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Also includes a test for retrieving rtpqos parameters, including a NULL RTP driver. Additionally, some further separation of the SIP internal API into headers was necessary. (closes issue #16652) Reported by: kkm Patches: 20100204__issue16652.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/501/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
42 lines
1.1 KiB
C
42 lines
1.1 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2010, Digium, Inc.
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*!
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* \file
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* \brief SIP dialplan functions header file
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*/
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#include "sip.h"
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#ifndef _SIP_DIALPLAN_FUNCTIONS_H
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#define _SIP_DIALPLAN_FUNCTIONS_H
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/*!
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* \brief Channel read dialplan function for SIP
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*/
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int sip_acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen);
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/*!
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* \brief register dialplan function tests
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*/
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void sip_dialplan_function_register_tests(void);
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/*!
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* \brief unregister dialplan function tests
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*/
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void sip_dialplan_function_unregister_tests(void);
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#endif /* !defined(_SIP_DIALPLAN_FUNCTIONS_H) */
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