chan_sip: Remove deprecated module.

ASTERISK-30297

Change-Id: Ic700168c80b68879d9cee8bb07afe2712fb17996

include/asterisk/ccss.h

@@ -518,7 +518,7 @@ struct ast_cc_monitor {
 	 * \details
 	 * When issuing a CC recall, some technologies will require
 	 * that a name other than the device name is dialed. For instance,
-	 * with SIP, a specific URI will be used which chan_sip will be able
+	 * with SIP, a specific URI will be used which sip will be able
 	 * to recognize as being a CC recall. Similarly, ISDN will need a specific
 	 * dial string to know that the call is a recall.
 	 */

include/asterisk/channel.h

@@ -616,7 +616,7 @@ struct ast_msg_data;
  * Structure to describe a channel "technology", ie a channel driver
  * See for examples:
  * \arg chan_iax2.c - The Inter-Asterisk exchange protocol
- * \arg chan_sip.c - The SIP channel driver
+ * \arg chan_pjsip.c - The SIP channel driver
  * \arg chan_dahdi.c - PSTN connectivity (TDM, PRI, T1/E1, FXO, FXS)
  *
  * \details

include/asterisk/config_options.h

@@ -124,8 +124,8 @@ struct aco_type {
 	const char *name;       /*!< The name of this type (must match XML documentation) */
 	const char *category;   /*!< A regular expression for matching categories to be allowed or denied */
 	const char *matchfield; /*!< An option name to match for this type (i.e. a 'type'-like column) */
-	const char *matchvalue; /*!< The value of the option to require for matching (i.e. 'peer' for type= in sip.conf) */
-	aco_matchvalue_func matchfunc;       /*!< A function for determining whether the option value matches (i.e. hassip= requires ast_true()) */
+	const char *matchvalue; /*!< The value of the option to require for matching (i.e. 'peer') */
+	aco_matchvalue_func matchfunc;       /*!< A function for determining whether the option value matches */
 	enum aco_category_op category_match; /*!< Whether the following category regex is a whitelist or blacklist */
 	size_t item_offset;                  /*!< The offset in the config snapshot for the global config or item config container */
 	unsigned int hidden;  /*!< Type is for internal purposes only and it and all options should not be visible to users */

include/asterisk/doxygen/architecture.h

@@ -497,7 +497,7 @@ Example dialplan:
 <code>exten => 5551212,n,Hangup()</code><br/>
 
  -# <b>Call Setup:</b> An incoming SIP INVITE begins this scenario.  It is received by
-    the SIP channel driver (chan_sip.c).  Specifically, the monitor thread in chan_sip
+    the SIP channel driver (chan_pjsip.c).  Specifically, the monitor thread in chan_pjsip
     is responsible for handling this incoming request.  Further, the monitor thread
     is responsible for completing any handshake necessary to complete the call setup
     process.
@@ -517,8 +517,8 @@ Example dialplan:
     code simply executes the ast_answer() API call.  This API call operates on an
     ast_channel.  It handles generic ast_channel hangup processing, as well as executes
     the answer callback function defined in the associated ast_channel_tech for the
-    active channel.  In this case, the sip_answer() function in chan_sip.c will get
-    executed to handle the SIP specific operations required to answer a call.
+    active channel.  In this case, the chan_pjsip_answer() function in chan_pjsip.c will
+    get executed to handle the SIP specific operations required to answer a call.
  -# <b>Play the File:</b> The next step of the dialplan says to play back a %sound file
     to the caller.  The <code>Playback()</code> application will be executed.
     The code for this application is in apps/app_playback.c.  The code in the application
@@ -562,7 +562,7 @@ Example dialplan:
 <code>exten => 5551212,n,Dial(IAX2/mypeer)</code><br/>
 
  -# <b>Call Setup:</b> An incoming SIP INVITE begins this scenario.  It is received by
-    the SIP channel driver (chan_sip.c).  Specifically, the monitor thread in chan_sip
+    the SIP channel driver (chan_pjsip.c).  Specifically, the monitor thread in chan_pjsip
     is responsible for handling this incoming request.  Further, the monitor thread
     is responsible for completing any handshake necessary to complete the call setup
     process.

include/asterisk/doxyref.h

@@ -250,7 +250,7 @@
 /*!
  * \page Config_rtp RTP configuration
  * \arg Implemented in \ref rtp.c
- * Used in \ref chan_sip.c (and various H.323 channels)
+ * Used in various H.323 channels
  * \section rtpconf rtp.conf
  * \verbinclude rtp.conf.sample
  */

include/asterisk/module.h

@@ -242,7 +242,7 @@ int ast_update_module_list_condition(int (*modentry)(const char *module, const c
 
 /*!
  * \brief Check if module with the name given is loaded
- * \param name Module name, like "chan_sip.so"
+ * \param name Module name, like "chan_pjsip.so"
  * \retval 1 if true
  * \retval 0 if false
  */

include/asterisk/stasis_channels.h

@@ -112,7 +112,7 @@ struct ast_channel_snapshot_base {
 	);
 	struct timeval creationtime; /*!< The time of channel creation */
 	int tech_properties;         /*!< Properties of the channel's technology */
-	AST_STRING_FIELD_EXTENDED(protocol_id); /*!< Channel driver protocol id (i.e. Call-ID for chan_sip/chan_pjsip) */
+	AST_STRING_FIELD_EXTENDED(protocol_id); /*!< Channel driver protocol id (i.e. Call-ID for chan_pjsip) */
 };
 
 /*!

include/asterisk/tcptls.h

@@ -26,7 +26,7 @@
  * in or out the DO_SSL macro.
  *
  * TLS/SSL support is basically implemented by reading from a config file
- * (currently manager.conf, http.conf and sip.conf) the names of the certificate
+ * (currently manager.conf, http.conf and pjsip.conf) the names of the certificate
  * files and cipher to use, and then run ssl_setup() to create an appropriate
  * data structure named ssl_ctx.
  *