chan_mgcp: Remove deprecated module.

Also removes res_pktcops to avoid merge conflicts
with ASTERISK~30301.

ASTERISK-30299

Change-Id: I41a316d327646a197b6f112f7f637aceb5111b41

channels/Makefile

@@ -31,7 +31,6 @@ $(call MOD_ADD_C,chan_pjsip,$(wildcard pjsip/*.c))
 $(call MOD_ADD_C,chan_dahdi,$(wildcard dahdi/*.c) sig_analog.c sig_pri.c sig_ss7.c)
 
 chan_dahdi.o: _ASTCFLAGS+=$(call get_menuselect_cflags,LOTS_OF_SPANS)
-chan_mgcp.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
 chan_unistim.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
 chan_phone.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
 chan_sip.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)

configs/samples/cli_aliases.conf.sample

@@ -174,7 +174,6 @@ agi dumphtml=agi dump html
 ael debug=ael set debug
 funcdevstate list=devstate list
 sip history=sip set history on
-mgcp set debug=mgcp set debug on
 abort shutdown=core abort shutdown
 stop now=core stop now
 stop gracefully=core stop gracefully

configs/samples/extensions.ael.sample

@@ -423,7 +423,6 @@ context ael-default {
 //       };
 
 // 6361 => Dial(IAX2/JaneDoe,,rm);                // ring without time limit
-// 6389 => Dial(MGCP/aaln/1@192.168.0.14);
 // 6394 => Dial(Local/6275/n);                    // this will dial ${MARK}
 
 // 6275 => &ael-stdexten(6275,${MARK});           // assuming ${MARK} is something like DAHDI/2

configs/samples/extensions.conf.sample

@@ -223,7 +223,7 @@ TRUNKMSD=1					; MSD digits to strip (usually 1 or 0)
 ;
 ; so that dialtone remains even after dialing a 9.  Please note that ignorepat
 ; only works with channels which receive dialtone from the PBX, such as DAHDI,
-; Phone, and VPB.  Other channels, such as IAX2, PJSIP, SIP and MGCP, which generate
+; Phone, and VPB.  Other channels, such as IAX2, PJSIP, and SIP, which generate
 ; generate their own dialtone and converse with the PBX only after a number is
 ; complete, are generally unaffected by ignorepat (unless DISA or another method
 ; is used to generate a dialtone after answering the channel).
@@ -694,7 +694,6 @@ include => demo
 ;exten => 6245,s+1,Hangup			; s+1, same as n
 ;exten => 6245,dial+101,VoiceMail(6245,b)	; Voicemail (busy)
 ;exten => 6361,1,Dial(IAX2/JaneDoe,,rm)		; ring without time limit
-;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
 ;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
 ;exten => 6391,1,Dial(JINGLE/asterisk@digium.com/mogorman@astjab.org) ;Dial via jingle using asterisk as the transport and calling mogorman.
 ;exten => 6394,1,Dial(Local/6275/n)		; this will dial ${MARK}

configs/samples/mgcp.conf.sample

@@ -1,142 +0,0 @@
-;
-; MGCP Configuration for Asterisk
-;
-[general]
-;port = 2427
-;bindaddr = 0.0.0.0
-
-; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
-;tos=cs3		; Sets TOS for signaling packets.
-;tos_audio=ef		; Sets TOS for RTP audio packets.
-;cos=3			; Sets 802.1p priority for signaling packets.
-;cos_audio=5		; Sets 802.1p priority for RTP audio packets.
-
-; --------------------- DIGIT TIMEOUTS ----------------------------
-firstdigittimeout = 30000 ; default 16000 = 16s
-gendigittimeout = 10000   ; default  8000 = 8s
-matchdigittimeout = 5000  ; defaults 3000 = 3s
-
-; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
-; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
-                              ; MGCP channel. Defaults to "no". An enabled jitterbuffer will
-                              ; be used only if the sending side can create and the receiving
-                              ; side can not accept jitter. The MGCP channel can accept jitter,
-                              ; thus an enabled jitterbuffer on the receive MGCP side will only
-                              ; be used if the sending side can create jitter and jbforce is
-                              ; also set to yes.
-
-; jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a MGCP
-                              ; channel. Defaults to "no".
-
-; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.
-
-; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
-                              ; resynchronized. Useful to improve the quality of the voice, with
-                              ; big jumps in/broken timestamps, usually sent from exotic devices
-                              ; and programs. Defaults to 1000.
-
-; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a MGCP
-                              ; channel. Two implementations are currently available - "fixed"
-                              ; (with size always equals to jbmax-size) and "adaptive" (with
-                              ; variable size, actually the new jb of IAX2). Defaults to fixed.
-
-; jbtargetextra = 40          ; This option only affects the jb when 'jbimpl = adaptive' is set.
-                              ; The option represents the number of milliseconds by which the new
-                              ; jitter buffer will pad its size. the default is 40, so without
-                              ; modification, the new jitter buffer will set its size to the jitter
-                              ; value plus 40 milliseconds. increasing this value may help if your
-                              ; network normally has low jitter, but occasionally has spikes.
-
-; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
-; ----------------------------------------------------------------------------------
-
-;[dlinkgw]
-;host = 192.168.0.64
-;context = default
-;directmedia = no
-;line => aaln/2
-;line => aaln/1
-
-;; The MGCP channel supports the following service codes:
-;; # - Transfer
-;; *67 - Calling Number Delivery Blocking
-;; *70 - Cancel Call Waiting
-;; *72 - Call Forwarding Activation
-;; *73 - Call Forwarding Deactivation
-;; *78 - Do Not Disturb Activation
-;; *79 - Do Not Disturb Deactivation
-;; *8 - Call pick-up
-;
-; known to work with Swissvoice IP10s
-;[192.168.1.20]
-;context=local
-;host=192.168.1.20
-;callerid = "John Doe" <123>
-;callgroup=0      ; in the range from 0 to 63
-;pickupgroup=0    ; in the range from 0 to 63
-;nat=no
-;threewaycalling=yes
-;transfer=yes     ; transfer requires threewaycalling=yes. Use FLASH to transfer
-;callwaiting=yes  ; this might be a cause of trouble for ip10s
-;cancallforward=yes
-;line => aaln/1
-;
-
-;[dph100]
-;
-; Supporting the DPH100M requires defining DLINK_BUGGY_FIRMWARE in
-; chan_mgcp.c in addition to enabling the slowsequence mode due to
-; bugs in the D-Link firmware
-;
-;context=local
-;host=dynamic
-;dtmfmode=none		; DTMF Mode can be 'none', 'rfc2833', or 'inband' or
-			; 'hybrid' which starts in none and moves to inband.  Default is none.
-;slowsequence=yes	; The DPH100M does not follow MGCP standards for sequencing
-;line => aaln/1
-
-; known to work with wave7optics FTTH LMGs
-;[192.168.1.20]
-;accountcode	= 1000		; record this in cdr as account identification for billing
-;amaflags	= billing	; record this in cdr as flagged for 'billing',
-				; 'documentation', or 'omit'
-;context	= local
-;host		= 192.168.1.20
-;wcardep	= aaln/*	; enables wildcard endpoint and sets it to 'aaln/*'
-				; another common format is '*'
-;callerid 	= "Duane Cox" <123>	; now lets setup line 1 using per endpoint configuration...
-;callwaiting	= no
-;callreturn	= yes
-;cancallforward = yes
-;directmedia	= no
-;transfer	= no
-;dtmfmode	= inband
-;setvar=one=1           ; Set channel variables associated with this incoming line
-;setvar=two=2
-;line => aaln/1			; now lets save this config to line1 aka aaln/1
-;clearvars=all          ; Reset list of variables back to none
-;callerid 	= "Duane Cox" <456>	; now lets setup line 2
-;callwaiting	= no
-;callreturn	= yes
-;cancallforward = yes
-;directmedia	= no
-;transfer	= no
-;dtmfmode	= inband
-;line => aaln/2			; now lets save this config to line2 aka aaln/2
-
-; PacketCable
-;[sbv5121e-mta.test.local]
-;host        = 10.0.1.3
-;callwaiting = 1
-;canreinvite = 1
-;dtmfmode    = rfc2833
-;amaflags    = BILLING
-;ncs         = yes ; Use NCS 1.0 signalling
-;pktcgatealloc = yes ; Allocate DQOS gate on CMTS
-;hangupongateremove = yes ; Hangup the channel if the CMTS close the gate
-;callerid    = 3622622225
-;accountcode = test-3622622225
-;line        = aaln/1
-;callerid    = 3622622226
-;accountcode = test-3622622226
-;line        = aaln/2

configs/samples/res_pktccops.conf.sample

@@ -1,32 +0,0 @@
-;; Sample res_pktccops.conf
-;
-;[general]
-;gateinfoperiod => 60 ; default 60s
-;gatetimeout = 150 ; default 150
-;t1 => 250 ; default 250s
-;t7 => 200 ; default 200s
-;t8 => 300 ; default 300s
-;keepalive => 60 ; default 60s
-;
-;[teszt]
-;host => 192.168.0.24
-;pool => 10.0.1.0 10.0.1.255
-;pool => 10.0.3.0 10.0.3.255
-;pool => 10.0.7.0 10.0.8.255
-;pool => 10.0.10.0 10.0.11.255
-;
-;[general]
-;gateinfoperiod => 60 ; default 60s
-;gatetimeout = 150 ; default 150
-;t1 => 250 ; default 250s
-;t7 => 200 ; default 200s
-;t8 => 300 ; default 300s
-;keepalive => 60 ; default 60s
-;
-;[test]
-;host => 192.168.0.24
-;pool => 10.0.1.0 10.0.1.255
-;pool => 10.0.3.0 10.0.3.255
-;pool => 10.0.7.0 10.0.8.255
-;pool => 10.0.10.0 10.0.11.255
-;

configure

@@ -34779,101 +34779,6 @@ printf "%s\n" "#define HAVE_LINUX_COMPILER_H 1" >>confdefs.h
 fi
 
 
-# Used in res/res_pktccops
-
-    if test "x${PBX_MSG_NOSIGNAL}" != "x1"; then
-	{ printf "%s\n" "$as_me:${as_lineno-$LINENO}: checking for MSG_NOSIGNAL in sys/socket.h" >&5
-printf %s "checking for MSG_NOSIGNAL in sys/socket.h... " >&6; }
-	saved_cppflags="${CPPFLAGS}"
-	if test "x${MSG_NOSIGNAL_DIR}" != "x"; then
-	    MSG_NOSIGNAL_INCLUDE="-I${MSG_NOSIGNAL_DIR}/include"
-	fi
-	CPPFLAGS="${CPPFLAGS} ${MSG_NOSIGNAL_INCLUDE}"
-
-	cat confdefs.h - <<_ACEOF >conftest.$ac_ext
-/* end confdefs.h.  */
- #include <sys/socket.h>
-int
-main (void)
-{
-#if defined(MSG_NOSIGNAL)
-				int foo = 0;
-			        #else
-			        int foo = bar;
-			        #endif
-				0
-
-  ;
-  return 0;
-}
-_ACEOF
-if ac_fn_c_try_compile "$LINENO"
-then :
-     { printf "%s\n" "$as_me:${as_lineno-$LINENO}: result: yes" >&5
-printf "%s\n" "yes" >&6; }
-		PBX_MSG_NOSIGNAL=1
-
-printf "%s\n" "#define HAVE_MSG_NOSIGNAL 1" >>confdefs.h
-
-
-
-else $as_nop
-     { printf "%s\n" "$as_me:${as_lineno-$LINENO}: result: no" >&5
-printf "%s\n" "no" >&6; }
-
-fi
-rm -f core conftest.err conftest.$ac_objext conftest.beam conftest.$ac_ext
-	CPPFLAGS="${saved_cppflags}"
-    fi
-
-
-
-    if test "x${PBX_SO_NOSIGPIPE}" != "x1"; then
-	{ printf "%s\n" "$as_me:${as_lineno-$LINENO}: checking for SO_NOSIGPIPE in sys/socket.h" >&5
-printf %s "checking for SO_NOSIGPIPE in sys/socket.h... " >&6; }
-	saved_cppflags="${CPPFLAGS}"
-	if test "x${SO_NOSIGPIPE_DIR}" != "x"; then
-	    SO_NOSIGPIPE_INCLUDE="-I${SO_NOSIGPIPE_DIR}/include"
-	fi
-	CPPFLAGS="${CPPFLAGS} ${SO_NOSIGPIPE_INCLUDE}"
-
-	cat confdefs.h - <<_ACEOF >conftest.$ac_ext
-/* end confdefs.h.  */
- #include <sys/socket.h>
-int
-main (void)
-{
-#if defined(SO_NOSIGPIPE)
-				int foo = 0;
-			        #else
-			        int foo = bar;
-			        #endif
-				0
-
-  ;
-  return 0;
-}
-_ACEOF
-if ac_fn_c_try_compile "$LINENO"
-then :
-     { printf "%s\n" "$as_me:${as_lineno-$LINENO}: result: yes" >&5
-printf "%s\n" "yes" >&6; }
-		PBX_SO_NOSIGPIPE=1
-
-printf "%s\n" "#define HAVE_SO_NOSIGPIPE 1" >>confdefs.h
-
-
-
-else $as_nop
-     { printf "%s\n" "$as_me:${as_lineno-$LINENO}: result: no" >&5
-printf "%s\n" "no" >&6; }
-
-fi
-rm -f core conftest.err conftest.$ac_objext conftest.beam conftest.$ac_ext
-	CPPFLAGS="${saved_cppflags}"
-    fi
-
-
 
 
 		if test "x${PBX_SDL}" != "x1" -a "${USE_SDL}" != "no"; then

configure.ac

@@ -2786,9 +2786,6 @@ fi
 AC_CHECK_HEADER([linux/compiler.h],
                 [AC_DEFINE_UNQUOTED([HAVE_LINUX_COMPILER_H], 1, [Define to 1 if your system has linux/compiler.h.])])
 
-# Used in res/res_pktccops
-AST_C_DEFINE_CHECK([MSG_NOSIGNAL], [MSG_NOSIGNAL], [sys/socket.h])
-AST_C_DEFINE_CHECK([SO_NOSIGPIPE], [SO_NOSIGPIPE], [sys/socket.h])
 
 AST_EXT_TOOL_CHECK([SDL], [sdl-config])
 AST_EXT_LIB_CHECK([SDL_IMAGE], [SDL_image], [IMG_Load], [SDL_image.h], [${SDL_LIB}], [${SDL_INCLUDE}])

doc/UPGRADE-staging/chan_mgcp_removal.txt

@@ -0,0 +1,7 @@
+Subject: chan_mgcp
+Master-Only: True
+
+This module was deprecated in Asterisk 19
+and is now being removed in accordance with
+the Asterisk Module Deprecation policy.
+

doc/asterisk.8

@@ -45,8 +45,8 @@ asterisk
 provides Private Branch eXchange (PBX), Interactive Voice Response (IVR),
 Automated Call Distribution (ACD), Voice over IP (VoIP) gatewaying,
 Conferencing, and a plethora of other telephony applications to a broad
-range of telephony devices including packet voice (SIP, IAX2, MGCP,
+range of telephony devices including packet voice (SIP, IAX2, H.323, Unistim)
-H.323, Unistim) devices (both endpoints and proxies), as well as traditional TDM
+devices (both endpoints and proxies), as well as traditional TDM
 hardware including T1, E1, ISDN PRI, GR-303, RBS, Loopstart, Groundstart,
 ISDN BRI and many more.
 .PP

doc/asterisk.sgml

@@ -52,8 +52,8 @@
    provides Private Branch eXchange (PBX), Interactive Voice Response (IVR),
    Automated Call Distribution (ACD), Voice over IP (VoIP) gatewaying,
    Conferencing, and a plethora of other telephony applications to a broad
-   range of telephony devices including packet voice (SIP, IAX2, MGCP,
+   range of telephony devices including packet voice (SIP, IAX2 H.323, Unistim)
-   H.323, Unistim) devices (both endpoints and proxies), as well as traditional TDM
+   devices (both endpoints and proxies), as well as traditional TDM
    hardware including T1, E1, ISDN PRI, GR-303, RBS, Loopstart, Groundstart,
    ISDN BRI and many more.
   </para>

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 \ref chan_mgcp.c (and various H.323 channels)
+ * Used in \ref chan_sip.c (and various H.323 channels)
  * \section rtpconf rtp.conf
  * \verbinclude rtp.conf.sample
  */

include/asterisk/pktccops.h

@@ -1,82 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2009, Attila Domjan
- *
- * Attila Domjan <attila.domjan.hu@gmail.com>
- *
- * 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 PacketCable COPS
- *
- * \author Attila Domjan <attila.domjan.hu@gmail.com>
- */
-
-#ifndef _ASTERISK_PKTCCOPS_H
-#define _ASTERISK_PKTCCOPS_H
-
-#include "asterisk/optional_api.h"
-
-#if defined(__cplusplus) || defined(c_plusplus)
-extern "C" {
-#endif
-
-enum {
-	GATE_SET,
-	GATE_INFO,
-	GATE_SET_HAVE_GATEID,
-	GATE_DEL
-};
-
-enum {
-	GATE_ALLOC_FAILED,
-	GATE_ALLOC_PROGRESS,
-	GATE_ALLOCATED,
-	GATE_CLOSED,
-	GATE_CLOSED_ERR,
-	GATE_OPEN,
-	GATE_DELETED,
-	GATE_TIMEOUT
-};
-
-struct cops_gate {
-	AST_LIST_ENTRY(cops_gate) list;
-	uint32_t gateid;
-	uint16_t trid;
-	time_t in_transaction;
-	uint32_t mta;
-	int state;
-	time_t allocated;
-	time_t checked;
-	time_t deltimer;
-	struct cops_cmts *cmts;
-	int (* got_dq_gi) (struct cops_gate *gate);
-	int (* gate_remove) (struct cops_gate *gate);
-	int (* gate_open) (struct cops_gate *gate);
-	void *tech_pvt;
-};
-
-
-AST_OPTIONAL_API(struct cops_gate *, ast_pktccops_gate_alloc,
-	(int cmd,  struct cops_gate *gate, uint32_t mta, uint32_t actcount,
-	 float bitrate, uint32_t psize, uint32_t ssip, uint16_t ssport,
-	 int (* const got_dq_gi) (struct cops_gate *gate),
-	 int (* const gate_remove) (struct cops_gate *gate)),
-	{ return NULL; });
-
-#if defined(__cplusplus) || defined(c_plusplus)
-}
-#endif
-
-#endif /* _ASTERISK_PKTCCOPS_H */

include/asterisk/res_hep.h

@@ -42,7 +42,6 @@ enum hepv3_capture_type {
 	HEPV3_CAPTURE_TYPE_SDP    = 0x03,
 	HEPV3_CAPTURE_TYPE_RTP    = 0x04,
 	HEPV3_CAPTURE_TYPE_RTCP   = 0x05,
-	HEPV3_CAPTURE_TYPE_MGCP   = 0x06,
 	HEPV3_CAPTURE_TYPE_MEGACO = 0x07,
 	HEPV3_CAPTURE_TYPE_M2UA   = 0x08,
 	HEPV3_CAPTURE_TYPE_M3UA   = 0x09,

main/asterisk.dynamics

@@ -2,7 +2,6 @@
 	*ast_adsi_*;
 	*ast_agi_*;
 	*ast_beep_*;
-	*ast_pktccops_*;
 	*ast_smdi_*;
 	*ast_monitor_*;
 	*ast_key_get;

menuselect/example_menuselect-tree

@@ -198,8 +198,6 @@
 		</member>
 		<member name="chan_local" displayname="Local Proxy Channel" remove_on_change="channels/chan_local.o channels/chan_local.so">
 		</member>
-		<member name="chan_mgcp" displayname="Media Gateway Control Protocol (MGCP)" remove_on_change="channels/chan_mgcp.o channels/chan_mgcp.so">
-		</member>
 		<member name="chan_sip" displayname="Session Initiation Protocol (SIP)" remove_on_change="channels/chan_sip.o channels/chan_sip.so">
 		</member>
 		<member name="chan_zap" displayname="Zapata Telephony" remove_on_change="channels/chan_zap.o channels/chan_zap.so">

menuselect/test/menuselect-tree

@@ -228,8 +228,6 @@
 </member>
 <member name="chan_local" displayname="Local Proxy Channel (Note: used internally by other modules)" remove_on_change="channels/chan_local.o channels/chan_local.so">
 </member>
-<member name="chan_mgcp" displayname="Media Gateway Control Protocol (MGCP)" remove_on_change="channels/chan_mgcp.o channels/chan_mgcp.so">
-</member>
 <member name="chan_sip" displayname="Session Initiation Protocol (SIP)" remove_on_change="channels/chan_sip.o channels/chan_sip.so">
         <depend>chan_local</depend>
 </member>

res/res_pktccops.exports.in

@@ -1,6 +0,0 @@
-{
-	global:
-		LINKER_SYMBOL_PREFIXast_pktccops_gate_alloc;
-	local:
-		*;
-};

tests/CI/buildAsterisk.sh

@@ -156,13 +156,13 @@ if [ $NO_MENUSELECT -eq 0 ] ; then
 		mod_disables+=" cdr_adaptive_odbc cdr_custom cdr_manager cdr_odbc cdr_pgsql cdr_radius"
 		mod_disables+=" cdr_tds"
 		mod_disables+=" cel_odbc cel_pgsql cel_radius cel_sqlite3_custom cel_tds"
-		mod_disables+=" chan_alsa chan_console chan_mgcp chan_motif chan_rtp chan_unistim"
+		mod_disables+=" chan_alsa chan_console chan_motif chan_rtp chan_unistim"
 		mod_disables+=" func_frame_trace func_pitchshift func_speex func_volume func_dialgroup"
 		mod_disables+=" func_periodic_hook func_sprintf func_enum func_extstate func_sysinfo func_iconv"
 		mod_disables+=" func_callcompletion func_version func_rand func_sha1 func_module func_md5"
 		mod_disables+=" pbx_dundi pbx_loopback"
 		mod_disables+=" res_ael_share res_calendar res_config_ldap res_config_pgsql res_corosync"
-		mod_disables+=" res_http_post res_pktccops res_rtp_multicast res_snmp res_xmpp"
+		mod_disables+=" res_http_post res_rtp_multicast res_snmp res_xmpp"
 	fi
 
 	runner menuselect/menuselect `gen_mods disable $mod_disables` menuselect.makeopts