1038 lines
31 KiB
C
1038 lines
31 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2021, Naveen Albert
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*
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* Naveen Albert <asterisk@phreaknet.org>
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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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/*! \file
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*
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* \brief Tone detection module
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*
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* \author Naveen Albert <asterisk@phreaknet.org>
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*
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* \ingroup resources
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*/
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/*** MODULEINFO
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<support_level>extended</support_level>
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***/
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#include "asterisk.h"
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#include <math.h>
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#include "asterisk/module.h"
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#include "asterisk/frame.h"
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#include "asterisk/format_cache.h"
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#include "asterisk/channel.h"
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#include "asterisk/dsp.h"
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#include "asterisk/pbx.h"
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#include "asterisk/audiohook.h"
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#include "asterisk/app.h"
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#include "asterisk/indications.h"
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#include "asterisk/conversions.h"
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/*** DOCUMENTATION
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<application name="WaitForTone" language="en_US">
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<since>
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<version>16.21.0</version>
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<version>18.7.0</version>
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<version>19.0.0</version>
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</since>
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<synopsis>
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Wait for tone
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</synopsis>
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<syntax>
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<parameter name="freq" required="true">
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<para>Frequency of the tone to wait for.</para>
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</parameter>
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<parameter name="duration_ms" required="false">
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<para>Minimum duration of tone, in ms. Default is 500ms.
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Using a minimum duration under 50ms is unlikely to produce
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accurate results.</para>
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</parameter>
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<parameter name="timeout" required="false">
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<para>Maximum amount of time, in seconds, to wait for specified tone.
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Default is forever.</para>
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</parameter>
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<parameter name="times" required="false">
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<para>Number of times the tone should be detected (subject to the
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provided timeout) before returning. Default is 1.</para>
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</parameter>
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<parameter name="options" required="false">
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<optionlist>
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<option name="d">
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<para>Custom decibel threshold to use. Default is 16.</para>
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</option>
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<option name="s">
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<para>Squelch tone.</para>
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</option>
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</optionlist>
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</parameter>
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</syntax>
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<description>
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<para>Waits for a single-frequency tone to be detected before dialplan execution continues.</para>
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<variablelist>
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<variable name="WAITFORTONESTATUS">
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<para>This indicates the result of the wait.</para>
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<value name="SUCCESS"/>
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<value name="ERROR"/>
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<value name="TIMEOUT"/>
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<value name="HANGUP"/>
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</variable>
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</variablelist>
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</description>
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<see-also>
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<ref type="application">PlayTones</ref>
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</see-also>
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</application>
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<application name="ToneScan" language="en_US">
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<since>
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<version>16.23.0</version>
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<version>18.9.0</version>
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<version>19.1.0</version>
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</since>
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<synopsis>
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Wait for period of time while scanning for call progress tones
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</synopsis>
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<syntax>
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<parameter name="zone" required="false">
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<para>Call progress zone. Default is the system default.</para>
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</parameter>
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<parameter name="timeout" required="false">
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<para>Maximum amount of time, in seconds, to wait for call progress
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or signal tones. Default is forever.</para>
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</parameter>
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<parameter name="threshold" required="false">
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<para>DSP threshold required for a match. A higher number will
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require a longer match and may reduce false positives, at the
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expense of false negatives. Default is 1.</para>
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</parameter>
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<parameter name="options" required="false">
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<optionlist>
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<option name="f">
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<para>Enable fax machine detection. By default, this is disabled.</para>
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</option>
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<option name="v">
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<para>Enable voice detection. By default, this is disabled.</para>
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</option>
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</optionlist>
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</parameter>
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</syntax>
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<description>
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<para>Waits for a a distinguishable call progress tone and then exits.
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Unlike a conventional scanner, this is not currently capable of
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scanning for modem carriers.</para>
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<variablelist>
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<variable name="TONESCANSTATUS">
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This indicates the result of the scan.
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<value name="RINGING">
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Audible ringback tone
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</value>
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<value name="BUSY">
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Busy tone
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</value>
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<value name="SIT">
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Special Information Tones
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</value>
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<value name="VOICE">
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Human voice detected
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</value>
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<value name="DTMF">
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DTMF digit
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</value>
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<value name="FAX">
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Fax (answering)
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</value>
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<value name="MODEM">
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Modem (answering)
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</value>
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<value name="DIALTONE">
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Dial tone
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</value>
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<value name="NUT">
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UK Number Unobtainable tone
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</value>
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<value name="TIMEOUT">
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Timeout reached before any positive detection
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</value>
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<value name="HANGUP">
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Caller hung up before any positive detection
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</value>
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</variable>
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</variablelist>
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</description>
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<see-also>
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<ref type="application">WaitForTone</ref>
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</see-also>
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</application>
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<function name="TONE_DETECT" language="en_US">
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<since>
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<version>16.21.0</version>
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<version>18.7.0</version>
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<version>19.0.0</version>
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</since>
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<synopsis>
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Asynchronously detects a tone
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</synopsis>
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<syntax>
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<parameter name="freq" required="true">
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<para>Frequency of the tone to detect. To disable frequency
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detection completely (e.g. for signal detection only),
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specify 0 for the frequency.</para>
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</parameter>
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<parameter name="duration_ms" required="false">
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<para>Minimum duration of tone, in ms. Default is 500ms.
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Using a minimum duration under 50ms is unlikely to produce
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accurate results.</para>
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</parameter>
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<parameter name="options">
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<optionlist>
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<option name="a">
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<para>Match immediately on Special Information Tones, instead of or in addition
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to a particular frequency.</para>
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</option>
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<option name="b">
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<para>Match immediately on a busy signal, instead of or in addition to
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a particular frequency.</para>
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</option>
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<option name="c">
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<para>Match immediately on a dial tone, instead of or in addition to
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a particular frequency.</para>
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</option>
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<option name="d">
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<para>Custom decibel threshold to use. Default is 16.</para>
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</option>
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<option name="g">
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<para>Go to the specified context,exten,priority if tone is received on this channel.
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Detection will not end automatically.</para>
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</option>
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<option name="h">
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<para>Go to the specified context,exten,priority if tone is transmitted on this channel.
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Detection will not end automatically.</para>
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</option>
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<option name="n">
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<para>Number of times the tone should be detected (subject to the
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provided timeout) before going to the destination provided in the <literal>g</literal>
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or <literal>h</literal> option. Default is 1.</para>
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</option>
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<option name="r">
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<para>Apply to received frames only. Default is both directions.</para>
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</option>
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<option name="s">
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<para>Squelch tone.</para>
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</option>
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<option name="t">
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<para>Apply to transmitted frames only. Default is both directions.</para>
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</option>
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<option name="x">
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<para>Destroy the detector (stop detection).</para>
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</option>
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</optionlist>
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</parameter>
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</syntax>
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<description>
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<para>The TONE_DETECT function detects a single-frequency tone and keeps
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track of how many times the tone has been detected.</para>
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<para>When reading this function (instead of writing), supply <literal>tx</literal>
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to get the number of times a tone has been detected in the TX direction and
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<literal>rx</literal> to get the number of times a tone has been detected in the
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RX direction.</para>
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<example title="intercept2600">
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same => n,Set(TONE_DETECT(2600,1000,g(got-2600,s,1))=) ; detect 2600 Hz
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same => n,Wait(15)
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same => n,NoOp(${TONE_DETECT(rx)})
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</example>
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<example title="dropondialtone">
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same => n,Set(TONE_DETECT(0,,bg(my-hangup,s,1))=) ; disconnect a call if we hear a busy signal
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same => n,Goto(somewhere-else)
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same => n(myhangup),Hangup()
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</example>
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<example title="removedetector">
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same => n,Set(TONE_DETECT(0,,x)=) ; remove the detector from the channel
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</example>
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</description>
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</function>
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***/
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struct detect_information {
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struct ast_dsp *dsp;
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struct ast_audiohook audiohook;
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int freq1;
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int freq2;
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int duration;
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int db;
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char *gototx;
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char *gotorx;
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unsigned short int squelch;
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unsigned short int tx;
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unsigned short int rx;
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int txcount;
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int rxcount;
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int hitsrequired;
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int signalfeatures;
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};
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enum td_opts {
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OPT_TX = (1 << 1),
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OPT_RX = (1 << 2),
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OPT_END_FILTER = (1 << 3),
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OPT_GOTO_RX = (1 << 4),
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OPT_GOTO_TX = (1 << 5),
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OPT_DECIBEL = (1 << 6),
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OPT_SQUELCH = (1 << 7),
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OPT_HITS_REQ = (1 << 8),
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OPT_SIT = (1 << 9),
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OPT_BUSY = (1 << 10),
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OPT_DIALTONE = (1 << 11),
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};
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enum {
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OPT_ARG_DECIBEL,
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OPT_ARG_GOTO_RX,
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OPT_ARG_GOTO_TX,
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OPT_ARG_HITS_REQ,
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/* note: this entry _MUST_ be the last one in the enum */
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OPT_ARG_ARRAY_SIZE,
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};
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AST_APP_OPTIONS(td_opts, {
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AST_APP_OPTION('a', OPT_SIT),
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AST_APP_OPTION('b', OPT_BUSY),
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AST_APP_OPTION('c', OPT_DIALTONE),
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AST_APP_OPTION_ARG('d', OPT_DECIBEL, OPT_ARG_DECIBEL),
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AST_APP_OPTION_ARG('g', OPT_GOTO_RX, OPT_ARG_GOTO_RX),
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AST_APP_OPTION_ARG('h', OPT_GOTO_TX, OPT_ARG_GOTO_TX),
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AST_APP_OPTION_ARG('n', OPT_HITS_REQ, OPT_ARG_HITS_REQ),
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AST_APP_OPTION('s', OPT_SQUELCH),
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AST_APP_OPTION('t', OPT_TX),
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AST_APP_OPTION('r', OPT_RX),
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AST_APP_OPTION('x', OPT_END_FILTER),
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});
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static void destroy_callback(void *data)
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{
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struct detect_information *di = data;
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ast_dsp_free(di->dsp);
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if (di->gotorx) {
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ast_free(di->gotorx);
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}
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if (di->gototx) {
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ast_free(di->gototx);
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}
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ast_audiohook_lock(&di->audiohook);
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ast_audiohook_detach(&di->audiohook);
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ast_audiohook_unlock(&di->audiohook);
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ast_audiohook_destroy(&di->audiohook);
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ast_free(di);
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return;
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}
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static const struct ast_datastore_info detect_datastore = {
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.type = "detect",
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.destroy = destroy_callback
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};
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static int detect_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
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{
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struct ast_datastore *datastore = NULL;
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struct detect_information *di = NULL;
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int match = 0;
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/* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
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if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
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return 0;
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}
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/* Grab datastore which contains our gain information */
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if (!(datastore = ast_channel_datastore_find(chan, &detect_datastore, NULL))) {
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return 0;
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}
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di = datastore->data;
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if (!frame || frame->frametype != AST_FRAME_VOICE) {
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return 0;
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}
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if (!(direction == AST_AUDIOHOOK_DIRECTION_READ ? di->rx : di->tx)) {
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return 0;
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}
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/* ast_dsp_process may free the frame and return a new one */
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frame = ast_frdup(frame);
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frame = ast_dsp_process(chan, di->dsp, frame);
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if (frame->frametype == AST_FRAME_DTMF) {
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char result = frame->subclass.integer;
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if (result == 'q') {
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int now;
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match = 1;
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if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
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di->rxcount = di->rxcount + 1;
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now = di->rxcount;
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} else {
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di->txcount = di->txcount + 1;
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now = di->txcount;
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}
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ast_debug(1, "TONE_DETECT just got a hit (#%d in this direction, waiting for %d total)\n", now, di->hitsrequired);
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if (now >= di->hitsrequired) {
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if (direction == AST_AUDIOHOOK_DIRECTION_READ && di->gotorx) {
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ast_async_parseable_goto(chan, di->gotorx);
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} else if (di->gototx) {
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ast_async_parseable_goto(chan, di->gototx);
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}
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}
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}
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}
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if (di->signalfeatures && !match) { /* skip unless there are call progress/signal options */
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int tstate, tcount;
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tcount = ast_dsp_get_tcount(di->dsp);
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tstate = ast_dsp_get_tstate(di->dsp);
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if (tstate > 0) {
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ast_debug(3, "tcount: %d, tstate: %d\n", tcount, tstate);
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switch (tstate) {
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case DSP_TONE_STATE_DIALTONE:
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if (di->signalfeatures & DSP_FEATURE_WAITDIALTONE) {
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match = 1;
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}
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break;
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case DSP_TONE_STATE_BUSY:
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if (di->signalfeatures & DSP_PROGRESS_BUSY) {
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match = 1;
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}
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break;
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case DSP_TONE_STATE_SPECIAL3:
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if (di->signalfeatures & DSP_PROGRESS_CONGESTION) {
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match = 1;
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}
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break;
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default: /* ignore */
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break;
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}
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if (match) {
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if (direction == AST_AUDIOHOOK_DIRECTION_READ && di->gotorx) {
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ast_async_parseable_goto(chan, di->gotorx);
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} else if (di->gototx) {
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ast_async_parseable_goto(chan, di->gototx);
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} else {
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ast_debug(3, "Detected call progress signal, but don't know where to go\n");
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}
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}
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}
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}
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/* this could be the duplicated frame or a new one, doesn't matter */
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ast_frfree(frame);
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return 0;
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}
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static int remove_detect(struct ast_channel *chan)
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{
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struct ast_datastore *datastore = NULL;
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struct detect_information *data;
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SCOPED_CHANNELLOCK(chan_lock, chan);
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datastore = ast_channel_datastore_find(chan, &detect_datastore, NULL);
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if (!datastore) {
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ast_log(AST_LOG_WARNING, "Cannot remove TONE_DETECT from %s: TONE_DETECT not currently enabled\n",
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ast_channel_name(chan));
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return -1;
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}
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data = datastore->data;
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if (ast_audiohook_remove(chan, &data->audiohook)) {
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ast_log(AST_LOG_WARNING, "Failed to remove TONE_DETECT audiohook from channel %s\n", ast_channel_name(chan));
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return -1;
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}
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if (ast_channel_datastore_remove(chan, datastore)) {
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ast_log(AST_LOG_WARNING, "Failed to remove TONE_DETECT datastore from channel %s\n",
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ast_channel_name(chan));
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return -1;
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}
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ast_datastore_free(datastore);
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return 0;
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}
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static int freq_parser(char *freqs, int *freq1, int *freq2) {
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char *f1, *f2, *f3;
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if (ast_strlen_zero(freqs)) {
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ast_log(LOG_ERROR, "No frequency specified\n");
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return -1;
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}
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f3 = ast_strdupa(freqs);
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f1 = strsep(&f3, "+");
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f2 = strsep(&f3, "+");
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if (!ast_strlen_zero(f3)) {
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ast_log(LOG_WARNING, "Only up to 2 frequencies may be specified: %s\n", freqs);
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return -1;
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}
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if (ast_str_to_int(f1, freq1)) {
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ast_log(LOG_WARNING, "Frequency must be an integer: %s\n", f1);
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return -1;
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}
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if (*freq1 < 0) {
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ast_log(LOG_WARNING, "Sorry, no negative frequencies: %d\n", *freq1);
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return -1;
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}
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if (!ast_strlen_zero(f2)) {
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ast_log(LOG_WARNING, "Sorry, currently only 1 frequency is supported\n");
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return -1;
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/* not supported just yet, but possibly will be in the future */
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if (ast_str_to_int(f2, freq2)) {
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ast_log(LOG_WARNING, "Frequency must be an integer: %s\n", f2);
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return -1;
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}
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if (*freq2 < 1) {
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ast_log(LOG_WARNING, "Sorry, positive frequencies only: %d\n", *freq2);
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return -1;
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}
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|
}
|
|
return 0;
|
|
}
|
|
|
|
static char* goto_parser(struct ast_channel *chan, char *loc) {
|
|
char *exten, *pri, *context, *parse;
|
|
char *dest;
|
|
int size;
|
|
parse = ast_strdupa(loc);
|
|
context = strsep(&parse, ",");
|
|
exten = strsep(&parse, ",");
|
|
pri = strsep(&parse, ",");
|
|
if (!exten) {
|
|
pri = context;
|
|
exten = NULL;
|
|
context = NULL;
|
|
} else if (!pri) {
|
|
pri = exten;
|
|
exten = context;
|
|
context = NULL;
|
|
}
|
|
ast_channel_lock(chan);
|
|
if (ast_strlen_zero(exten)) {
|
|
exten = ast_strdupa(ast_channel_exten(chan));
|
|
}
|
|
if (ast_strlen_zero(context)) {
|
|
context = ast_strdupa(ast_channel_context(chan));
|
|
}
|
|
ast_channel_unlock(chan);
|
|
|
|
/* size + 3: for 1 null terminator + 2 commas */
|
|
size = strlen(context) + strlen(exten) + strlen(pri) + 3;
|
|
dest = ast_malloc(size + 1);
|
|
if (!dest) {
|
|
ast_log(LOG_ERROR, "Failed to parse goto: %s,%s,%s\n", context, exten, pri);
|
|
return NULL;
|
|
}
|
|
snprintf(dest, size, "%s,%s,%s", context, exten, pri);
|
|
return dest;
|
|
}
|
|
|
|
static int detect_read(struct ast_channel *chan, const char *cmd, char *data, char *buffer, size_t buflen)
|
|
{
|
|
struct ast_datastore *datastore = NULL;
|
|
struct detect_information *di = NULL;
|
|
|
|
if (!chan) {
|
|
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
|
|
return -1;
|
|
}
|
|
|
|
ast_channel_lock(chan);
|
|
if (!(datastore = ast_channel_datastore_find(chan, &detect_datastore, NULL))) {
|
|
ast_channel_unlock(chan);
|
|
return -1; /* function not initiated yet, so nothing to read */
|
|
} else {
|
|
ast_channel_unlock(chan);
|
|
di = datastore->data;
|
|
}
|
|
|
|
if (strchr(data, 't')) {
|
|
snprintf(buffer, buflen, "%d", di->txcount);
|
|
} else if (strchr(data, 'r')) {
|
|
snprintf(buffer, buflen, "%d", di->rxcount);
|
|
} else {
|
|
ast_log(LOG_WARNING, "Invalid direction: %s\n", data);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int parse_signal_features(struct ast_flags *flags)
|
|
{
|
|
int features = 0;
|
|
|
|
if (ast_test_flag(flags, OPT_SIT)) {
|
|
features |= DSP_PROGRESS_CONGESTION;
|
|
}
|
|
if (ast_test_flag(flags, OPT_BUSY)) {
|
|
features |= DSP_PROGRESS_BUSY;
|
|
}
|
|
if (ast_test_flag(flags, OPT_DIALTONE)) {
|
|
features |= DSP_FEATURE_WAITDIALTONE;
|
|
}
|
|
|
|
return features;
|
|
}
|
|
|
|
static int detect_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
|
|
{
|
|
char *parse;
|
|
struct ast_datastore *datastore = NULL;
|
|
struct detect_information *di = NULL;
|
|
struct ast_flags flags = { 0 };
|
|
char *opt_args[OPT_ARG_ARRAY_SIZE];
|
|
struct ast_dsp *dsp;
|
|
int freq1 = 0, freq2 = 0, duration = 500, db = 16, squelch = 0, hitsrequired = 1;
|
|
int signalfeatures = 0;
|
|
|
|
AST_DECLARE_APP_ARGS(args,
|
|
AST_APP_ARG(freqs);
|
|
AST_APP_ARG(duration);
|
|
AST_APP_ARG(options);
|
|
);
|
|
|
|
if (!chan) {
|
|
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
|
|
return -1;
|
|
}
|
|
parse = ast_strdupa(data);
|
|
AST_STANDARD_APP_ARGS(args, parse);
|
|
|
|
if (!ast_strlen_zero(args.options)) {
|
|
ast_app_parse_options(td_opts, &flags, opt_args, args.options);
|
|
}
|
|
if (ast_test_flag(&flags, OPT_END_FILTER)) {
|
|
return remove_detect(chan);
|
|
}
|
|
if (freq_parser(args.freqs, &freq1, &freq2)) {
|
|
return -1;
|
|
}
|
|
if (!ast_strlen_zero(args.duration) && (ast_str_to_int(args.duration, &duration) || duration < 1)) {
|
|
ast_log(LOG_WARNING, "Invalid duration: %s\n", args.duration);
|
|
return -1;
|
|
}
|
|
if (ast_test_flag(&flags, OPT_HITS_REQ) && !ast_strlen_zero(opt_args[OPT_ARG_HITS_REQ])) {
|
|
if ((ast_str_to_int(opt_args[OPT_ARG_HITS_REQ], &hitsrequired) || hitsrequired < 1)) {
|
|
ast_log(LOG_WARNING, "Invalid number hits required: %s\n", opt_args[OPT_ARG_HITS_REQ]);
|
|
return -1;
|
|
}
|
|
}
|
|
if (ast_test_flag(&flags, OPT_DECIBEL) && !ast_strlen_zero(opt_args[OPT_ARG_DECIBEL])) {
|
|
if ((ast_str_to_int(opt_args[OPT_ARG_DECIBEL], &db) || db < 1)) {
|
|
ast_log(LOG_WARNING, "Invalid decibel level: %s\n", opt_args[OPT_ARG_DECIBEL]);
|
|
return -1;
|
|
}
|
|
}
|
|
signalfeatures = parse_signal_features(&flags);
|
|
|
|
ast_channel_lock(chan);
|
|
if (!(datastore = ast_channel_datastore_find(chan, &detect_datastore, NULL))) {
|
|
if (!(datastore = ast_datastore_alloc(&detect_datastore, NULL))) {
|
|
ast_channel_unlock(chan);
|
|
return 0;
|
|
}
|
|
if (!(di = ast_calloc(1, sizeof(*di)))) {
|
|
ast_datastore_free(datastore);
|
|
ast_channel_unlock(chan);
|
|
return 0;
|
|
}
|
|
ast_audiohook_init(&di->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Tone Detector", AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
|
|
di->audiohook.manipulate_callback = detect_callback;
|
|
if (!(dsp = ast_dsp_new())) {
|
|
ast_datastore_free(datastore);
|
|
ast_channel_unlock(chan);
|
|
ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
|
|
return -1;
|
|
}
|
|
di->signalfeatures = signalfeatures; /* we're not including freq detect */
|
|
if (freq1 > 0) {
|
|
signalfeatures |= DSP_FEATURE_FREQ_DETECT;
|
|
ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
|
|
}
|
|
ast_dsp_set_features(dsp, signalfeatures);
|
|
di->dsp = dsp;
|
|
di->txcount = 0;
|
|
di->rxcount = 0;
|
|
ast_debug(1, "Keeping our ears open for %s Hz, %d db\n", args.freqs, db);
|
|
datastore->data = di;
|
|
ast_channel_datastore_add(chan, datastore);
|
|
ast_audiohook_attach(chan, &di->audiohook);
|
|
} else {
|
|
di = datastore->data;
|
|
dsp = di->dsp;
|
|
di->signalfeatures = signalfeatures; /* we're not including freq detect */
|
|
if (freq1 > 0) {
|
|
signalfeatures |= DSP_FEATURE_FREQ_DETECT;
|
|
ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
|
|
}
|
|
ast_dsp_set_features(dsp, signalfeatures);
|
|
}
|
|
di->duration = duration;
|
|
di->gotorx = NULL;
|
|
di->gototx = NULL;
|
|
/* resolve gotos now, in case a full context,exten,pri wasn't specified */
|
|
if (ast_test_flag(&flags, OPT_GOTO_RX) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO_RX])) {
|
|
di->gotorx = goto_parser(chan, opt_args[OPT_ARG_GOTO_RX]);
|
|
}
|
|
if (ast_test_flag(&flags, OPT_GOTO_TX) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO_TX])) {
|
|
di->gototx = goto_parser(chan, opt_args[OPT_ARG_GOTO_TX]);
|
|
}
|
|
di->db = db;
|
|
di->hitsrequired = hitsrequired;
|
|
di->squelch = ast_test_flag(&flags, OPT_SQUELCH);
|
|
di->tx = 1;
|
|
di->rx = 1;
|
|
if (ast_strlen_zero(args.options) || ast_test_flag(&flags, OPT_TX)) {
|
|
di->tx = 1;
|
|
di->rx = 0;
|
|
}
|
|
if (ast_strlen_zero(args.options) || ast_test_flag(&flags, OPT_RX)) {
|
|
di->rx = 1;
|
|
di->tx = 0;
|
|
}
|
|
ast_channel_unlock(chan);
|
|
|
|
return 0;
|
|
}
|
|
|
|
enum {
|
|
OPT_APP_DECIBEL = (1 << 0),
|
|
OPT_APP_SQUELCH = (1 << 1),
|
|
};
|
|
|
|
enum {
|
|
OPT_APP_ARG_DECIBEL,
|
|
/* note: this entry _MUST_ be the last one in the enum */
|
|
OPT_APP_ARG_ARRAY_SIZE,
|
|
};
|
|
|
|
AST_APP_OPTIONS(wait_exec_options, BEGIN_OPTIONS
|
|
AST_APP_OPTION_ARG('d', OPT_APP_DECIBEL, OPT_APP_ARG_DECIBEL),
|
|
AST_APP_OPTION('s', OPT_APP_SQUELCH),
|
|
END_OPTIONS);
|
|
|
|
static int wait_exec(struct ast_channel *chan, const char *data)
|
|
{
|
|
char *appdata;
|
|
struct ast_flags flags = {0};
|
|
char *opt_args[OPT_APP_ARG_ARRAY_SIZE];
|
|
double timeoutf = 0;
|
|
int freq1 = 0, freq2 = 0, timeout = 0, duration = 500, times = 1, db = 16, squelch = 0;
|
|
struct ast_frame *frame = NULL;
|
|
struct ast_dsp *dsp;
|
|
struct timeval start;
|
|
int remaining_time = 0;
|
|
int hits = 0;
|
|
AST_DECLARE_APP_ARGS(args,
|
|
AST_APP_ARG(freqs);
|
|
AST_APP_ARG(duration);
|
|
AST_APP_ARG(timeout);
|
|
AST_APP_ARG(times);
|
|
AST_APP_ARG(options);
|
|
);
|
|
|
|
appdata = ast_strdupa(data);
|
|
AST_STANDARD_APP_ARGS(args, appdata);
|
|
|
|
if (!ast_strlen_zero(args.options)) {
|
|
ast_app_parse_options(wait_exec_options, &flags, opt_args, args.options);
|
|
}
|
|
if (freq_parser(args.freqs, &freq1, &freq2)) {
|
|
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
|
|
return -1;
|
|
}
|
|
if (!ast_strlen_zero(args.timeout) && (sscanf(args.timeout, "%30lf", &timeoutf) != 1 || timeoutf < 0)) {
|
|
ast_log(LOG_WARNING, "Invalid timeout: %s\n", args.timeout);
|
|
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
|
|
return -1;
|
|
}
|
|
timeout = 1000 * timeoutf;
|
|
if (!ast_strlen_zero(args.duration) && (ast_str_to_int(args.duration, &duration) || duration < 1)) {
|
|
ast_log(LOG_WARNING, "Invalid duration: %s\n", args.duration);
|
|
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
|
|
return -1;
|
|
}
|
|
if (!ast_strlen_zero(args.times) && (ast_str_to_int(args.times, ×) || times < 1)) {
|
|
ast_log(LOG_WARNING, "Invalid number of times: %s\n", args.times);
|
|
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
|
|
return -1;
|
|
}
|
|
if (ast_test_flag(&flags, OPT_APP_DECIBEL) && !ast_strlen_zero(opt_args[OPT_APP_ARG_DECIBEL])) {
|
|
if ((ast_str_to_int(opt_args[OPT_APP_ARG_DECIBEL], &db) || db < 1)) {
|
|
ast_log(LOG_WARNING, "Invalid decibel level: %s\n", opt_args[OPT_APP_ARG_DECIBEL]);
|
|
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
|
|
return -1;
|
|
}
|
|
}
|
|
squelch = ast_test_flag(&flags, OPT_APP_SQUELCH);
|
|
if (!(dsp = ast_dsp_new())) {
|
|
ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
|
|
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "ERROR");
|
|
return -1;
|
|
}
|
|
ast_dsp_set_features(dsp, DSP_FEATURE_FREQ_DETECT);
|
|
ast_dsp_set_freqmode(dsp, freq1, duration, db, squelch);
|
|
ast_debug(1, "Waiting for %s Hz, %d time(s), timeout %d ms, %d db\n", args.freqs, times, timeout, db);
|
|
start = ast_tvnow();
|
|
do {
|
|
if (timeout > 0) {
|
|
remaining_time = ast_remaining_ms(start, timeout);
|
|
if (remaining_time <= 0) {
|
|
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "TIMEOUT");
|
|
break;
|
|
}
|
|
}
|
|
if (ast_waitfor(chan, 1000) > 0) {
|
|
if (!(frame = ast_read(chan))) {
|
|
ast_debug(1, "Channel '%s' did not return a frame; probably hung up.\n", ast_channel_name(chan));
|
|
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "HANGUP");
|
|
break;
|
|
} else if (frame->frametype == AST_FRAME_VOICE) {
|
|
frame = ast_dsp_process(chan, dsp, frame);
|
|
if (frame->frametype == AST_FRAME_DTMF) {
|
|
char result = frame->subclass.integer;
|
|
if (result == 'q') {
|
|
hits++;
|
|
ast_debug(1, "We just detected %s Hz (hit #%d)\n", args.freqs, hits);
|
|
if (hits >= times) {
|
|
ast_frfree(frame);
|
|
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "SUCCESS");
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
ast_frfree(frame);
|
|
} else {
|
|
pbx_builtin_setvar_helper(chan, "WAITFORTONESTATUS", "HANGUP");
|
|
}
|
|
} while (timeout == 0 || remaining_time > 0);
|
|
ast_dsp_free(dsp);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static char *waitapp = "WaitForTone";
|
|
static char *scanapp = "ToneScan";
|
|
|
|
static int scan_exec(struct ast_channel *chan, const char *data)
|
|
{
|
|
char *appdata;
|
|
double timeoutf = 0;
|
|
int timeout = 0;
|
|
struct ast_frame *frame = NULL, *frame2 = NULL;
|
|
struct ast_dsp *dsp = NULL, *dsp2 = NULL;
|
|
struct timeval start;
|
|
int remaining_time = 0;
|
|
int features, match = 0, fax = 0, voice = 0, threshold = 1;
|
|
AST_DECLARE_APP_ARGS(args,
|
|
AST_APP_ARG(zone);
|
|
AST_APP_ARG(timeout);
|
|
AST_APP_ARG(threshold);
|
|
AST_APP_ARG(options);
|
|
);
|
|
|
|
appdata = ast_strdupa(data);
|
|
AST_STANDARD_APP_ARGS(args, appdata);
|
|
|
|
if (!ast_strlen_zero(args.timeout) && (sscanf(args.timeout, "%30lf", &timeoutf) != 1 || timeout < 0)) {
|
|
ast_log(LOG_WARNING, "Invalid timeout: %s\n", args.timeout);
|
|
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
|
|
return -1;
|
|
}
|
|
if (!ast_strlen_zero(args.threshold) && (ast_str_to_int(args.threshold, &threshold) || threshold < 1)) {
|
|
ast_log(LOG_WARNING, "Invalid threshold: %s\n", args.threshold);
|
|
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
|
|
return -1;
|
|
}
|
|
timeout = 1000 * timeoutf;
|
|
|
|
if (!ast_strlen_zero(args.options) && strchr(args.options, 'f')) {
|
|
fax = 1;
|
|
}
|
|
if (!ast_strlen_zero(args.options) && strchr(args.options, 'v')) {
|
|
voice = 1;
|
|
}
|
|
|
|
if (!(dsp = ast_dsp_new())) {
|
|
ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
|
|
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
|
|
return -1;
|
|
}
|
|
|
|
if (!ast_strlen_zero(args.zone)) {
|
|
if (ast_dsp_set_call_progress_zone(dsp, args.zone)) {
|
|
ast_log(LOG_WARNING, "Invalid call progress zone: %s\n", args.zone);
|
|
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
|
|
ast_dsp_free(dsp);
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
if (fax) {
|
|
if (!(dsp2 = ast_dsp_new())) {
|
|
ast_dsp_free(dsp);
|
|
ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
|
|
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "ERROR");
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
features = DSP_PROGRESS_RINGING; /* audible ringback tone */
|
|
features |= DSP_PROGRESS_BUSY; /* busy signal */
|
|
features |= DSP_PROGRESS_CONGESTION; /* SIT tones (not reorder!) */
|
|
features |= DSP_PROGRESS_TALK; /* voice. */
|
|
features |= DSP_FEATURE_WAITDIALTONE; /* dial tone */
|
|
features |= DSP_FEATURE_FREQ_DETECT; /* modem answer */
|
|
if (voice) {
|
|
features |= DSP_TONE_STATE_TALKING; /* voice */
|
|
}
|
|
ast_dsp_set_features(dsp, features);
|
|
/* all modems begin negotiating with Bell 103. An answering modem just sends mark tone, or 2225 Hz */
|
|
ast_dsp_set_freqmode(dsp, 2225, 400, 16, 0); /* this needs to be pretty short, or the progress tones code will think this is voice */
|
|
|
|
if (fax) { /* fax detect uses same tone detect internals as modem and causes things to not work as intended, so use a separate DSP if needed. */
|
|
ast_dsp_set_features(dsp2, DSP_FEATURE_FAX_DETECT); /* fax tone */
|
|
ast_dsp_set_faxmode(dsp2, DSP_FAXMODE_DETECT_CED); /* we only care about the answering side (CED), not originating (CNG) */
|
|
}
|
|
|
|
ast_debug(1, "Starting tone scan, timeout: %d ms, threshold: %d\n", timeout, threshold);
|
|
start = ast_tvnow();
|
|
do {
|
|
if (timeout > 0) {
|
|
remaining_time = ast_remaining_ms(start, timeout);
|
|
if (remaining_time <= 0) {
|
|
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "TIMEOUT");
|
|
break;
|
|
}
|
|
}
|
|
if (ast_waitfor(chan, 1000) > 0) {
|
|
if (!(frame = ast_read(chan))) {
|
|
ast_debug(1, "Channel '%s' did not return a frame; probably hung up.\n", ast_channel_name(chan));
|
|
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "HANGUP");
|
|
break;
|
|
} else if (frame->frametype == AST_FRAME_VOICE) {
|
|
if (fax) {
|
|
frame2 = ast_frdup(frame);
|
|
}
|
|
frame = ast_dsp_process(chan, dsp, frame);
|
|
if (frame->frametype == AST_FRAME_DTMF) {
|
|
char result = frame->subclass.integer;
|
|
match = 1;
|
|
if (result == 'q') {
|
|
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "MODEM");
|
|
} else {
|
|
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "DTMF");
|
|
}
|
|
} else if (fax) {
|
|
char result;
|
|
frame2 = ast_dsp_process(chan, dsp2, frame2);
|
|
result = frame2->subclass.integer;
|
|
if (frame2->frametype == AST_FRAME_DTMF) {
|
|
if (result == 'e') {
|
|
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "FAX");
|
|
match = 1;
|
|
} else {
|
|
ast_debug(1, "Ignoring inactionable event\n"); /* shouldn't happen */
|
|
}
|
|
}
|
|
ast_frfree(frame2);
|
|
}
|
|
if (!match) {
|
|
int tstate, tcount;
|
|
tcount = ast_dsp_get_tcount(dsp);
|
|
tstate = ast_dsp_get_tstate(dsp);
|
|
if (tstate > 0) {
|
|
ast_debug(3, "tcount: %d, tstate: %d\n", tcount, tstate);
|
|
if (tcount >= threshold) {
|
|
match = 1;
|
|
switch (tstate) {
|
|
case DSP_TONE_STATE_RINGING:
|
|
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "RINGING");
|
|
break;
|
|
case DSP_TONE_STATE_DIALTONE:
|
|
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "DIALTONE");
|
|
break;
|
|
case DSP_TONE_STATE_TALKING:
|
|
/* even if we don't specify this feature, it's still checked, so we always need to handle it.
|
|
Even if we are looking for it, we need to wait a while or tones will be interpreted
|
|
as voice, because this will match first (and this should match last). */
|
|
if (voice && tcount > 15 && tcount >= threshold) {
|
|
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "VOICE");
|
|
} else {
|
|
match = 0;
|
|
}
|
|
break;
|
|
case DSP_TONE_STATE_BUSY:
|
|
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "BUSY");
|
|
break;
|
|
case DSP_TONE_STATE_SPECIAL3:
|
|
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "SIT");
|
|
break;
|
|
case DSP_TONE_STATE_HUNGUP: /* UK only */
|
|
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "NUT");
|
|
break;
|
|
default:
|
|
match = 0;
|
|
ast_debug(1, "Something else we weren't expecting? tstate: %d, #%d\n", tstate, tcount);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
ast_frfree(frame);
|
|
} else {
|
|
pbx_builtin_setvar_helper(chan, "TONESCANSTATUS", "HANGUP");
|
|
}
|
|
} while (!match && (timeout == 0 || remaining_time > 0));
|
|
ast_dsp_free(dsp);
|
|
if (dsp2) {
|
|
ast_dsp_free(dsp2);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static struct ast_custom_function detect_function = {
|
|
.name = "TONE_DETECT",
|
|
.read = detect_read,
|
|
.write = detect_write,
|
|
};
|
|
|
|
static int unload_module(void)
|
|
{
|
|
int res;
|
|
|
|
res = ast_unregister_application(waitapp);
|
|
res |= ast_unregister_application(scanapp);
|
|
res |= ast_custom_function_unregister(&detect_function);
|
|
|
|
return res;
|
|
}
|
|
|
|
static int load_module(void)
|
|
{
|
|
int res;
|
|
|
|
res = ast_register_application_xml(waitapp, wait_exec);
|
|
res |= ast_register_application_xml(scanapp, scan_exec);
|
|
res |= ast_custom_function_register(&detect_function);
|
|
|
|
return res;
|
|
}
|
|
|
|
AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Tone detection module");
|