res_pjsip_session: Delay sending BYE if a re-INVITE transaction is in progress.

Given the scenario where a PJSIP channel is in a native RTP bridge with direct
media and the channel is then hung up the code will currently re-INVITE the channel
back to Asterisk and send a BYE at the same time. Many SIP implementations dislike
this greatly.

This change makes it so that if a re-INVITE transaction is in progress the BYE
is queued to occur after the completion of the transaction (be it through normal
means or a timeout).

Review: https://reviewboard.asterisk.org/r/4248/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429409 65c4cc65-6c06-0410-ace0-fbb531ad65f3

include/asterisk/res_pjsip_session.h

@@ -433,6 +433,14 @@ struct ast_sip_session *ast_sip_session_create_outgoing(struct ast_sip_endpoint
 	struct ast_sip_contact *contact, const char *location, const char *request_user,
 	struct ast_format_cap *req_caps);
 
+/*!
+ * \brief Terminate a session and, if possible, send the provided response code
+ *
+ * \param session The session to terminate
+ * \param response The response code to use for termination if possible
+ */
+void ast_sip_session_terminate(struct ast_sip_session *session, int response);
+
 /*!
  * \brief Defer local termination of a session until remote side terminates, or an amount of time passes
  *