From 14a985560ed5830aa3e1b5880890a59a5d0f0c2f Mon Sep 17 00:00:00 2001 From: Mark Michelson Date: Mon, 4 Jun 2012 20:26:12 +0000 Subject: Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- include/asterisk/sip_api.h | 51 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 51 insertions(+) create mode 100644 include/asterisk/sip_api.h (limited to 'include/asterisk/sip_api.h') diff --git a/include/asterisk/sip_api.h b/include/asterisk/sip_api.h new file mode 100644 index 000000000..018785a8d --- /dev/null +++ b/include/asterisk/sip_api.h @@ -0,0 +1,51 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 2012, Digium, Inc. + * + * Mark Michelson + * + * 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. + */ + +#ifndef __ASTERISK_SIP_H +#define __ASTERISK_SIP_H + +#if defined(__cplusplus) || defined(c_plusplus) +extern "C" { +#endif + +#include "asterisk/optional_api.h" +#include "asterisk/config.h" + +/*! + * \brief Send a customized SIP INFO request + * + * \param headers The headers to add to the INFO request + * \param content_type The content type header to add + * \param conten The body of the INFO request + * \param useragent_filter If non-NULL, only send the INFO if the + * recipient's User-Agent contains useragent_filter as a substring + * + * \retval 0 Success + * \retval non-zero Failure + */ +int ast_sipinfo_send(struct ast_channel *chan, + struct ast_variable *headers, + const char *content_type, + const char *content, + const char *useragent_filter); + +#if defined(__cplusplus) || defined(c_plusplus) +} +#endif + +#endif /* __ASTERISK_SIP_H */ -- cgit v1.2.3