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Kinsey Moore e5210366e4 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		13 years ago
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h323	Asterisk media architecture conversion - no more format bitfields	15 years ago
misdn	Multiple revisions 369001-369002	13 years ago
sip	Clean up chan_sip	13 years ago
Makefile	Extract sig_ss7 out of chan_dahdi.	15 years ago
chan_agent.c	Add some additional documentation for core AMI events	13 years ago
chan_alsa.c	Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)	13 years ago
chan_bridge.c	Finalize ast_channel opaquification	13 years ago
chan_console.c	Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)	13 years ago
chan_dahdi.c	Add hangupcause translation support	13 years ago
chan_gtalk.c	Add a new unified Jingle, Google Jingle, and Google Talk channel driver written from scratch called chan_motif.	13 years ago
chan_h323.c	Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)	13 years ago
chan_iax2.c	chan_iax2: Fix a segfault introduced by call ID logging	13 years ago
chan_jingle.c	Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)	13 years ago
chan_local.c	Add some additional documentation for core AMI events	13 years ago
chan_mgcp.c	Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)	13 years ago
chan_misdn.c	Release B channel allocation on error path in chan_misdn.	13 years ago
chan_motif.c	Do not consider failure to read the configuration file in chan_motif to be a show stopper for loading Asterisk by returning decline instead of failure.	13 years ago
chan_multicast_rtp.c	Opaquification for ast_format structs in struct ast_channel	13 years ago
chan_nbs.c	Opaquification for ast_format structs in struct ast_channel	13 years ago
chan_oss.c	chan_oss: fix "sample rate" error message	13 years ago
chan_phone.c	Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)	13 years ago
chan_phone.h	rename DialTone.h to chan_phone.h because chan_phone.c is the only file using it	17 years ago
chan_sip.c	Clean up chan_sip	13 years ago
chan_sip.exports.in	Merge changes dealing with support for Digium phones.	13 years ago
chan_skinny.c	Various small chan_skinny fixes and cleanup	13 years ago
chan_unistim.c	Remove code, that operate with cdr in attempt_transfer(). That was removed somewhere between 1.2 and 1.4 and acidentaly put back in chan_unistim.	13 years ago
chan_vpb.cc	Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)	13 years ago
console_board.c	Multiple revisions 369001-369002	13 years ago
console_gui.c	Multiple revisions 369001-369002	13 years ago
console_video.c	Multiple revisions 369001-369002	13 years ago
console_video.h	implement the 'freeze' function for incoming frames;	17 years ago
iax2-parser.c	Multiple revisions 369001-369002	13 years ago
iax2-parser.h	Asterisk media architecture conversion - no more format bitfields	15 years ago
iax2-provision.c	Multiple revisions 369001-369002	13 years ago
iax2-provision.h	Don't reload a configuration file if nothing has changed.	18 years ago
iax2.h	Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff	14 years ago
misdn_config.c	Multiple revisions 369001-369002	13 years ago
sig_analog.c	Add hangupcause translation support	13 years ago
sig_analog.h	push 'outgoing' flag from sig_XXX up to chan_dahdi	13 years ago
sig_pri.c	Add hangupcause translation support	13 years ago
sig_pri.h	Clear ISDN channel resetting state if the peer continues to use it.	13 years ago
sig_ss7.c	Add hangupcause translation support	13 years ago
sig_ss7.h	Setup DSP when SS7 call is connected or early media is available.	13 years ago
vcodecs.c	Multiple revisions 369001-369002	13 years ago
vgrabbers.c	Multiple revisions 369001-369002	13 years ago