[svn-commits] coreyfarrell: trunk r418448 - in /trunk: configs/ include/asterisk/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Sun Jul 13 00:05:57 CDT 2014
Author: coreyfarrell
Date: Sun Jul 13 00:05:49 2014
New Revision: 418448
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=418448
Log:
Remove files left behind on removal of h323, jingle and jabber.
This change removes h323.conf.sample, jingle.h, jabber.h left behind by r3698.
Review: https://reviewboard.asterisk.org/r/3755/
Modified:
trunk/configs/h323.conf.sample
trunk/include/asterisk/jabber.h
trunk/include/asterisk/jingle.h
Modified: trunk/configs/h323.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/h323.conf.sample?view=diff&rev=418448&r1=418447&r2=418448
==============================================================================
--- trunk/configs/h323.conf.sample (original)
+++ trunk/configs/h323.conf.sample Sun Jul 13 00:05:49 2014
@@ -1,210 +1,0 @@
-; The NuFone Network's
-; Open H.323 driver configuration
-;
-[general]
-port = 1720
-;bindaddr = 1.2.3.4 ; this SHALL contain a single, valid IP address for this machine
-;
-; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
-;tos_audio=ef ; Sets TOS for RTP audio packets.
-;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
-;
-; You may specify a global default AMA flag for iaxtel calls. It must be
-; one of 'default', 'omit', 'billing', or 'documentation'. These flags
-; are used in the generation of call detail records.
-;
-;amaflags = default
-;
-; You may specify a default account for Call Detail Records in addition
-; to specifying on a per-user basis
-;
-;accountcode=lss0101
-;
-; You can fine tune codecs here using "allow" and "disallow" clauses
-; with specific codecs. Use "all" to represent all formats.
-;
-;disallow=all
-;allow=all ; turns on all installed codecs
-;disallow=g723.1 ; Hm... Proprietary, don't use it...
-;allow=gsm ; Always allow GSM, it's cool :)
-;allow=ulaw ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
- ; for framing options
-;autoframing=yes ; Set packetization based on the remote endpoint's (ptime)
- ; preferences. Defaults to no.
-;
-; User-Input Mode (DTMF)
-;
-; valid entries are: rfc2833, inband, cisco, h245-signal
-; default is rfc2833
-;dtmfmode=rfc2833
-;
-; Default RTP Payload to send RFC2833 DTMF on. This is used to
-; interoperate with broken gateways which cannot successfully
-; negotiate a RFC2833 payload type in the TerminalCapabilitySet.
-; To specify required payload type, put it after colon in dtmfmode
-; option like
-;dtmfmode=rfc2833:101
-; or
-;dtmfmode=cisco:121
-;
-; Set the gatekeeper
-; DISCOVER - Find the Gk address using multicast
-; DISABLE - Disable the use of a GK
-; <IP address> or <Host name> - The acutal IP address or hostname of your GK
-;gatekeeper = DISABLE
-;
-;
-; Tell Asterisk whether or not to accept Gatekeeper
-; routed calls or not. Normally this should always
-; be set to yes, unless you want to have finer control
-; over which users are allowed access to Asterisk.
-; Default: YES
-;
-;AllowGKRouted = yes
-;
-; When the channel works without gatekeeper, there is possible to
-; reject calls from anonymous (not listed in users) callers.
-; Default is to allow anonymous calls.
-;
-;AcceptAnonymous = yes
-;
-; Optionally you can determine a user by Source IP versus its H.323 alias.
-; Default behavour is to determine user by H.323 alias.
-;
-;UserByAlias=no
-;
-; Default context gets used in siutations where you are using
-; the GK routed model or no type=user was found. This gives you
-; the ability to either play an invalid message or to simply not
-; use user authentication at all.
-;
-;context=default
-;
-; Use this option to help Cisco (or other) gateways to setup backward voice
-; path to pass inband tones to calling user (see, for example,
-; http://www.cisco.com/warp/public/788/voip/ringback.html)
-;
-; Add PROGRESS information element to SETUP message sent on outbound calls
-; to notify about required backward voice path. Valid values are:
-; 0 - don't add PROGRESS information element (default);
-; 1 - call is not end-end ISDN, further call progress information can
-; possibly be available in-band;
-; 3 - origination address is non-ISDN (Cisco accepts this value only);
-; 8 - in-band information or an appropriate pattern is now available;
-;progress_setup = 3
-;
-; Add PROGRESS information element (IE) to ALERT message sent on incoming
-; calls to notify about required backwared voice path. Valid values are:
-; 0 - don't add PROGRESS IE (default);
-; 8 - in-band information or an appropriate pattern is now available;
-;progress_alert = 8
-;
-; Generate PROGRESS message when H.323 audio path has established to create
-; backward audio path at other end of a call.
-;progress_audio = yes
-;
-; Specify how to inject non-standard information into H.323 messages. When
-; the channel receives messages with tunneled information, it automatically
-; enables the same option for all further outgoing messages independedly on
-; options has been set by the configuration. This behavior is required, for
-; example, for Cisco CallManager when Q.SIG tunneling is enabled for a
-; gateway where Asterisk lives.
-; The option can be used multiple times, one option per line.
-;tunneling=none ; Totally disable tunneling (default)
-;tunneling=cisco ; Enable Cisco-specific tunneling
-;tunneling=qsig ; Enable tunneling via Q.SIG messages
-;
-; Specify how to pass hold notification to remote party. Default is to
-; use H.450.4 supplementary service message.
-;hold=none ; Do not pass hold/retrieve notifications
-;hold=notify ; Use H.225 NOTIFY message
-;hold=q931only ; Use stripped H.225 NOTIFY message (Q.931 part
-; ; only, usable for Cisco CallManager)
-;hold=h450 ; Pass notification as H.450.4 supplementary
-; ; service
-;
-;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
-; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
- ; H323 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 H323 channel can accept jitter,
- ; thus an enabled jitterbuffer on the receive H323 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 H323
- ; 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, usualy sent from exotic devices
- ; and programs. Defaults to 1000.
-
-; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a H323
- ; channel. Two implementations are currenlty available - "fixed"
- ; (with size always equals to jbmax-size) and "adaptive" (with
- ; variable size, actually the new jb of IAX2). Defaults to fixed.
-
-; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
-;-----------------------------------------------------------------------------------
-;
-; H.323 Alias definitions
-;
-; Type 'h323' will register aliases to the endpoint
-; and Gatekeeper, if there is one.
-;
-; Example: if someone calls time at your.asterisk.box.com
-; Asterisk will send the call to the extension 'time'
-; in the context default
-;
-; [default]
-; exten => time,1,Answer
-; exten => time,2,Playback,current-time
-;
-; Keyword's 'prefix' and 'e164' are only make sense when
-; used with a gatekeeper. You can specify either a prefix
-; or E.164 this endpoint is responsible for terminating.
-;
-; Example: The H.323 alias 'det-gw' will tell the gatekeeper
-; to route any call with the prefix 1248 to this alias. Keyword
-; e164 is used when you want to specifiy a full telephone
-; number. So a call to the number 18102341212 would be
-; routed to the H.323 alias 'time'.
-;
-;[time]
-;type=h323
-;e164=18102341212
-;context=default
-;
-;[det-gw]
-;type=h323
-;prefix=1248,1313
-;context=detroit
-;
-;
-; Inbound H.323 calls from BillyBob would land in the incoming
-; context with a maximum of 4 concurrent incoming calls
-;
-;
-; Note: If keyword 'incominglimit' are omitted Asterisk will not
-; enforce any maximum number of concurrent calls.
-;
-;[BillyBob]
-;type=user
-;host=192.168.1.1
-;context=incoming
-;incominglimit=4
-;h245Tunneling=no
-;
-;
-; Outbound H.323 call to Larry using SlowStart
-;
-;[Larry]
-;type=peer
-;host=192.168.2.1
-;fastStart=no
-
-
-
Modified: trunk/include/asterisk/jabber.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/jabber.h?view=diff&rev=418448&r1=418447&r2=418448
==============================================================================
--- trunk/include/asterisk/jabber.h (original)
+++ trunk/include/asterisk/jabber.h Sun Jul 13 00:05:49 2014
@@ -1,224 +1,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2010, Digium, Inc.
- *
- * Matt O'Gorman <mogorman at digium.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 AJI - The Asterisk Jabber Interface
- * \arg \ref AJI_intro
- * \ref res_jabber.c
- * \author Matt O'Gorman <mogorman at digium.com>
- * IKSEMEL http://iksemel.jabberstudio.org
- *
- * \page AJI_intro AJI - The Asterisk Jabber Interface
- *
- * The Asterisk Jabber Interface, AJI, publishes an API for
- * modules to use jabber communication. res_jabber.c implements
- * a Jabber client and a component that can connect as a service
- * to Jabber servers.
- *
- * \section External dependencies
- * AJI use the IKSEMEL library found at http://iksemel.jabberstudio.org/
- *
- * \section Files
- * - res_jabber.c
- * - jabber.h
- * - chan_gtalk.c
- *
- */
-
-#ifndef _ASTERISK_JABBER_H
-#define _ASTERISK_JABBER_H
-
-#ifdef HAVE_OPENSSL
-
-#include <openssl/ssl.h>
-#include <openssl/err.h>
-#define TRY_SECURE 2
-#define SECURE 4
-
-#endif /* HAVE_OPENSSL */
-/* file is read by blocks with this size */
-#define NET_IO_BUF_SIZE 4096
-/* Return value for timeout connection expiration */
-#define IKS_NET_EXPIRED 12
-
-#include <iksemel.h>
-#include "asterisk/astobj.h"
-#include "asterisk/linkedlists.h"
-
-/*
- * As per RFC 3920 - section 3.1, the maximum length for a full Jabber ID
- * is 3071 bytes.
- * The ABNF syntax for jid :
- * jid = [node "@" ] domain [ "/" resource ]
- * Each allowable portion of a JID (node identifier, domain identifier,
- * and resource identifier) MUST NOT be more than 1023 bytes in length,
- * resulting in a maximum total size (including the '@' and '/' separators)
- * of 3071 bytes.
- */
-#define AJI_MAX_JIDLEN 3071
-#define AJI_MAX_RESJIDLEN 1023
-#define AJI_MAX_ATTRLEN 256
-
-#define MUC_NS "http://jabber.org/protocol/muc"
-
-enum aji_state {
- AJI_DISCONNECTING,
- AJI_DISCONNECTED,
- AJI_CONNECTING,
- AJI_CONNECTED
-};
-
-enum {
- AJI_AUTOPRUNE = (1 << 0),
- AJI_AUTOREGISTER = (1 << 1),
- AJI_AUTOACCEPT = (1 << 2),
-};
-
-enum {
- AJI_XEP0248 = (1 << 0),
- AJI_PUBSUB = (1 << 1),
- AJI_PUBSUB_AUTOCREATE = (1 << 2),
-};
-
-enum aji_btype {
- AJI_USER = 0,
- AJI_TRANS = 1,
- AJI_UTRANS = 2,
-};
-
-struct aji_version {
- char version[50];
- int jingle;
- struct aji_capabilities *parent;
- struct aji_version *next;
-};
-
-struct aji_capabilities {
- char node[200];
- struct aji_version *versions;
- struct aji_capabilities *next;
-};
-
-struct aji_resource {
- int status;
- char resource[AJI_MAX_RESJIDLEN];
- char *description;
- struct aji_version *cap;
- int priority;
- struct aji_resource *next;
-};
-
-struct aji_message {
- char *from;
- char *message;
- char id[25];
- struct timeval arrived;
- AST_LIST_ENTRY(aji_message) list;
-};
-
-struct aji_buddy {
- ASTOBJ_COMPONENTS_FULL(struct aji_buddy, AJI_MAX_JIDLEN, 1);
- char channel[160];
- struct aji_resource *resources;
- enum aji_btype btype;
- struct ast_flags flags;
-};
-
-struct aji_buddy_container {
- ASTOBJ_CONTAINER_COMPONENTS(struct aji_buddy);
-};
-
-struct aji_transport_container {
- ASTOBJ_CONTAINER_COMPONENTS(struct aji_transport);
-};
-
-struct aji_client {
- ASTOBJ_COMPONENTS(struct aji_client);
- char password[160];
- char user[AJI_MAX_JIDLEN];
- char serverhost[AJI_MAX_RESJIDLEN];
- char pubsub_node[AJI_MAX_RESJIDLEN];
- char statusmessage[256];
- char name_space[256];
- char sid[10]; /* Session ID */
- char mid[6]; /* Message ID */
- char context[AST_MAX_CONTEXT];
- iksid *jid;
- iksparser *p;
- iksfilter *f;
- ikstack *stack;
-#ifdef HAVE_OPENSSL
- SSL_CTX *ssl_context;
- SSL *ssl_session;
- const SSL_METHOD *ssl_method;
- unsigned int stream_flags;
-#endif /* HAVE_OPENSSL */
- enum aji_state state;
- int port;
- int debug;
- int usetls;
- int forcessl;
- int usesasl;
- int keepalive;
- int allowguest;
- int timeout;
- int message_timeout;
- int authorized;
- int distribute_events;
- int send_to_dialplan;
- struct ast_flags flags;
- int component; /* 0 client, 1 component */
- struct aji_buddy_container buddies;
- AST_LIST_HEAD(messages,aji_message) messages;
- void *jingle;
- pthread_t thread;
- int priority;
- enum ikshowtype status;
-};
-
-struct aji_client_container{
- ASTOBJ_CONTAINER_COMPONENTS(struct aji_client);
-};
-
-/* !Send XML stanza over the established XMPP connection */
-int ast_aji_send(struct aji_client *client, iks *x);
-/*! Send jabber chat message from connected client to jabber URI */
-int ast_aji_send_chat(struct aji_client *client, const char *address, const char *message);
-/*! Send jabber chat message from connected client to a groupchat using
- * a given nickname */
-int ast_aji_send_groupchat(struct aji_client *client, const char *nick, const char *address, const char *message);
-/*! Disconnect jabber client */
-int ast_aji_disconnect(struct aji_client *client);
-int ast_aji_check_roster(void);
-void ast_aji_increment_mid(char *mid);
-/*! Open Chat session */
-int ast_aji_create_chat(struct aji_client *client,char *room, char *server, char *topic);
-/*! Invite to opened Chat session */
-int ast_aji_invite_chat(struct aji_client *client, char *user, char *room, char *message);
-/*! Join/leave existing Chat session */
-int ast_aji_join_chat(struct aji_client *client, char *room, char *nick);
-int ast_aji_leave_chat(struct aji_client *client, char *room, char *nick);
-/*! Get a client via its name. Increases refcount of client by 1 */
-struct aji_client *ast_aji_get_client(const char *name);
-struct aji_client_container *ast_aji_get_clients(void);
-/*! Destructor function for buddies to be used with ASTOBJ_UNREF */
-void ast_aji_buddy_destroy(struct aji_buddy *obj);
-/*! Destructor function for clients to be used with ASTOBJ_UNREF after calls to ast_aji_get_client */
-void ast_aji_client_destroy(struct aji_client *obj);
-
-#endif
Modified: trunk/include/asterisk/jingle.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/jingle.h?view=diff&rev=418448&r1=418447&r2=418448
==============================================================================
--- trunk/include/asterisk/jingle.h (original)
+++ trunk/include/asterisk/jingle.h Sun Jul 13 00:05:49 2014
@@ -1,66 +1,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2005, Digium, Inc.
- *
- * Matt O'Gorman <mogorman at digium.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 Jingle definitions for chan_jingle
- *
- * \ref chan_jingle.c
- *
- * \author Matt O'Gorman <mogorman at digium.com>
- */
-
-
-#ifndef _ASTERISK_JINGLE_H
-#define _ASTERISK_JINGLE_H
-
-#include <iksemel.h>
-#include "asterisk/astobj.h"
-
-
-/* Jingle Constants */
-
-#define JINGLE_NODE "jingle"
-#define GOOGLE_NODE "session"
-
-#define JINGLE_NS "urn:xmpp:tmp:jingle"
-#define JINGLE_AUDIO_RTP_NS "urn:xmpp:tmp:jingle:apps:audio-rtp"
-#define JINGLE_VIDEO_RTP_NS "urn:xmpp:tmp:jingle:apps:video"
-#define JINGLE_ICE_UDP_NS "urn:xmpp:tmp:jingle:transports:ice-udp"
-#define JINGLE_DTMF_NS "urn:xmpp:tmp:jingle:dtmf"
-
-#define GOOGLE_NS "http://www.google.com/session"
-#define GOOGLE_JINGLE_NS "urn:xmpp:jingle:1"
-#define GOOGLE_AUDIO_NS "http://www.google.com/session/phone"
-#define GOOGLE_VIDEO_NS "http://www.google.com/session/video"
-#define GOOGLE_TRANSPORT_NS "http://www.google.com/transport/p2p"
-
-#define JINGLE_SID "sid"
-#define GOOGLE_SID "id"
-
-#define JINGLE_INITIATE "session-initiate"
-
-#define JINGLE_ACCEPT "session-accept"
-#define GOOGLE_ACCEPT "accept"
-
-#define JINGLE_NEGOTIATE "transport-info"
-#define GOOGLE_NEGOTIATE "candidates"
-
-#define JINGLE_INFO "session-info"
-#define JINGLE_TERMINATE "session-terminate"
-
-#endif
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