Clean up chan_sip

This clean up was broken out from
https://reviewboard.asterisk.org/r/1976/ and addresses the following:
 - struct sip_refer converted to use the stringfields API.
 - sip_{refer|notify}_allocate -> sip_{notify|refer}_alloc to match
   other *alloc functions.
 - Replace get_msg_text, get_msg_text2 and get_pidf_body -> No, not
   get_pidf_msg_text_body3 but get_content, to match add_content.
 - get_body doesn't get the request body, renamed to get_content_line.
 - get_body_by_line doesn't get the body line, and is just a simple if
   test. Moved code inline and removed function.
 - Remove camelCase in struct sip_peer peer state variables,
   onHold -> onhold, inUse -> inuse, inRinging -> ringing.
 - Remove camelCase in struct sip_request rlPart1 -> rlpart1,
   rlPart2 -> rlpart2.
 - Rename instances of pvt->randdata to pvt->nonce because that is what
   it is, no need to update struct sip_pvt because _it already has a
   nonce field_.
 - Removed struct sip_pvt randdata stringfield.
 - Remove useless (and inconsistent) 'header' suffix on variables in
   handle_request_subscribe.
 - Use ast_strdupa on Event header in handle_request_subscribe to avoid
   overly complicated strncmp calls to find the event package.
 - Move get_destination check in handle_request_subscribe to avoid
   duplicate checking for packages that don't need it.
 - Move extension state callback management in handle_request_subscribe
   to avoid duplicate checking for packages that don't need it.
 - Remove duplicate append_date prototype.
 - Rename append_date -> add_date to match other add_xxx functions.
 - Added add_expires helper function, removed code that manually added
   expires header.
 - Remove _header suffix on add_diversion_header (no other header adding
   functions have this).
 - Don't pass req->debug to request handle_request_XXXXX handlers if req
   is also being passed.
 - Don't pass req->ignore to check_auth as req is already being passed.
 - Don't create a subscription in handle_request_subscribe if
   p->expiry == 0.
 - Don't walk of the back of referred_by_name when splitting string in
   get_refer_info
 - Remove duplicate check for no dialog in handle_incoming when
   sipmethod == SIP_REFER, handle_request_refer checks for that.

Review: https://reviewboard.asterisk.org/r/1993/
Patch-by: gareth


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370636 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2 files changed, 32 insertions, 32 deletions

diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index 76766fb17..6f75f97e3 100644
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -785,14 +785,14 @@ struct sip_socket {
  * Then call parse_request() and req.method = find_sip_method();
  * to initialize the other fields. The \r\n at the end of each line is
  * replaced by \0, so that data[] is not a conforming SIP message anymore.
- * After this processing, rlPart1 is set to non-NULL to remember
+ * After this processing, rlpart1 is set to non-NULL to remember
  * that we can run get_header() on this kind of packet.
  *
  * parse_request() splits the first line as follows:
  * Requests have in the first line      method uri SIP/2.0
- *      rlPart1 = method; rlPart2 = uri;
+ *      rlpart1 = method; rlpart2 = uri;
  * Responses have in the first line     SIP/2.0 NNN description
- *      rlPart1 = SIP/2.0; rlPart2 = NNN + description;
+ *      rlpart1 = SIP/2.0; rlpart2 = NNN + description;
  *
  * For outgoing packets, we initialize the fields with init_req() or init_resp()
  * (which fills the first line to "METHOD uri SIP/2.0" or "SIP/2.0 code text"),
@@ -802,8 +802,8 @@ struct sip_socket {
  * \endverbatim
  */
 struct sip_request {
-	ptrdiff_t rlPart1;      /*!< Offset of the SIP Method Name or "SIP/2.0" protocol version */
-	ptrdiff_t rlPart2;      /*!< Offset of the Request URI or Response Status */
+	ptrdiff_t rlpart1;      /*!< Offset of the SIP Method Name or "SIP/2.0" protocol version */
+	ptrdiff_t rlpart2;      /*!< Offset of the Request URI or Response Status */
 	int headers;            /*!< # of SIP Headers */
 	int method;             /*!< Method of this request */
 	int lines;              /*!< Body Content */
@@ -824,7 +824,7 @@ struct sip_request {
 
 /* \brief given a sip_request and an offset, return the char * that resides there
  *
- * It used to be that rlPart1, rlPart2, and the header and line arrays were character
+ * It used to be that rlpart1, rlpart2, and the header and line arrays were character
  * pointers. They are now offsets into the ast_str portion of the sip_request structure.
  * To avoid adding a bunch of redundant pointer arithmetic to the code, this macro is
  * provided to retrieve the string at a particular offset within the request's buffer
@@ -918,26 +918,27 @@ struct _map_x_s {
 	const char *s;
 };
 
-/*! \brief Structure to handle SIP transfers. Dynamically allocated when needed
-	\note OEJ: Should be moved to string fields */
+/*! \brief Structure to handle SIP transfers. Dynamically allocated when needed */
 struct sip_refer {
-	char refer_to[AST_MAX_EXTENSION];             /*!< Place to store REFER-TO extension */
-	char refer_to_domain[AST_MAX_EXTENSION];      /*!< Place to store REFER-TO domain */
-	char refer_to_urioption[AST_MAX_EXTENSION];   /*!< Place to store REFER-TO uri options */
-	char refer_to_context[AST_MAX_EXTENSION];     /*!< Place to store REFER-TO context */
-	char referred_by[AST_MAX_EXTENSION];          /*!< Place to store REFERRED-BY extension */
-	char referred_by_name[AST_MAX_EXTENSION];     /*!< Place to store REFERRED-BY extension */
-	char refer_contact[AST_MAX_EXTENSION];        /*!< Place to store Contact info from a REFER extension */
-	char replaces_callid[SIPBUFSIZE];             /*!< Replace info: callid */
-	char replaces_callid_totag[SIPBUFSIZE/2];     /*!< Replace info: to-tag */
-	char replaces_callid_fromtag[SIPBUFSIZE/2];   /*!< Replace info: from-tag */
-	struct sip_pvt *refer_call;                   /*!< Call we are referring. This is just a reference to a
-	                                               * dialog owned by someone else, so we should not destroy
-	                                               * it when the sip_refer object goes.
-	                                               */
-	int attendedtransfer;                         /*!< Attended or blind transfer? */
-	int localtransfer;                            /*!< Transfer to local domain? */
-	enum referstatus status;                      /*!< REFER status */
+	AST_DECLARE_STRING_FIELDS(
+		AST_STRING_FIELD(refer_to);             /*!< Place to store REFER-TO extension */
+		AST_STRING_FIELD(refer_to_domain);      /*!< Place to store REFER-TO domain */
+		AST_STRING_FIELD(refer_to_urioption);   /*!< Place to store REFER-TO uri options */
+		AST_STRING_FIELD(refer_to_context);     /*!< Place to store REFER-TO context */
+		AST_STRING_FIELD(referred_by);          /*!< Place to store REFERRED-BY extension */
+		AST_STRING_FIELD(referred_by_name);     /*!< Place to store REFERRED-BY extension */
+		AST_STRING_FIELD(refer_contact);        /*!< Place to store Contact info from a REFER extension */
+		AST_STRING_FIELD(replaces_callid);      /*!< Replace info: callid */
+		AST_STRING_FIELD(replaces_callid_totag);   /*!< Replace info: to-tag */
+		AST_STRING_FIELD(replaces_callid_fromtag); /*!< Replace info: from-tag */
+	);
+	struct sip_pvt *refer_call;                     /*!< Call we are referring. This is just a reference to a
+							 * dialog owned by someone else, so we should not destroy
+							 * it when the sip_refer object goes.
+							 */
+	int attendedtransfer;                           /*!< Attended or blind transfer? */
+	int localtransfer;                              /*!< Transfer to local domain? */
+	enum referstatus status;                        /*!< REFER status */
 };
 
 /*! \brief Struct to handle custom SIP notify requests. Dynamically allocated when needed */
@@ -1006,7 +1007,6 @@ struct sip_pvt {
 		AST_STRING_FIELD(callid);       /*!< Global CallID */
 		AST_STRING_FIELD(initviabranch); /*!< The branch ID from the topmost Via header in the initial request */
 		AST_STRING_FIELD(initviasentby); /*!< The sent-by from the topmost Via header in the initial request */
-		AST_STRING_FIELD(randdata);     /*!< Random data */
 		AST_STRING_FIELD(accountcode);  /*!< Account code */
 		AST_STRING_FIELD(realm);        /*!< Authorization realm */
 		AST_STRING_FIELD(nonce);        /*!< Authorization nonce */
@@ -1304,9 +1304,9 @@ struct sip_peer {
 	struct sip_auth_container *auth;/*!< Realm authentication credentials */
 	int amaflags;                   /*!< AMA Flags (for billing) */
 	int callingpres;                /*!< Calling id presentation */
-	int inUse;                      /*!< Number of calls in use */
-	int inRinging;                  /*!< Number of calls ringing */
-	int onHold;                     /*!< Peer has someone on hold */
+	int inuse;                      /*!< Number of calls in use */
+	int ringing;                    /*!< Number of calls ringing */
+	int onhold;                     /*!< Peer has someone on hold */
 	int call_limit;                 /*!< Limit of concurrent calls */
 	int t38_maxdatagram;            /*!< T.38 FaxMaxDatagram override */
 	int busy_level;                 /*!< Level of active channels where we signal busy */
diff --git a/channels/sip/security_events.c b/channels/sip/security_events.c
index b4433d007..cde9f12f3 100644
--- a/channels/sip/security_events.c
+++ b/channels/sip/security_events.c
@@ -125,7 +125,7 @@ void sip_report_inval_password(const struct sip_pvt *p, const char *response_cha
                 },
                 .common.session_id  = session_id,
 
-		.challenge	    = p->randdata,
+		.challenge	    = p->nonce,
 		.received_challenge = response_challenge,
 		.received_hash	    = response_hash,
         };
@@ -206,7 +206,7 @@ void sip_report_failed_challenge_response(const struct sip_pvt *p, const char *r
                 },
                 .common.session_id = session_id,
 
-                .challenge         = p->randdata,
+                .challenge         = p->nonce,
                 .response          = response,
                 .expected_response = expected_response,
         };
@@ -242,7 +242,7 @@ void sip_report_chal_sent(const struct sip_pvt *p)
                 },
                 .common.session_id = session_id,
 
-                .challenge         = p->randdata,
+                .challenge         = p->nonce,
         };
 
 	if (!ast_strlen_zero(p->from)) { /* When dialing, show account making call */