */
 void sip_config_parser_register_tests(void);
diff --git a/channels/sip/include/dialog.h b/channels/sip/include/dialog.h
new file mode 100644
index 000000000..554aa5e6a
--- /dev/null
+++ b/channels/sip/include/dialog.h
@@ -0,0 +1,75 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2010, Digium, Inc.
+ *
+ * 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.
+ */
+
+/*!
+ * \file
+ * \brief sip dialog management header file
+ */
+
+#include "sip.h"
+
+#ifndef _SIP_DIALOG_H
+#define _SIP_DIALOG_H
+
+/*! \brief
+ * when we create or delete references, make sure to use these
+ * functions so we keep track of the refcounts.
+ * To simplify the code, we allow a NULL to be passed to dialog_unref().
+ */
+#define dialog_ref(arg1,arg2) dialog_ref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)
+#define dialog_unref(arg1,arg2) dialog_unref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)
+struct sip_pvt *dialog_ref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func);
+struct sip_pvt *dialog_unref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func);
+
+struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
+				 int useglobal_nat, const int intended_method, struct sip_request *req);
+void sip_scheddestroy(struct sip_pvt *p, int ms);
+int sip_cancel_destroy(struct sip_pvt *p);
+
+/*! \brief Destroy SIP call structure.
+ * Make it return NULL so the caller can do things like
+ *	foo = sip_destroy(foo);
+ * and reduce the chance of bugs due to dangling pointers.
+ */
+struct sip_pvt *sip_destroy(struct sip_pvt *p);
+
+/*! \brief Destroy SIP call structure.
+ * Make it return NULL so the caller can do things like
+ *	foo = sip_destroy(foo);
+ * and reduce the chance of bugs due to dangling pointers.
+ */
+void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist);
+/*!
+ * \brief Unlink a dialog from the dialogs container, as well as any other places
+ * that it may be currently stored.
+ *
+ * \note A reference to the dialog must be held before calling this function, and this
+ * function does not release that reference.
+ */
+void *dialog_unlink_all(struct sip_pvt *dialog, int lockowner, int lockdialoglist);
+
+/*! \brief Acknowledges receipt of a packet and stops retransmission
+ * called with p locked*/
+int __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
+
+/*! \brief Pretend to ack all packets
+ * called with p locked */
+void __sip_pretend_ack(struct sip_pvt *p);
+
+/*! \brief Acks receipt of packet, keep it around (used for provisional responses) */
+int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
+
+#endif /* defined(_SIP_DIALOG_H) */
diff --git a/channels/sip/include/dialplan_functions.h b/channels/sip/include/dialplan_functions.h
new file mode 100644
index 000000000..1600d4314
--- /dev/null
+++ b/channels/sip/include/dialplan_functions.h
@@ -0,0 +1,41 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2010, Digium, Inc.
+ *
+ * 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.
+ */
+
+/*!
+ * \file
+ * \brief SIP dialplan functions header file
+ */
+
+#include "sip.h"
+
+#ifndef _SIP_DIALPLAN_FUNCTIONS_H
+#define _SIP_DIALPLAN_FUNCTIONS_H
+
+/*!
+ * \brief Channel read dialplan function for SIP
+ */
+int sip_acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen);
+
+/*!
+ * \brief register dialplan function tests
+ */
+void sip_dialplan_function_register_tests(void);
+/*!
+ * \brief unregister dialplan function tests
+ */
+void sip_dialplan_function_unregister_tests(void);
+
+#endif /* !defined(_SIP_DIALPLAN_FUNCTIONS_H) */
diff --git a/channels/sip/include/globals.h b/channels/sip/include/globals.h
new file mode 100644
index 000000000..0d7131d87
--- /dev/null
+++ b/channels/sip/include/globals.h
@@ -0,0 +1,42 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2010, Digium, Inc.
+ *
+ * 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.
+ */
+
+/*!
+ * \file
+ * \brief sip global declaration header file
+ */
+
+#include "sip.h"
+
+#ifndef _SIP_GLOBALS_H
+#define _SIP_GLOBALS_H
+
+extern struct sockaddr_in bindaddr;     /*!< UDP: The address we bind to */
+extern struct sched_context *sched;     /*!< The scheduling context */
+
+/*! \brief Definition of this channel for PBX channel registration */
+extern const struct ast_channel_tech sip_tech;
+
+/*! \brief This version of the sip channel tech has no send_digit_begin
+ * callback so that the core knows that the channel does not want
+ * DTMF BEGIN frames.
+ * The struct is initialized just before registering the channel driver,
+ * and is for use with channels using SIP INFO DTMF.
+ */
+extern struct ast_channel_tech sip_tech_info;
+
+#endif /* !defined(SIP_GLOBALS_H) */
+
diff --git a/channels/sip/include/sip_utils.h b/channels/sip/include/sip_utils.h
index 549f9392f..47297ab68 100644
--- a/channels/sip/include/sip_utils.h
+++ b/channels/sip/include/sip_utils.h
@@ -22,8 +22,11 @@
 #ifndef _SIP_UTILS_H
 #define _SIP_UTILS_H
 
-/*! 
- * \brief converts ascii port to int representation. 
+/* wrapper macro to tell whether t points to one of the sip_tech descriptors */
+#define IS_SIP_TECH(t)  ((t) == &sip_tech || (t) == &sip_tech_info)
+
+/*!
+ * \brief converts ascii port to int representation.
  *
  * \arg pt[in] string that contains a port.
  * \arg standard[in] port to return in case the port string input is NULL
@@ -40,4 +43,41 @@ unsigned int port_str2int(const char *pt, unsigned int standard);
 const char *find_closing_quote(const char *start, const char *lim);
 
 
+/*! \brief Convert SIP hangup causes to Asterisk hangup causes */
+int hangup_sip2cause(int cause);
+
+/*! \brief Convert Asterisk hangup causes to SIP codes
+\verbatim
+ Possible values from causes.h
+        AST_CAUSE_NOTDEFINED    AST_CAUSE_NORMAL        AST_CAUSE_BUSY
+        AST_CAUSE_FAILURE       AST_CAUSE_CONGESTION    AST_CAUSE_UNALLOCATED
+
+	In addition to these, a lot of PRI codes is defined in causes.h
+	...should we take care of them too ?
+
+	Quote RFC 3398
+
+   ISUP Cause value                        SIP response
+   ----------------                        ------------
+   1  unallocated number                   404 Not Found
+   2  no route to network                  404 Not found
+   3  no route to destination              404 Not found
+   16 normal call clearing                 --- (*)
+   17 user busy                            486 Busy here
+   18 no user responding                   408 Request Timeout
+   19 no answer from the user              480 Temporarily unavailable
+   20 subscriber absent                    480 Temporarily unavailable
+   21 call rejected                        403 Forbidden (+)
+   22 number changed (w/o diagnostic)      410 Gone
+   22 number changed (w/ diagnostic)       301 Moved Permanently
+   23 redirection to new destination       410 Gone
+   26 non-selected user clearing           404 Not Found (=)
+   27 destination out of order             502 Bad Gateway
+   28 address incomplete                   484 Address incomplete
+   29 facility rejected                    501 Not implemented
+   31 normal unspecified                   480 Temporarily unavailable
+\endverbatim
+*/
+const char *hangup_cause2sip(int cause);
+
 #endif