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# docker一键运行阿里云SDMMRCP和FreeSWITCH
## 版本
FreeSWITCH 1.10.10
SDM单机版支持100路asr并发+100路tts并发
## 系统要求
linux
docker
docker compose (安装docker engine时会自动安装好)
lua5.1
luarocks
jq
openssl
## 业务要求
在阿里云开通“智能语音交互”业务完成创建项目并生成AccessKeyID和AccessKeySecret需要得到AccessKeyID、AccessKeySecret和appkey这几个参数注意AccessKey要具有使用智能语音交互的权限
## 使用方法
初次使用时,当确定满足系统要求和业务要求的情况下,直接执行
```sh
./install.sh
```
如果没有执行权限就手动加一下`chmod +x install.sh`。
安装过程中会要求填入阿里云的AccessKeyID、AccessKeySecret、appkey参数这些参数以后可以在`sdm/data/nls-cloud-sdm/conf/`中的配置文件中修改。
安装过程会随机生成FreeSWITCH的ESL密码和默认SIP密码这些参数以后可以在`switch/conf/`中的配置文件中修改。
安装程序会自动启动容器。
### 进入fs_cli
```sh
./fs_cli.sh
```
如果没有执行权限就手动加一下`chmod +x fs_cli.sh`。
如果执行后无法进入请检查:
- 检查switch容器是否启动
- 检查.fs_cli_conf中的密码和端口是否与event_socket.conf.xml中一致
- 也有可能是FreeSWITCH服务还没有完全启动好。
### 重启服务
无论修改sdm配置还是switch配置都得在服务重启后才会生效fs_cli中也有reloadxml等重载配置的命令
```sh
# 重启sdm和switch
./restart.sh
# 重启sdm
./restart.sh sdm
# 重启switch
./restart.sh switch
```
> 在switch/scripts中写的lua脚本修改后不需要重启服务就能生效。
## 默认端口
这个包完全修改了FreeSWITCH的默认端口号具体如下
| 业务 | 端口号 |
|---|---|
| internal_sip_port | 62260 |
| internal_tls_port | 62261 |
| external_sip_port | 62280 |
| external_tls_port | 62281 |
| ws-binding | 62266 |
| wss-binding | 62243 |
| event_socket | 62221 |
| rtp-start-port | 30000 |
| rtp-end-port | 39999 |
| (mrcp)rtp-port-min | 20000 |
| (mrcp)rtp-port-max | 29999 |
除了FreeSWITCH外SDM也是以host模式启动其占用的端口都是阿里云出厂时的默认值
| 业务 | 端口号 |
|---|---|
| SIP | 7010 |
| MRCP | 1544、1554 |
| RTP | 10000 - 20000 |
## 测试演示
### 分机注册
容器启动后通过SIP端口使用1001-1019分机号和安装时生成的默认SIP密码应该可以正确注册成功。
### 回声演示
分机注册成功后用分机呼叫9196应该可以启动回声应答你说什么它同时给你放什么
### TTS演示
进入fs_cli执行下面的命令
```sh
bgapi originate user/1001 &lua(demo/tts.lua)
```
1001应该可以接到电话电话接起来后可以听到“你好我是机器人XXX很高兴认识你……”。
### ASR演示
进入fs_cli执行下面的命令
```sh
bgapi originate user/1001 &lua(demo/asr.lua)
```
1001应该可以接到电话电话接起来后可以跟机器人做“复读游戏”你说一句它用asr识别后再用tts给你复读一句。
> 如果asr.lua执行出错请按照`switch/scripts/demo/asr.lua`中的注释安装lua依赖包再试试。
## 开发建议
请不要把业务代码提交到这个仓库里来,建议你把仓库接到你的业务目录中后直接删除.git目录让你的业务仓库接管后面的代码管理工作。
安装成功后删除switch/sounds/sounds.tar.bz2文件可以节省一点磁盘空间。

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version: "3.1"
services:
switch:
image: registry.cn-shanghai.aliyuncs.com/wandoubaba/sdmswitch:1.10.10
restart: always
container_name: switch
volumes:
- ./switch/.fs_cli_conf:/root/.fs_cli_conf
- ./switch/conf:/usr/local/freeswitch/conf
- ./switch/grammar:/usr/local/freeswitch/grammar
- ./switch/recordings:/usr/local/freeswitch/recordings
- ./switch/scripts:/usr/local/freeswitch/scripts
- ./switch/sounds:/usr/local/freeswitch/sounds
- ./switch/log:/usr/local/freeswitch/log
- /usr/local/share/lua/5.1:/usr/local/share/lua/5.1
- /usr/local/lib/lua/5.1:/usr/local/lib/lua/5.1
network_mode: host
command: ["freeswitch"]
depends_on:
- sdm
sdm:
image: registry.cn-shanghai.aliyuncs.com/nls-cloud/sdm
restart: always
container_name: sdm
volumes:
- ./sdm/data:/home/admin/disk
- ./sdm/logs:/home/admin/logs
network_mode: host
privileged: true
command: ["standalone"]

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#!/bin/bash
docker exec -it switch fs_cli

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#!/bin/bash
set -e
# Check if Docker is installed
if ! command -v docker &> /dev/null; then
echo "Docker is not installed. Please refer to https://docs.docker.com/engine/install/ to install Docker Engine."
exit 1
fi
# Check if 'switch' container exists
if docker ps -a --format '{{.Names}}' | grep -q '^switch$'; then
container_id=$(docker ps -a --format '{{.ID}}' --filter name=switch)
echo "The 'switch' container already exists. Container ID: $container_id. Installation cannot proceed."
exit 1
fi
# Check if 'sdm' container exists
if docker ps -a --format '{{.Names}}' | grep -q '^sdm$'; then
container_id=$(docker ps -a --format '{{.ID}}' --filter name=sdm)
echo "The 'sdm' container already exists. Container ID: $container_id. Installation cannot proceed."
exit 1
fi
# Check if jq is installed
if ! dpkg -s jq >/dev/null 2>&1; then
echo "jq package is not installed. Please install it using 'sudo apt-get install jq' and try again."
exit 1
fi
# Check if openssl is installed
if ! dpkg -s openssl >/dev/null 2>&1; then
echo "openssl package is not installed. Please install it using 'sudo apt-get install openssl' and try again."
exit 1
fi
# Define prompt with retry function
prompt_with_retry() {
local prompt=$1
local max_attempts=$2
local input
for (( i=1; i<=$max_attempts; i++ )); do
read -p "$prompt" input
if [[ -n "$input" ]]; then
echo "$input"
return 0
else
echo "Input cannot be blank. Please try again."
fi
done
echo "Input cannot be blank. Install terminated."
exit 1
}
# Prompt user for Aliyun AccessKeyId
access_key_id=$(prompt_with_retry "Enter Aliyun AccessKeyId: " 3)
# Prompt user for Aliyun AccessKeySecret
access_key_secret=$(prompt_with_retry "Enter Aliyun AccessKeySecret: " 3)
# Prompt user for Aliyun appkey
app_key=$(prompt_with_retry "Enter Aliyun appkey: " 3)
# Update values in json files
jq --arg access_key_id "$access_key_id" '.AccessKeyId = $access_key_id' sdm/data/nls-cloud-sdm/conf/nlstoken.json > tmp.json && mv tmp.json sdm/data/nls-cloud-sdm/conf/nlstoken.json
jq --arg access_key_secret "$access_key_secret" '.AccessKeySecret = $access_key_secret' sdm/data/nls-cloud-sdm/conf/nlstoken.json > tmp.json && mv tmp.json sdm/data/nls-cloud-sdm/conf/nlstoken.json
jq --arg app_key "$app_key" '.appkey = $app_key' sdm/data/nls-cloud-sdm/conf/service-asr.json > tmp.json && mv tmp.json sdm/data/nls-cloud-sdm/conf/service-asr.json
jq --arg app_key "$app_key" '.appkey = $app_key' sdm/data/nls-cloud-sdm/conf/service-tts.json > tmp.json && mv tmp.json sdm/data/nls-cloud-sdm/conf/service-tts.json
# Generate ESL password hash
esl_password=$(openssl rand -hex 8)
# Update event_socket.conf.xml file
sed -i "s/\(<param name=\"password\" value=\"\)[^\"]*\(\"\/>\)/\1$esl_password\2/" switch/conf/autoload_configs/event_socket.conf.xml
# Update .fs_cli_conf file
sed -i "s/\(password => \)[^\"]*/\1$esl_password/" switch/.fs_cli_conf
# Generate default SIP password hash
default_password=$(openssl rand -hex 8)
# Update vars.xml file
sed -i "s/\(<X-PRE-PROCESS cmd=\"set\" data=\"default_password=\)[^\"]*\(\"\/>\)/\1$default_password\2/" switch/conf/vars.xml
# Extract sounds package
echo "Extracting sounds package..."
cd switch/sounds/
if [[ -f sounds.tar.bz2 ]]; then
tar -xvjf sounds.tar.bz2
echo "Sounds package extracted."
else
echo "Sounds package not found."
cd ../..
exit 1
fi
cd ../..
# Run docker compose
docker compose up -d
echo "ESL Password: $esl_password"
echo "Default SIP Password: $default_password"
echo "Install complete."

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#!/bin/bash
set -e
usage() {
echo "Usage: restart.sh [sdm|switch]"
echo " sdm - Restart 'sdm' service"
echo " switch - Restart 'switch' service"
echo " (none) - Restart both 'sdm' and 'switch' service"
echo " -h - Show this help message"
}
if [[ $# -eq 0 ]]; then
read -p "Do you want to restart both 'sdm' and 'switch' services? (y/n) [default: y]: " choice
choice="${choice:-y}"
if [[ $choice == "y" ]]; then
echo "Restarting 'sdm' service..."
docker restart sdm
echo "Restarting 'switch' service..."
docker restart switch
else
echo "No services restarted."
fi
elif [[ $# -eq 1 ]]; then
if [[ $1 == "sdm" ]]; then
echo "Restarting 'sdm' service..."
docker restart sdm
elif [[ $1 == "switch" ]]; then
echo "Restarting 'switch' service..."
docker restart switch
elif [[ $1 == "-h" ]]; then
usage
else
echo "Invalid argument: $1"
usage
fi
else
echo "Invalid number of arguments"
usage
fi

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<?xml version="1.0" encoding="UTF-8"?>
<!-- UniMRCP client document -->
<unimrcpclient xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:noNamespaceSchemaLocation="unimrcpclient.xsd" version="1.0" subfolder="client-profiles">
<properties>
<!-- If the attribute "type" is set to "auto", IP address is determined implicitly by the hostname. This is the default setting. -->
<ip type="auto"/>
<!-- If the attribute "type" is set to "iface", IP address is determined by the specified name of network interface/adapter. -->
<!-- <ip type="iface">eth0</ip>-->
<!-- IP address can also be specified explicitly. -->
<!--<ip>127.0.0.1</ip>-->
<!-- <ext-ip>a.b.c.d</ext-ip> -->
<server-ip>127.0.0.1</server-ip>
</properties>
<components>
<!-- Factory of MRCP resources -->
<resource-factory>
<resource id="speechsynth" enable="true"/>
<resource id="speechrecog" enable="true"/>
<resource id="recorder" enable="true"/>
<resource id="speakverify" enable="true"/>
</resource-factory>
<!-- SofiaSIP MRCPv2 signaling agent -->
<sip-uac id="SIP-Agent-1" type="SofiaSIP">
<!-- By default, "ip" and "ext-ip" addresses set in the properties will be used.
They still can be explicitly specified (overriden) here.
-->
<!-- <sip-ip>10.10.0.1</sip-ip> -->
<!-- <sip-ext-ip>a.b.c.d</sip-ext-ip> -->
<sip-port>0</sip-port>
<sip-transport>udp,tcp</sip-transport>
<ua-name>UniMRCP SofiaSIP</ua-name>
<sdp-origin>AliMrcpClient</sdp-origin>
<!-- <sip-t1>500</sip-t1> -->
<!-- <sip-t2>4000</sip-t2> -->
<!-- <sip-t4>4000</sip-t4> -->
<!-- <sip-t1x64>32000</sip-t1x64> -->
<!-- <sip-timer-c>185000</sip-timer-c> -->
<!-- <sip-message-output>true</sip-message-output> -->
<!-- <sip-message-dump>sofia-sip-uac.log</sip-message-dump> -->
</sip-uac>
<!-- UniRTSP MRCPv1 signaling agent -->
<rtsp-uac id="RTSP-Agent-1" type="UniRTSP">
<max-connection-count>100</max-connection-count>
<!-- <request-timeout>5000</request-timeout> -->
<sdp-origin>AliMrcpClient</sdp-origin>
</rtsp-uac>
<!-- MRCPv2 connection agent -->
<mrcpv2-uac id="MRCPv2-Agent-1">
<max-connection-count>100</max-connection-count>
<offer-new-connection>false</offer-new-connection>
<rx-buffer-size>1024</rx-buffer-size>
<tx-buffer-size>1024</tx-buffer-size>
<!-- <request-timeout>5000</request-timeout> -->
</mrcpv2-uac>
<!-- Media processing engine -->
<media-engine id="Media-Engine-1">
<realtime-rate>1</realtime-rate>
</media-engine>
<!-- Factory of RTP terminations -->
<rtp-factory id="RTP-Factory-1">
<!-- By default, "ip" and "ext-ip" addresses set in the properties will be used.
They still can be explicitly specified (overriden) here.
-->
<!-- <rtp-ip>10.10.0.1</rtp-ip> -->
<!-- <rtp-ext-ip>a.b.c.d</rtp-ext-ip> -->
<rtp-port-min>6000</rtp-port-min>
<rtp-port-max>7000</rtp-port-max>
</rtp-factory>
</components>
<settings>
<!-- common (default) RTP/RTCP settings -->
<rtp-settings id="RTP-Settings-1">
<jitter-buffer>
<adaptive>1</adaptive>
<playout-delay>50</playout-delay>
<max-playout-delay>600</max-playout-delay>
<time-skew-detection>1</time-skew-detection>
</jitter-buffer>
<ptime>20</ptime>
<codecs>PCMU PCMA L16/96/8000 telephone-event/101/8000</codecs>
<!-- <codecs>PCMU PCMA L16/96/8000 PCMU/97/16000 PCMA/98/16000 L16/99/16000</codecs> -->
<!-- enable/disable RTCP support -->
<rtcp enable="false">
<!-- RTCP BYE policies (RTCP must be enabled first)
0 - disable RTCP BYE
1 - send RTCP BYE at the end of session
2 - send RTCP BYE also at the end of each talkspurt (input)
-->
<rtcp-bye>1</rtcp-bye>
<!-- rtcp transmission interval in msec (set 0 to disable) -->
<tx-interval>5000</tx-interval>
<!-- period (timeout) to check for new rtcp messages in msec (set 0 to disable) -->
<rx-resolution>1000</rx-resolution>
</rtcp>
</rtp-settings>
</settings>
</unimrcpclient>

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<?xml version="1.0" encoding="UTF-8"?>
<alimrcp-server xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:noNamespaceSchemaLocation="alimrcp-server.xsd" version="1.0">
<properties>
<!--<ip type="auto"/>-->
<!-- <ip type="iface">eth0</ip>-->
<ip>0.0.0.0</ip>
<!-- <ext-ip>a.b.c.d</ext-ip> -->
</properties>
<components>
<resource-factory>
<resource id="speechsynth" enable="true"/>
<resource id="speechrecog" enable="true"/>
<resource id="recorder" enable="false"/>
<resource id="speakverify" enable="false"/>
</resource-factory>
<!-- SofiaSIP MRCPv2 signaling agent -->
<sip-uas id="SIP-Agent-1" type="SofiaSIP">
<!-- By default, "ip" and "ext-ip" addresses, set in the properties, are used. These parameters can
explicitly be specified per "sip-uas" by means of "sip-ip" and "sip-ext-ip" correspondingly. -->
<!-- <sip-ip>10.10.0.1</sip-ip> -->
<!-- <sip-ext-ip>a.b.c.d</sip-ext-ip> -->
<sip-port>7010</sip-port>
<sip-transport>udp,tcp</sip-transport>
<!-- <force-destination>true</force-destination> -->
<ua-name>UniMRCP SofiaSIP</ua-name>
<sdp-origin>AliMrcpServer</sdp-origin>
<!-- <sip-t1>500</sip-t1> -->
<!-- <sip-t2>4000</sip-t2> -->
<!-- <sip-t4>4000</sip-t4> -->
<!-- <sip-t1x64>32000</sip-t1x64> -->
<!-- default 600 = 5min, now 3 years-->>
<sip-session-expires>189216000</sip-session-expires>
<sip-min-session-expires>120</sip-min-session-expires>
<!-- <sip-message-output>true</sip-message-output> -->
<!-- <sip-message-dump>sofia-sip-uas.log</sip-message-dump> -->
<!-- <mrcp-draft11-version>true</mrcp-draft11-version> -->
</sip-uas>
<!-- UniRTSP MRCPv1 signaling agent -->
<rtsp-uas id="RTSP-Agent-1" type="UniRTSP">
<rtsp-port>1554</rtsp-port>
<resource-map>
<param name="speechsynth" value="speechsynthesizer"/>
<param name="speechrecog" value="speechrecognizer"/>
</resource-map>
<max-connection-count>100</max-connection-count>
<inactivity-timeout>600</inactivity-timeout>
<sdp-origin>AliMrcpServer</sdp-origin>
</rtsp-uas>
<!-- MRCPv2 connection agent -->
<mrcpv2-uas id="MRCPv2-Agent-1">
<!-- <mrcp-ip>10.10.0.1</mrcp-ip> -->
<mrcp-port>1544</mrcp-port>
<max-connection-count>300</max-connection-count>
<max-shared-use-count>100</max-shared-use-count>
<force-new-connection>true</force-new-connection> <!-- 是否禁止连接复用true表示禁止 -->
<rx-buffer-size>1024</rx-buffer-size> <!-- 表示接收到的MRCP消息的最大长度超过则多次接收 -->
<tx-buffer-size>1024</tx-buffer-size> <!-- 表示发送时的MRCP消息的最大长度超过则多次发送 -->
<inactivity-timeout>600</inactivity-timeout>
<termination-timeout>3</termination-timeout>
</mrcpv2-uas>
<media-engine id="Media-Engine-1">
<realtime-rate>1</realtime-rate>
</media-engine>
<!-- Factory of RTP terminations -->
<rtp-factory id="RTP-Factory-1">
<rtp-port-min>10000</rtp-port-min>
<rtp-port-max>20000</rtp-port-max>
</rtp-factory>
<!-- Factory of plugins (MRCP engines) -->
<plugin-factory>
<engine id="alimrcp-tts" name="alimrcp_tts" enable="true">
<max-channel-count>100</max-channel-count>
<param name="sdk-log-level" value="5"/>
<param name="sdm-metrics-host" value="0.0.0.0:7009"/> <!-- 默认监听到本机所有ip上的7009端口, 可修改或置空(不监控) -->
<param name="tts-save-audio" value="0"/> <!-- 默认不保存tts合成的录音, 如需保存则改为1 -->
<param name="text-encoding-gb2312" value="0"/> <!-- 表示接收到ivr的合成文本编码方式, 默认为0, 即utf8, 如果是gb2312需修改该参数为1 -->
</engine>
<engine id="alimrcp-asr" name="alimrcp_asr" enable="true">
<max-channel-count>100</max-channel-count>
<param name="sdk-log-level" value="5"/>
<param name="sdm-metrics-host" value="0.0.0.0:7009"/> <!-- 默认监听到本机所有ip上的7009端口, 可修改或置空(不监控) -->
<param name="text-encoding-gb2312" value="0"/>
<param name="wav-uri-prefix" value=""/>
<param name="recognize-mode-continuous" value="0"/> <!-- 无需修改, 如要修改建议同时修改sip-session-expires为一个无限大的值 -->
<param name="no-input-timeout" value="5000"/> <!-- 无话超时时间, 单位毫秒, 不可过低, 如果ivr传递了, 则以ivr传递的为准 -->
<param name="speech-complete-timeout" value="800"/> <!-- vad尾点断句间隔, 单位毫秒, 范围[200,2000], 如果ivr传递了, 则以ivr传递的为准 -->
<!--<param name="sensitivity-level" value="0.2"/>--> <!-- 噪音阈值参数、灵敏度参数可以参考标准RFC协议取值范围[-1,1]等同asr的speech_noise_threshold -->
<param name="speech-cache-count" value="10"/> <!-- should be greater than 0, suggest [1,20] -->
<param name="save-waveform" value="1"/> <!-- 是否保存用户录音默认保存但还受IVR传参影响 -->
<param name="ignore-ivr-save-waveform" value="0"/> <!-- 是否忽略IVR传递的save-waveform参数默认不忽略若要忽略请置为1 -->
<param name="return-more-detail" value="1"/> <!-- 是否返回更多ASR相关信息, 默认返回 -->
<param name="support-message-body" value="0"/> <!-- 当no-input-timeout或者no-match等状态时是否返回消息体默认没有-->
<param name="force-stop-recognize" value="1"/>
<param name="xml-type" value="application/nlsml+xml"/>
<param name="asr-result-template" value="../conf/asr_result_template.xml"/>
<!--<param name="grammar-transformer" value="../script/RequestTransformer.lua"/>-->
<param name="script-when-hangup" value="HangupNotifierDemo"/> <!-- python script -->
<param name="script-function-when-hangup" value=""/> <!-- function of python script -->
<param name="script-of-nlu" value="NluClientDemo"/> <!-- nlu python script -->
<param name="script-function-of-nlu" value=""/> <!-- nlu function of python script -->
</engine>
</plugin-factory>
</components>
<settings>
<!-- RTP/RTCP settings -->
<rtp-settings id="RTP-Settings-1">
<jitter-buffer>
<adaptive>1</adaptive>
<playout-delay>50</playout-delay>
<max-playout-delay>600</max-playout-delay>
<time-skew-detection>1</time-skew-detection>
</jitter-buffer>
<ptime>20</ptime>
<codecs own-preference="false">PCMU PCMA L16/96/8000 telephone-event/101/8000</codecs>
<!-- Enable/disable RTCP support -->
<rtcp enable="false">
<!--
RTCP BYE policies (RTCP must be enabled first)
0 - disable RTCP BYE
1 - send RTCP BYE at the end of session
2 - send RTCP BYE also at the end of each talkspurt (input)
-->
<rtcp-bye>1</rtcp-bye>
<!-- RTCP transmission interval in msec (set 0 to disable) -->
<tx-interval>5000</tx-interval>
<!-- Period (timeout) to check for new RTCP messages in msec (set 0 to disable) -->
<rx-resolution>1000</rx-resolution>
</rtcp>
</rtp-settings>
</settings>
<profiles>
<!-- MRCPv2 default profile -->
<mrcpv2-profile id="uni2">
<sip-uas>SIP-Agent-1</sip-uas>
<mrcpv2-uas>MRCPv2-Agent-1</mrcpv2-uas>
<media-engine>Media-Engine-1</media-engine>
<rtp-factory>RTP-Factory-1</rtp-factory>
<rtp-settings>RTP-Settings-1</rtp-settings>
</mrcpv2-profile>
<!-- MRCPv1 default profile -->
<mrcpv1-profile id="uni1">
<rtsp-uas>RTSP-Agent-1</rtsp-uas>
<media-engine>Media-Engine-1</media-engine>
<rtp-factory>RTP-Factory-1</rtp-factory>
<rtp-settings>RTP-Settings-1</rtp-settings>
</mrcpv1-profile>
</profiles>
</alimrcp-server>

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<!--
此模板用以描述,如何在特殊项目里调整报文返回格式,您可以
1. 拷贝此文件并重命名为asr_result_template.xml
2. 删除文件中相关的注释信息
3. 按照实际需要调整xml结构、删除不需要的变量等
-->
<?xml version="1.0" encoding="utf-8"?>
<result>
<interpretation grammar="$$asr_grammar_id$$" confidence="100">
<instance>
<!--
$$XXXX$$ 是变量内容sdm内部会将其替换为实际有效数据支持的所有变量均在下方列出实际使用时可以按需删掉不需要的变量其中
1. $$asr_result$$ : 表示ASR识别结果可能为空
2. $$asr_begin_time$$ : 表示ASR有效语音的相对开始时间
3. $$asr_end_time$$ : 表示ASR有效语音的相对结束时间
4. $$asr_task_id$$ : 表示本次ASR识别的task_id可以用来和服务端各链路进行关联
5. $$asr_code$$ : 表示示本次ASR识别的的状态码等同于实时转写状态码
6. $$asr_code_text$$ : 表示示本次ASR识别的的状态码对应的文本描述等同于实时转写
7. $$mrcp_code$$ : 表示mrcp协议定义的状态码比如 000 001 002 006等
8. $$mrcp_code_text$$ : 表示mrcp协议定义的状态码所对应的文本描述比如success、no-match、no-input-timeout、recognizer-error等
9. $$asr_waveform_path$$ : 表示本次ASR识别保存下来的原始录音(ivr传递过来的),命名一般是: {session id}-index.wav
10. $$asr_waveform_uri$$ : 表示本次ASR识别保存下来的原始录音(ivr传递过来的),命名一般是:{wav-uri-prefix}/{session id}-index.wav
-->
<result>$$asr_result$$</result>
<beginTime>$$asr_begin_time$$</beginTime>
<endTime>$$asr_end_time$$</endTime>
<taskId>$$asr_task_id$$</taskId>
<asrCode>$$asr_code$$</asrCode>
<asrCodeText>$$asr_code_text$$</asrCodeText>
<mrcpCode>$$mrcp_code$$</mrcpCode>
<mrcpCodeText>$$mrcp_code_text$$</mrcpCodeText>
<waveformPath>$$asr_waveform_path$$</waveformPath>
<waveformUri>$$asr_waveform_uri$$</waveformUri>
</instance>
<input mode="speech">$$asr_result$$</input>
</interpretation>
</result>

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<?xml version="1.0" encoding="UTF-8"?>
<!-- UniMRCP client document -->
<unimrcpclient xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
xsi:noNamespaceSchemaLocation="../unimrcpclient.xsd"
version="1.0">
<settings>
<!-- SIP MRCPv2 settings -->
<sip-settings id="UniMRCP-SIP-Settings">
<!--
Server IP address can explicitly be specified per "sip-settings". Otherwise, the server IP
address defaults to "server-ip" set in the properties, which in turn defaults to "ip".
-->
<!-- <server-ip>10.10.0.1</server-ip> -->
<server-port>7010</server-port>
<!-- <force-destination>true</force-destination> -->
</sip-settings>
<!-- RTSP MRCPv1 settings -->
<rtsp-settings id="UniMRCP-RTSP-Settings">
<!--
Server IP address can explicitly be specified per "rtsp-settings". Otherwise, the server IP
address defaults to "server-ip" set in the properties, which in turn defaults to "ip".
-->
<!-- <server-ip>10.10.0.1</server-ip> -->
<server-port>1554</server-port>
<!-- <force-destination>true</force-destination> -->
<resource-location>media</resource-location>
<resource-map>
<param name="speechsynth" value="speechsynthesizer"/>
<param name="speechrecog" value="speechrecognizer"/>
</resource-map>
</rtsp-settings>
</settings>
<profiles>
<!-- UniMRCP MRCPv2 profile -->
<mrcpv2-profile id="uni2">
<sip-uac>SIP-Agent-1</sip-uac>
<mrcpv2-uac>MRCPv2-Agent-1</mrcpv2-uac>
<media-engine>Media-Engine-1</media-engine>
<rtp-factory>RTP-Factory-1</rtp-factory>
<sip-settings>UniMRCP-SIP-Settings</sip-settings>
<rtp-settings>RTP-Settings-1</rtp-settings>
</mrcpv2-profile>
<!-- UniMRCP MRCPv1 profile -->
<mrcpv1-profile id="uni1">
<rtsp-uac>RTSP-Agent-1</rtsp-uac>
<media-engine>Media-Engine-1</media-engine>
<rtp-factory>RTP-Factory-1</rtp-factory>
<rtsp-settings>UniMRCP-RTSP-Settings</rtsp-settings>
<rtp-settings>RTP-Settings-1</rtp-settings>
</mrcpv1-profile>
<!-- More profiles may follow. -->
</profiles>
</unimrcpclient>

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<?xml version="1.0" encoding="UTF-8"?>
<!-- Logger Document -->
<aptlogger>
<!-- Set the log output mode
CONSOLE console output
FILE log file output
CONSOLE,FILE both console and the log file output
-->
<output>CONSOLE</output>
<!-- default 1 GB = 1024 * 1024 * 1024 -->
<size>1073741824</size>
<!-- Set the format of the log messages
DATE date output
TIME time output
PRIORITY priority output
MARK file:line marker output
THREAD calling thread identifier output
-->
<headers>DATE,TIME,PRIORITY</headers>
<!-- Set the priority of the default log source to one of
EMERGENCY system is unusable
ALERT action must be taken immediately
CRITICAL critical condition
ERROR error condition
WARNING warning condition
NOTICE normal, but significant condition
INFO informational message
DEBUG debug-level message
-->
<priority>INFO</priority>
<!-- Set the masking mode of the default log source to one of
NONE log everything as is
COMPLETE mask private data completely
ENCRYPTED encrypt private data
-->
<masking>NONE</masking>
<!--
Besides the default log source, there can be additional log sources,
which may have different priority levels and log masking modes set.
This allows to apply flexible filtering rules per log source.
-->
<sources>
<source name="MPF" priority="INFO" masking="NONE"/>
<source name="UNIRTSP" priority="INFO" masking="NONE"/>
<source name="SOFIASIP" priority="INFO" masking="NONE"/>
</sources>
</aptlogger>

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user admin;
worker_processes 1;
worker_rlimit_nofile 100000;
error_log "pipe:/opt/taobao/install/cronolog/sbin/cronolog /home/admin/cai/logs/cronolog/%Y/%m/%Y-%m-%d-error_log" warn;
pid /home/admin/cai/tengine.pid;
events {
use epoll;
worker_connections 20480;
}
http {
include mime.types;
default_type application/octet-stream;
root htdocs;
sendfile on;
tcp_nopush on;
server_tokens off;
keepalive_timeout 0;
client_header_timeout 1m;
send_timeout 1m;
client_max_body_size 100m;
variables_hash_max_size 1024;
index index.html index.htm;
log_format proxyformat "$remote_addr $request_time_usec $http_x_readtime [$time_local] \"$request_method http://$host:$server_port:$request_uri\" $status $body_bytes_sent \"$http_referer\" \"$http_user_agent\" \"$md5_encode_cookie_unb\" \"$md5_encode_cookie_cookie2\"";
#access_log "pipe:/opt/taobao/install/cronolog/sbin/cronolog /home/admin/cai/logs/cronolog/%Y/%m/%Y-%m-%d-%H-taobao-access_log" proxyformat;
log_not_found off;
gzip on;
gzip_http_version 1.0;
gzip_comp_level 6;
gzip_min_length 1024;
gzip_proxied any;
gzip_vary on;
gzip_disable msie6;
gzip_buffers 96 8k;
gzip_types text/xml text/plain text/css application/javascript application/x-javascript application/rss+xml application/json;
proxy_redirect off;
proxy_buffers 128 8k;
proxy_intercept_errors on;
proxy_headers_hash_max_size 51200;
proxy_headers_hash_bucket_size 6400;
proxy_set_header Host $host;
proxy_set_header X-Real-IP $remote_addr;
proxy_set_header Web-Server-Type nginx;
proxy_set_header WL-Proxy-Client-IP $remote_addr;
proxy_set_header X-Forwarded-For $proxy_add_x_forwarded_for;
proxy_read_timeout 5m;
#websocket
map $http_upgrade $connection_upgrade {
default upgrade;
'' close;
}
server {
listen 7008 default_server;
location /metrics {
proxy_pass http://127.0.0.1:7009/metrics;
}
location /sdm_metrics {
proxy_pass http://127.0.0.1:7009/metrics;
}
location ~ /maybeOtherPath/prefix {
rewrite '^/maybeOtherPath/prefix/(.*)$' /$1;
}
location ~ \.(fla|pcm|mp3|wav)$ {
#access_log assets.access.log;
default_type audio/x-wav;
expires 180d;
root /home/admin/nls-cloud-sdm/var/;
if ($arg_download){
add_header Content-Disposition "attachment; filename=$arg_download";
}
}
}
}

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{
"AccessKeyId": "",
"AccessKeySecret": "",
"Domain": "nls-meta.cn-shanghai.aliyuncs.com",
"ServerVersion": "2018-05-18",
"ServerResourcePath": "/pop/2018-05-18/tokens",
"IntervalTime": 72000,
"DefaultToken": "",
"Desc": "私有云调用情况下,仅仅设置DefalutToken字段为default其他字段不要求公有云调用情况下务必设置DefaultToken为空而除DefaultToken之外的其他所有字段务必不为空"
}

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alog.rootLogger=INFO
alog.max_msg_len=1310720
##### CONSOLE LOG ######
alog.appender.consoleAppender=ConsoleAppender
alog.appender.consoleAppender.layout.LogPattern=[%%d] [%%l] [%%t,%%F -- %%f():%%n] [%%m]
###### ASR LOG #######
alog.appender.asrAppender=FileAppender
alog.appender.asrAppender.fileName=../log/sdm-asr-request.log
alog.appender.asrAppender.max_file_size=128
alog.appender.asrAppender.delay_time=24
alog.appender.asrAppender.async_flush=true
alog.appender.asrAppender.layout=PatternLayout
alog.appender.asrAppender.layout.LogPattern=%%m
alog.appender.asrAppender.log_keep_count=25
###### TTS LOG #######
alog.appender.ttsAppender=FileAppender
alog.appender.ttsAppender.fileName=../log/sdm-tts-request.log
alog.appender.ttsAppender.max_file_size=128
alog.appender.ttsAppender.delay_time=24
alog.appender.ttsAppender.async_flush=true
alog.appender.ttsAppender.layout=PatternLayout
alog.appender.ttsAppender.layout.LogPattern=%%m
alog.appender.ttsAppender.log_keep_count=25
###### LOG CONTROL ######
alog.logger.sdm.asr=INFO, asrAppender
alog.logger.sdm.tts=INFO, ttsAppender

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{
"url": "wss://nls-gateway.cn-shanghai.aliyuncs.com/ws/v1",
"appkey": "",
"format": "pcm",
"sample_rate": 8000,
"enable_intermediate_result": 1,
"enable_punctuation_prediction": 1,
"enable_inverse_text_normalization": 1,
"enable_semantic_sentence_detection": 0,
"enable_ignore_sentence_timeout": 1,
"enable_ner": 0,
"ner_name": "aca",
"enbale_gender_detect": 0,
"desc": {
"notice": "此处仅仅是描述信息,无需修改",
"url": "访问ASR的url, 如果是调用公有云ASR无需变动如果是调用私有云ASR需根据现场部署情况进行设置一般为:vip://gateway-ip.vipserver_1:port_1,gateway-ip.vipserver_2:port_2/vipTargetDomain其中vipServerIp:port可以有一个或者多个用逗号分割端口不指定默认为80最后指定vipServer的Domain",
"appkey": "appkey, 调用公有云ASR时需要在阿里云官网获取, 调用私有云ASR时固定设置为(全小写):default",
"format": "编码格式pcm或opu, 保持pcm不变",
"sample_rate": "采样率, 8000或16000, 保持8000不变",
"enable_intermediate_result": "设置是否返回中间识别结果, 可选参数. 服务端默认0",
"enable_punctuation_prediction": "设置是否在识别结果中添加标点, 可选参数. 服务端默认0",
"enable_inverse_text_normalization": "设置是否在识别结果中执行文本正则化/数字转换, 比如'一千六百五十'会返回'1650', 可选参数. 服务端默认0",
"enable_semantic_sentence_detection": "设置是否使用语义断句, 可选参数. 服务端默认0, 保持配置0不变",
"customization_id": "设置热词定制模型id, 默认不设置除非已经训练了定制模型且通过本配置生效则需要在上面json格式上增加",
"vocabulary_id": "设置热词id, 默认不设置除非已经训练了定制热词且通过本配置生效则需要在上面json格式上增加",
"class_vocabulary_id": "设置类热词id, 默认不设置除非已经训练了定制类热词且通过本配置生效则需要在上面json格式上增加",
"als_am_id": "在am混部时候设置的am模型id",
"enable_ner": "启动ner、地址解析功能默认不启用",
"enbale_gender_detect": "是否开启性别识别,默认不启用"
}
}

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{
"url": "wss://nls-gateway.cn-shanghai.aliyuncs.com/ws/v1",
"appkey": "",
"format": "pcm",
"sample_rate": 8000,
"voice": "xiaoyun",
"volume": 50,
"speech_rate": 0,
"pitch_rate": 0,
"method": 0,
"desc": {
"url": "访问TTS的url, 如果是调用公有云TTS无需变动如果是调用私有云TTS需根据现场部署情况进行设置一般为:vip://gateway-ip.vipserver_1:port_1,gateway-ip.vipserver_2:port_2/vipTargetDomain其中vipServerIp:port可以有一个或者多个用逗号分割端口不指定默认为80最后指定vipServer的Domain",
"appkey": "appkey, 调用公有云TTS时需要在阿里云官网获取, 调用私有云TTS时固定设置为(全小写):default",
"format": "编码格式支持pcm, wav, mp3, 保持pcm不变",
"sample_rate": "采样率, 8000或16000, 保持8000不变",
"volume": "音量, 范围是0~100, 可选参数, 默认50",
"voice": "发音人, 支持xiaoyun(女), xiaogang(男)等",
"speech_rate": "语速, 范围是-500~500, 可选参数, 默认是0",
"pitch_rate": "语调, 范围是-500~500, 可选参数, 默认是0",
"method": "合成方法, 可选参数, 默认是0. 参数含义0:不带录音的参数合成; 1:带录音的拼接合成; 2:不带录音的拼接合成; 3:带录音的参数合成",
"notice": "此处仅仅是描述信息,无需修改"
}
}

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# coding=utf-8
import os
import time
import sys
# Demo示例需要在具体项目具体场景下自行实现相关调用逻辑
def hangupNotify(channelId, vendorParams, asrResult, asrBeginTime, asrEndTime):
# channelId 唯一sessionid 是mrcp-server产生的
# vendorParams ivr发送给mrcp-server的RECOGNIZE消息时携带的扩展参数
# asrResult : 识别结果
# asrBeginTime : 识别结果的开始时间
# asrEndTime : 识别结果的结束时间
print "handupNotify = ", channelId, vendorParams, asrResult, asrBeginTime, asrEndTime
if len(asrResult) == 0 or (asrBeginTime < 0 and asrEndTime < 0):
# 说明挂机时没有识别结果
return
# 挂机时可能有识别结果需要将该结果抛给nlu
# TODO
if __name__ == '__main__':
hangupNotify("123f", "ke=1", "hi", 124, -1)
print "Game Over"

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# coding=utf-8
import os
import time
import sys
import urllib
import json
import requests
#import xmltodict
# Demo示例需要在具体项目具体场景下自行实现相关调用逻辑
class XiaoMiNlu:
def __init__(self):
self.nlpUrl = 'http://39.106.157.134:8110/chatbot/chat/ivr?'
self.appId = 402
def makeUrl(self, sessionId, asrResult, asrBeginTime, asrEndTime, emotionValue, vendorParams):
#http://39.106.157.134:8110/chatbot/chat/ivr?AppId=402&Utterance=%E4%BD%A0%E5%A5%BD&VendorParam={%22index%22:%221%22}&SessionId=ee8a166cd1cc4bab
url = self.nlpUrl + "AppId=" + str(self.appId) + "&Utterance=" + urllib.quote(asrResult)
if sessionId is not None and sessionId != "":
url = url + "&SessionId=" + sessionId
if vendorParams is not None and vendorParams != "":
#按需将 k1=v1;k2=v2;k3=v3的形式转成json格式(云小蜜调用要求)
items = vendorParams.split(';')
d = dict()
for item in items:
key,value = item.split('=')
d[key] = value
url = url + "&VendorParam=" + json.dumps(d)
return url
def request(self, ssionId, asrResult, asrBeginTime, asrEndTime, emotionValue, vendorParams):
url = self.makeUrl(ssionId, asrResult, asrBeginTime, asrEndTime, emotionValue, vendorParams)
req = requests.get(url)
if(req.status_code != 200):
print req.text
print "ERROR url(%s), request Fail(%d)" % (url, req.status_code)
return ""
'''
{
"SessionId": "123f",
"MessageId": "601762b2951e4a2381e26ec0a2703fc7",
"Messages": [
{
"Type": "Text",
"Text": {
"Content": "{\"INTEL_CUST_HEAD\":{\"RESP_DESC\":\"\",\"APP_ID\":\" \",\"RESP_CODE\":\"\",\"TIMESTAMP\":\"\",\"TRANS_ID\":\"\"},\"INTEL_CUST_BODY\":{\"RESP_DESC\":\"\",\"OPERTYPE\":\"0\",\"RESP_CODE\":\"0000\",\"BROADCAST1\":\"engine \",\"INTERRUPT1\":\"\"}}",
"AnswerSource": "ChitChat"
}
}
]
}
'''
# 按需对NLU的返回进行解析获取需要关注的内容并打包为xml格式返回此处示例只获取"Messages->Text->Content"字段对应的值
print "result = ", req.text
jss = json.loads(req.text.encode('utf-8'))
reply = ""
if(jss.has_key('Messages') and jss['Messages'][0].has_key('Text')):
result = jss['Messages'][0]['Text']
if result.has_key('Content'):
reply = result['Content'].encode('utf-8')
# 把 json格式转为xml格式(MRCPSERVER和IVR交互时只有XML格式)
reply_json = json.loads(reply)
#print type(reply_json), reply_json
#xmlstr = xmltodict.unparse(reply_json, full_document=False)
xmlstr = "<f>fff</f>"
#print "xmlstr = ", xmlstr
return xmlstr
def nluRequest(sessionId, asrResult, asrBeginTime, asrEndTime, emotionValue, vendorParams):
print "python nluRequest = ", sessionId, asrResult, asrBeginTime, asrEndTime, emotionValue, vendorParams
''' 需要NLU服务关注asrResult的取值范围
NO-MTACH : 检测到了语音(vad检测到了起始点)但ASR并没有得到有效的结果这种一般是客户声音过小背景音过大声音有能量但无法听到实际内容导致
NO-INPUT-TIMEOUT : 无话超时也即用户侧没有说法全是静音
ASR-ERROR : 调用ASR进行识别可能需要NLU服务进行兜底处理
其他情况 : 用户语音的ASR识别结果
'''
if asrResult == "NO-MTACH":
print "check speech, but no valid result!"
elif asrResult == "NO-INPUT-TIMEOUT":
print "check no speech, user may be not speak!"
elif asrResult == "ASR-ERROR":
print "call ASR failure, you should hold this!"
else:
print "check speech, and recognized success: ", asrResult
#time.sleep(1)
# 注意:一定要返回如下格式
# 返回<k1>fff</k1>类似结构该结构即是xml格式(去掉xml头))
return "<key>value</key>"
nlu = XiaoMiNlu()
return nlu.request(sessionId, asrResult, asrBeginTime, asrEndTime, emotionValue, vendorParams)
if __name__ == '__main__':
print nluRequest("123f", "你好", 1234, 4569, 5, "ke=1;phone=123;callid=zxcvb"), "\n"
print nluRequest("123f", "WELCOME", 1234, 4569, 0, "ke=1"), "\n"
print nluRequest("123f", "NO-MTACH", 1234, 4569, 0, "ke=1"), "\n"
print nluRequest("123f", "NO-INPUT-TIMEOUT", 1234, 1, 4569, "ke=1"), "\n"
print nluRequest("123f", "ASR-ERROR", 1234, 4569, 0, ""), "\n"
print "Game Over"

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function grammarTransformer(grammar)
-- see : https://www.runoob.com/lua/lua-tables.html
--print("c++ call lua function parse")
mytable = {}
--mytable["SPEECH_URL"] = "custom:ws://11.164.63.252:8101/ws/v1"
--print("mytable 索引为 1 的元素是 ", mytable[1])
--print("mytable 索引为 wow 的元素是 ", mytable["wow"])
return mytable
end
function split(str, reps)
local resultStrList = {}
string.gsub(str, '[^'..reps..']+', function (w)
table.insert(resultStrList, w)
end)
return resultStrList
end
function vendorParamsTransformer(vendorParams)
-- vendorParams = k1=v1;k2=v2;k3=v3
-- 将verndorParam按需拆分为key:value对并可在此处进行相应修改
--print("input : ", vendorParams)
mytable = {}
local list = split(vendorParams, ";")
for _, s in ipairs(list) do
print(s)
local pos = string.find(s, '=')
if pos then
local key = string.sub(s, 0, pos - 1)
local value = string.sub(s, pos + 1)
--print("key = ", key, ", value = ", value)
mytable[key] = value
end
end
--mytable["SPEECH_VOCAB_ID"] = "11111"
return mytable
end

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2
sdm/logs/nls-cloud-sdm/.gitignore vendored Executable file
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*
!.gitignore

5
switch/.fs_cli_conf Normal file
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[default]
host => 127.0.0.1
port => 62221
password => 305852127a3c0a6c
debug => 6

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<configuration name="abstraction.conf" description="Abstraction">
<apis>
<api name="user_name" description="Return Name for extension" syntax="&lt;exten&gt;" parse="(.*)" destination="user_data" argument="$1@default var effective_caller_id_name"/>
</apis>
</configuration>

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<configuration name="acl.conf" description="Network Lists">
<network-lists>
<!--
These ACL's are automatically created on startup.
rfc1918.auto - RFC1918 Space
nat.auto - RFC1918 Excluding your local lan.
localnet.auto - ACL for your local lan.
loopback.auto - ACL for your local lan.
-->
<list name="lan" default="allow">
<!-- <node type="deny" cidr="192.168.42.0/24"/> -->
<!-- <node type="allow" cidr="192.168.42.42/32"/> -->
</list>
<!--
This will traverse the directory adding all users
with the cidr= tag to this ACL, when this ACL matches
the users variables and params apply as if they
digest authenticated.
-->
<list name="domains" default="deny">
<!-- domain= is special it scans the domain from the directory to build the ACL -->
<node type="allow" domain="$${domain}"/>
<!-- use cidr= if you wish to allow ip ranges to this domains acl. -->
<!-- <node type="allow" cidr="192.168.0.0/24"/> -->
<node type="allow" cidr="127.0.0.1/32"/>
</list>
</network-lists>
</configuration>

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<configuration name="alsa.conf" description="Soundcard Endpoint">
<settings>
<!--Default dialplan and caller-id info -->
<param name="dialplan" value="XML"/>
<param name="cid-name" value="N800 Alsa"/>
<param name="cid-num" value="5555551212"/>
<!--audio sample rate and interval -->
<param name="sample-rate" value="8000"/>
<param name="codec-ms" value="20"/>
</settings>
</configuration>

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<configuration name="amqp.conf" description="mod_amqp">
<producers>
<profile name="default">
<connections>
<connection name="primary">
<param name="hostname" value="localhost"/>
<param name="virtualhost" value="/"/>
<param name="username" value="guest"/>
<param name="password" value="guest"/>
<param name="port" value="5673"/>
<param name="heartbeat" value="0"/>
</connection>
<connection name="secondary">
<param name="hostname" value="localhost"/>
<param name="virtualhost" value="/"/>
<param name="username" value="guest"/>
<param name="password" value="guest"/>
<param name="port" value="5672"/>
<param name="heartbeat" value="0"/>
</connection>
</connections>
<params>
<param name="exchange-name" value="TAP.Events"/>
<param name="exchange-type" value="topic"/>
<param name="circuit_breaker_ms" value="10000"/>
<param name="reconnect_interval_ms" value="1000"/>
<param name="send_queue_size" value="5000"/>
<param name="enable_fallback_format_fields" value="1"/>
<!-- The routing key is made from the format string, using the header values in the event specified in the format_fields.-->
<!-- Fields that are prefixed with a # are treated as literals rather than doing a header lookup -->
<param name="format_fields" value="#FreeSWITCH,FreeSWITCH-Hostname,Event-Name,Event-Subclass,Unique-ID"/>
<!-- If enable_fallback_format_fields is enabled, then you can | separate event headers, and if the first does not exist
then the system will check additional configured header values.
-->
<!-- <param name="format_fields" value="#FreeSWITCH,FreeSWITCH-Hostname|#Unknown,Event-Name,Event-Subclass,Unique-ID"/> -->
<!-- <param name="event_filter" value="SWITCH_EVENT_ALL"/> -->
<param name="event_filter" value="SWITCH_EVENT_CHANNEL_CREATE,SWITCH_EVENT_CHANNEL_DESTROY,SWITCH_EVENT_HEARTBEAT,SWITCH_EVENT_DTMF"/>
</params>
</profile>
</producers>
<commands>
<profile name="default">
<connections>
<connection name="primary">
<param name="hostname" value="localhost"/>
<param name="virtualhost" value="/"/>
<param name="username" value="guest"/>
<param name="password" value="guest"/>
<param name="port" value="5672"/>
<param name="heartbeat" value="0"/>
</connection>
</connections>
<params>
<param name="exchange-name" value="TAP.Commands"/>
<param name="binding_key" value="commandBindingKey"/>
<param name="reconnect_interval_ms" value="1000"/>
<param name="queue-passive" value="false"/>
<param name="queue-durable" value="false"/>
<param name="queue-exclusive" value="false"/>
<param name="queue-auto-delete" value="true"/>
</params>
</profile>
</commands>
<logging>
<profile name="default">
<connections>
<connection name="primary">
<param name="hostname" value="localhost"/>
<param name="virtualhost" value="/"/>
<param name="username" value="guest"/>
<param name="password" value="guest"/>
<param name="port" value="5672"/>
<param name="heartbeat" value="0"/>
</connection>
</connections>
<params>
<param name="exchange-name" value="TAP.Logging"/>
<param name="send_queue_size" value="5000"/>
<param name="reconnect_interval_ms" value="1000"/>
<param name="log-levels" value="debug,info,notice,warning,err,crit,alert"/>
</params>
</profile>
</logging>
</configuration>

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<configuration name="amr.conf">
<settings>
<!-- AMR modes (supported bitrates) :
mode 0 AMR 4.75 kbps
mode 1 AMR 5.15 kbps
mode 2 AMR 5.9 kbps
mode 3 AMR 6.7 kbps
mode 4 AMR 7.4 kbps
mode 5 AMR 7.95 kbps
mode 6 AMR 10.2 kbps
mode 7 AMR 12.2 kbps
-->
<param name="default-bitrate" value="7"/>
<!-- Enable VoLTE specific FMTP -->
<param name="volte" value="0"/>
<!-- Enable automatic bitrate variation during the call based on RTCP feedback -->
<param name="adjust-bitrate" value="0"/>
<!-- force OA when originating -->
<param name="force-oa" value="0"/>
</settings>
</configuration>

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<configuration name="amrwb.conf">
<settings>
<!-- AMRWB modes (supported bitrates):
mode 8 AMR-WB_23.85 23.85 Kbit/s
mode 7 AMR-WB_23.05 23.05 Kbit/s
mode 7 AMR-WB_19.85 19.85 Kbit/s
mode 6 AMR-WB_18.25 18.25 Kbit/s
mode 5 AMR-WB_15.85 15.85 Kbit/s
mode 4 AMR-WB_14.25 14.25 Kbit/s
mode 3 AMR-WB_12.65 12.65 Kbit/s
mode 2 AMR-WB_8.85 8.85 Kbit/s
mode 1 AMR-WB_6.60 6.60 Kbit/s
-->
<param name="default-bitrate" value="8"/>
<!-- Enable VoLTE specific FMTP -->
<param name="volte" value="1"/>
<!-- Enable automatic bitrate variation during the call based on RTCP feedback -->
<param name="adjust-bitrate" value="0"/>
<!-- force OA when originating -->
<param name="force-oa" value="0"/>
<!-- don't mirror mode-set in SDP answer, but use our own (default-bitrate). -->
<param name="mode-set-overwrite" value="0"/>
</settings>
</configuration>

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<configuration name="avcodec.conf" description="AVCodec Config">
<settings>
<!-- max bitrate the system support, truncate if over limit -->
<!-- <param name="max-bitrate" value="5mb"/> -->
<!-- <param name="rtp-slice-size" value="1200"/> -->
<!-- minimum time to generate a new key frame in ms /> -->
<!-- <param name="key-frame-min-freq" value="250"/> -->
<!-- integer of cpus, or 'auto', or 'cpu/<divisor>/<max> -->
<param name="dec-threads" value="1"/>
<param name="enc-threads" value="cpu/2/4"/>
</settings>
<profiles>
<profile name="H263">
</profile>
<profile name="H263+">
</profile>
<profile name="H264">
<!-- <param name="dec-threads" value="1"/> -->
<!-- <param name="enc-threads" value="cpu/2/4"/> -->
<!-- <param name="profile" value="baseline"/> -->
<!-- <param name="level" value="41"/> -->
<!-- <param name="timebase" value="1/90"/> -->
<!--
#define AV_CODEC_FLAG_UNALIGNED (1 << 0)
#define AV_CODEC_FLAG_QSCALE (1 << 1)
#define AV_CODEC_FLAG_4MV (1 << 2)
#define AV_CODEC_FLAG_OUTPUT_CORRUPT (1 << 3)
#define AV_CODEC_FLAG_QPEL (1 << 4)
#define AV_CODEC_FLAG_PASS1 (1 << 9)
#define AV_CODEC_FLAG_PASS2 (1 << 10)
#define AV_CODEC_FLAG_LOOP_FILTER (1 << 11)
#define AV_CODEC_FLAG_GRAY (1 << 13)
#define AV_CODEC_FLAG_PSNR (1 << 15)
#define AV_CODEC_FLAG_TRUNCATED (1 << 16)
#define AV_CODEC_FLAG_INTERLACED_DCT (1 << 18)
#define AV_CODEC_FLAG_LOW_DELAY (1 << 19)
#define AV_CODEC_FLAG_GLOBAL_HEADER (1 << 22)
#define AV_CODEC_FLAG_BITEXACT (1 << 23)
#define AV_CODEC_FLAG_AC_PRED (1 << 24)
#define AV_CODEC_FLAG_INTERLACED_ME (1 << 29)
#define AV_CODEC_FLAG_CLOSED_GOP (1U << 31)
-->
<param name="flags" value="LOOP_FILTER|PSNR"/>
<!--
#define FF_CMP_SAD 0
#define FF_CMP_SSE 1
#define FF_CMP_SATD 2
#define FF_CMP_DCT 3
#define FF_CMP_PSNR 4
#define FF_CMP_BIT 5
#define FF_CMP_RD 6
#define FF_CMP_ZERO 7
#define FF_CMP_VSAD 8
#define FF_CMP_VSSE 9
#define FF_CMP_NSSE 10
#define FF_CMP_W53 11
#define FF_CMP_W97 12
#define FF_CMP_DCTMAX 13
#define FF_CMP_DCT264 14
#define FF_CMP_MEDIAN_SAD 15
#define FF_CMP_CHROMA 256
-->
<!-- <param name="me-cmp" value="1"/> -->
<!-- <param name="me-range" value="16"/> -->
<!-- <param name="max-b-frames" value="3"/> -->
<!-- <param name="refs" value="3"/> -->
<!-- <param name="gop-size" value="250"/> -->
<!-- <param name="keyint-min" value="25"/> -->
<!-- <param name="i-quant-factor" value="0.71"/> -->
<!-- <param name="b-quant-factor" value="0.76923078"/> -->
<!-- <param name="qcompress" value="0.6"/> -->
<!-- <param name="qmin" value="10"/> -->
<!-- <param name="qmax" value="51"/> -->
<!-- <param name="max-qdiff" value="4"/> -->
<!--
enum AVColorSpace {
AVCOL_SPC_RGB = 0, ///< order of coefficients is actually GBR, also IEC 61966-2-1 (sRGB)
AVCOL_SPC_BT709 = 1, ///< also ITU-R BT1361 / IEC 61966-2-4 xvYCC709 / SMPTE RP177 Annex B
AVCOL_SPC_UNSPECIFIED = 2,
AVCOL_SPC_RESERVED = 3,
AVCOL_SPC_FCC = 4, ///< FCC Title 47 Code of Federal Regulations 73.682 (a)(20)
AVCOL_SPC_BT470BG = 5, ///< also ITU-R BT601-6 625 / ITU-R BT1358 625 / ITU-R BT1700 625 PAL & SECAM / IEC 61966-2-4 xvYCC601
AVCOL_SPC_SMPTE170M = 6, ///< also ITU-R BT601-6 525 / ITU-R BT1358 525 / ITU-R BT1700 NTSC
AVCOL_SPC_SMPTE240M = 7, ///< functionally identical to above
AVCOL_SPC_YCGCO = 8, ///< Used by Dirac / VC-2 and H.264 FRext, see ITU-T SG16
AVCOL_SPC_YCOCG = AVCOL_SPC_YCGCO,
AVCOL_SPC_BT2020_NCL = 9, ///< ITU-R BT2020 non-constant luminance system
AVCOL_SPC_BT2020_CL = 10, ///< ITU-R BT2020 constant luminance system
AVCOL_SPC_SMPTE2085 = 11, ///< SMPTE 2085, Y'D'zD'x
AVCOL_SPC_CHROMA_DERIVED_NCL = 12, ///< Chromaticity-derived non-constant luminance system
AVCOL_SPC_CHROMA_DERIVED_CL = 13, ///< Chromaticity-derived constant luminance system
AVCOL_SPC_ICTCP = 14, ///< ITU-R BT.2100-0, ICtCp
AVCOL_SPC_NB ///< Not part of ABI
};
-->
<param name="colorspace" value="0"/>
<!--
enum AVColorRange {
AVCOL_RANGE_UNSPECIFIED = 0,
AVCOL_RANGE_MPEG = 1, ///< the normal 219*2^(n-8) "MPEG" YUV ranges
AVCOL_RANGE_JPEG = 2, ///< the normal 2^n-1 "JPEG" YUV ranges
AVCOL_RANGE_NB ///< Not part of ABI
};
-->
<param name="color-range" value="2"/>
<!-- x264 private options-->
<options>
<option name="preset" value="veryfast"/>
<option name="intra_refresh" value="1"/>
<option name="tune" value="animation+zerolatency"/>
<option name="sc_threshold" value="40"/>
<option name="b_strategy" value="1"/>
<option name="crf" value="18"/>
</options>
</profile>
<profile name="H265">
</profile>
<profile name="conference">
<param name="dec-threads" value="1"/>
<param name="enc-threads" value="cpu/2/4"/>
<codecs>
<!-- profiles will be parsed at runtime
to overwrite this profile params if codec matches -->
<codec name="H263" profile="H263"/>
<codec name="H264" profile="H264"/>
<codec name="H264" profile="conference-H264"/>
</codecs>
</profile>
<profile name="conference-H264">
<options>
<option name="preset" value="veryfast"/>
<option name="intra_refresh" value="1"/>
<option name="tune" value="animation+zerolatency"/>
<option name="sc_threshold" value="40"/>
<option name="b_strategy" value="1"/>
<option name="crf" value="10"/>
</options>
</profile>
</profiles>
</configuration>
<configuration name="avformat.conf" description="AVFormat Config">
<settings>
<param name="colorspace" value="1"/>
</settings>
</configuration>

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<configuration name="avmd.conf" description="AVMD config">
<settings>
<!-- Edit these settings to change default behaviour
of each avmd session. Settings can be overwritten
by values passed dynamically per each session -->
<!-- Global settings -->
<!-- define/undefine this to enable/disable logging of avmd
intermediate computations to log -->
<param name="debug" value="0"/>
<!-- define/undef this to enable/disable verbose logging (and reporting to the console)
of detection status and other diagnostics like parameters avmd session has been started with,
change of configuration parameters, beep detection status after session ended
(stop event is fired independently of this setting and beep status included there) -->
<param name="report_status" value="1"/>
<!-- define/undefine this to enable/disable faster computation
of arcus cosine - table will be created mapping floats
to integers and returning arc cos values given these integer
indices into table -->
<param name="fast_math" value="0"/>
<!-- Global settings end -->
<!-- Per call (session) settings. These settings can be overwritten
with custom/different values per each avmd session -->
<!-- define/undefine this to classify avmd beep detection as valid
only when there is required number of consecutive elements
in the SMA buffer without reset -->
<param name="require_continuous_streak" value="1"/>
<!-- required number of consecutive elements in the SMA buffer
without reset. This parameter helps to avoid false beeps, bigger this value is
smaller the probability of getting false detection -->
<param name="sample_n_continuous_streak" value="3"/>
<!-- define number of samples to skip starting from the beginning
of the frame and/or after reset has happened. This serves the purpose of skipping first few
estimations on each frame, as these estimations may be inaccurate. This parameter also helps
to give more robust detections when it's value is increased (up to scertain limit of about 60). -->
<param name="sample_n_to_skip" value="0"/>
<param name="require_continuous_streak_amp" value="1"/>
<param name="sample_n_continuous_streak_amp" value="3"/>
<!-- define/undefine this to enable/disable simplified estimation
of frequency based on approximation of sin(x) with (x)
in the range x=[0,PI/2] -->
<param name="simplified_estimation" value="1"/>
<!-- define/undefine to enable/disable avmd on internal channel -->
<param name="inbound_channel" value="0"/>
<!-- define/undefine to enable/disable avmd on external channel -->
<param name="outbound_channel" value="1"/>
<!-- determines the mode of detection, default is both amplitude and frequency -->
<param name="detection_mode" value="2"/>
<!-- number of detection threads running per each avmd session -->
<param name="detectors_n" value="36"/>
<!-- number of lagged detection threads running per each avmd session -->
<param name="detectors_lagged_n" value="1"/>
<!-- Per call settings end -->
</settings>
</configuration>

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<configuration name="mod_blacklist.conf" description="Blacklist module">
<lists>
<!--
Example blacklist, the referenced file contains blacklisted items, one entry per line
NOTE: make sure the file exists and is readable by FreeSWITCH.
<list name="example" filename="$${conf_dir}/blacklists/example.list"/>
-->
</lists>
</configuration>

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<configuration name="callcenter.conf" description="CallCenter">
<settings>
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
<!--<param name="dbname" value="/dev/shm/callcenter.db"/>-->
<!--<param name="cc-instance-id" value="single_box"/>-->
</settings>
<queues>
<queue name="support@default">
<param name="strategy" value="longest-idle-agent"/>
<param name="moh-sound" value="$${hold_music}"/>
<!--<param name="record-template" value="$${recordings_dir}/${strftime(%Y-%m-%d-%H-%M-%S)}.${destination_number}.${caller_id_number}.${uuid}.wav"/>-->
<param name="time-base-score" value="system"/>
<param name="max-wait-time" value="0"/>
<param name="max-wait-time-with-no-agent" value="0"/>
<param name="max-wait-time-with-no-agent-time-reached" value="5"/>
<param name="tier-rules-apply" value="false"/>
<param name="tier-rule-wait-second" value="300"/>
<param name="tier-rule-wait-multiply-level" value="true"/>
<param name="tier-rule-no-agent-no-wait" value="false"/>
<param name="discard-abandoned-after" value="60"/>
<param name="abandoned-resume-allowed" value="false"/>
</queue>
</queues>
<!-- WARNING: Configuration of XML Agents will be updated into the DB upon restart. -->
<!-- WARNING: Configuration of XML Tiers will reset the level and position if those were supplied. -->
<!-- WARNING: Agents and Tiers XML config shouldn't be used in a multi FS shared DB setup (Not currently supported anyway) -->
<agents>
<!--<agent name="1000@default" type="callback" contact="[leg_timeout=10]user/1000@default" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />-->
</agents>
<tiers>
<!-- If no level or position is provided, they will default to 1. You should do this to keep db value on restart. -->
<!-- <tier agent="1000@default" queue="support@default" level="1" position="1"/> -->
</tiers>
</configuration>

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<configuration name="cdr_csv.conf" description="CDR CSV Format">
<settings>
<!-- 'cdr-csv' will always be appended to log-base -->
<!--<param name="log-base" value="/var/log"/>-->
<param name="default-template" value="example"/>
<!-- This is like the info app but after the call is hung up -->
<!--<param name="debug" value="true"/>-->
<param name="rotate-on-hup" value="true"/>
<!-- may be a b or ab -->
<param name="legs" value="a"/>
<!-- Only log in Master.csv -->
<!-- <param name="master-file-only" value="true"/> -->
</settings>
<templates>
<template name="sql">INSERT INTO cdr VALUES ("${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}");</template>
<template name="example">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}"</template>
<template name="snom">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}","${read_codec}","${write_codec}","${sip_user_agent}","${call_clientcode}","${sip_rtp_rxstat}","${sip_rtp_txstat}","${sofia_record_file}"</template>
<template name="linksys">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}","${sip_user_agent}","${sip_p_rtp_stat}"</template>
<template name="asterisk">"${accountcode}","${caller_id_number}","${destination_number}","${context}","${caller_id}","${channel_name}","${bridge_channel}","${last_app}","${last_arg}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${amaflags}","${uuid}","${userfield}"</template>
<template name="opencdrrate">"${uuid}","${signal_bond}","${direction}","${ani}","${destination_number}","${answer_stamp}","${end_stamp}","${billsec}","${accountcode}","${userfield}","${network_addr}","${regex('${original_caller_id_name}'|^.)}","${sip_gateway_name}"</template>
</templates>
</configuration>

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<configuration name="cdr_mongodb.conf" description="MongoDB CDR logger">
<settings>
<!-- Hostnames and IPv6 addrs not supported (yet) -->
<param name="host" value="127.0.0.1"/>
<param name="port" value="27017"/>
<!-- Namespace format is database.collection -->
<param name="namespace" value="test.cdr"/>
<!-- If true, create CDR for B-leg of call (default: true) -->
<param name="log-b-leg" value="false"/>
</settings>
</configuration>

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<configuration name="cdr_pg_csv.conf" description="CDR PG CSV Format">
<settings>
<!-- See parameters for PQconnectdb() at http://www.postgresql.org/docs/8.4/static/libpq-connect.html -->
<param name="db-info" value="host=localhost dbname=cdr connect_timeout=10" />
<!-- CDR table name -->
<!--<param name="db-table" value="cdr"/>-->
<!-- Log a-leg (a), b-leg (b) or both (ab) -->
<param name="legs" value="a"/>
<!-- Directory in which to spool failed SQL inserts -->
<!-- <param name="spool-dir" value="$${log_dir}/cdr-pg-csv"/> -->
<!-- Disk spool format if DB connection/insert fails - csv (default) or sql -->
<param name="spool-format" value="csv"/>
<param name="rotate-on-hup" value="true"/>
<!-- This is like the info app but after the call is hung up -->
<!--<param name="debug" value="true"/>-->
</settings>
<schema>
<field var="local_ip_v4"/>
<field var="caller_id_name"/>
<field var="caller_id_number"/>
<field var="destination_number"/>
<field var="context"/>
<field var="start_stamp"/>
<field var="answer_stamp"/>
<field var="end_stamp"/>
<field var="duration" quote="false"/>
<field var="billsec" quote="false"/>
<field var="hangup_cause"/>
<field var="uuid"/>
<field var="bleg_uuid"/>
<field var="accountcode"/>
<field var="read_codec"/>
<field var="write_codec"/>
<!-- <field var="sip_hangup_disposition"/> -->
<!-- <field var="ani"/> -->
</schema>
</configuration>

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<configuration name="cdr_sqlite.conf" description="SQLite CDR">
<settings>
<!-- SQLite database name (.db suffix will be automatically appended) -->
<!-- <param name="db-name" value="cdr"/> -->
<!-- CDR table name -->
<!-- <param name="db-table" value="cdr"/> -->
<!-- Log a-leg (a), b-leg (b) or both (ab) -->
<param name="legs" value="a"/>
<!-- Default template to use when inserting records -->
<param name="default-template" value="example"/>
<!-- This is like the info app but after the call is hung up -->
<!--<param name="debug" value="true"/>-->
</settings>
<templates>
<!-- Note that field order must match SQL table schema, otherwise insert will fail -->
<template name="example">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}",${duration},${billsec},"${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}"</template>
</templates>
</configuration>

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<configuration name="cepstral.conf" description="Cepstral TTS configuration">
<settings>
<!--
Possible encodings:
* utf-8
* us-ascii
* iso8859-1 (default)
* iso8859-15
-->
<param name="encoding" value="utf-8"/>
</settings>
</configuration>

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<configuration name="cidlookup.conf" description="cidlookup Configuration">
<settings>
<!-- comment out url to not setup a url based lookup -->
<param name="url" value="http://query.voipcnam.com/query.php?api_key=MYAPIKEY&amp;number=${caller_id_number}"/>
<!-- comment out whitepages-apikey to not use whitepages.com, you must
get an API key from http://developer.whitepages.com/ -->
<param name="whitepages-apikey" value="MYAPIKEY"/>
<!-- set to false to not cache (in memcache) results from the url query -->
<param name="cache" value="true"/>
<!-- expire is in seconds -->
<param name="cache-expire" value="86400"/>
<param name="odbc-dsn" value="phone:phone:phone"/>
<!-- comment out sql to not setup a database (directory) lookup -->
<param name="sql" value="
SELECT name||' ('||type||')' AS name
FROM phonebook p JOIN numbers n ON p.id = n.phonebook_id
WHERE n.number='${caller_id_number}'
LIMIT 1
"/>
<!-- comment out citystate-sql to not setup a database (city/state)
lookup -->
<param name="citystate-sql" value="
SELECT ratecenter||' '||state as name
FROM npa_nxx_company_ocn
WHERE npa = ${caller_id_number:1:3} AND nxx = ${caller_id_number:4:3}
LIMIT 1
"/>
</settings>
</configuration>

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<!-- http://wiki.freeswitch.org/wiki/Mod_conference -->
<!-- None of these paths are real if you want any of these options you need to really set them up -->
<configuration name="conference.conf" description="Audio Conference">
<!-- Advertise certain presence on startup . -->
<advertise>
<room name="3001@$${domain}" status="FreeSWITCH"/>
</advertise>
<!-- These are the default keys that map when you do not specify a caller control group -->
<!-- Note: none and default are reserved names for group names. Disabled if dist-dtmf member flag is set. -->
<caller-controls>
<group name="default">
<control action="mute" digits="0"/>
<control action="deaf mute" digits="*"/>
<control action="energy up" digits="9"/>
<control action="energy equ" digits="8"/>
<control action="energy dn" digits="7"/>
<control action="vol talk up" digits="3"/>
<control action="vol talk zero" digits="2"/>
<control action="vol talk dn" digits="1"/>
<control action="vol listen up" digits="6"/>
<control action="vol listen zero" digits="5"/>
<control action="vol listen dn" digits="4"/>
<control action="hangup" digits="#"/>
</group>
</caller-controls>
<!-- Profiles are collections of settings you can reference by name. -->
<profiles>
<!--If no profile is specified it will default to "default"-->
<profile name="default">
<!-- Directory to drop CDR's
'auto' means $PREFIX/logs/conference_cdr/<confernece_uuid>.cdr.xml
a non-absolute path means $PREFIX/logs/<value>/<confernece_uuid>.cdr.xml
absolute path means <value>/<confernece_uuid>.cdr.xml
-->
<!-- <param name="cdr-log-dir" value="auto"/> -->
<!-- Domain (for presence) -->
<param name="domain" value="$${domain}"/>
<!-- Sample Rate-->
<param name="rate" value="8000"/>
<!-- Number of milliseconds per frame -->
<param name="interval" value="20"/>
<!-- Energy level required for audio to be sent to the other users -->
<param name="energy-level" value="100"/>
<!--Can be | delim of waste|mute|deaf|dist-dtmf waste will always transmit data to each channel
even during silence. dist-dtmf propagates dtmfs to all other members, but channel controls
via dtmf will be disabled. -->
<!-- <param name="member-flags" value="waste"/> -->
<!-- Name of the caller control group to use for this profile -->
<!-- <param name="caller-controls" value="some name"/> -->
<!-- Name of the caller control group to use for the moderator in this profile -->
<!-- <param name="moderator-controls" value="some name"/> -->
<!-- TTS Engine to use -->
<!-- <param name="tts-engine" value="cepstral"/> -->
<!-- TTS Voice to use -->
<!-- <param name="tts-voice" value="david"/> -->
<!-- If TTS is enabled all audio-file params beginning with -->
<!-- 'say:' will be considered text to say with TTS -->
<!-- Override the default path here, after which you use relative paths in the other sound params -->
<!-- Note: The default path is the conference's first caller's sound_prefix -->
<!-- <param name="sound-prefix" value="$${sound_prefix}"/> -->
<!-- File to play to acknowledge succees -->
<!-- <param name="ack-sound" value="beep.wav"/> -->
<!-- File to play to acknowledge failure -->
<!-- <param name="nack-sound" value="beeperr.wav"/> -->
<!-- File to play to acknowledge muted -->
<param name="muted-sound" value="conference/conf-muted.wav"/>
<!-- File to play to acknowledge unmuted -->
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<!-- File to play if you are alone in the conference -->
<param name="alone-sound" value="conference/conf-alone.wav"/>
<!-- File to play endlessly (nobody will ever be able to talk) -->
<!-- <param name="perpetual-sound" value="perpetual.wav"/> -->
<!-- File to play when you're alone (music on hold)-->
<param name="moh-sound" value="$${hold_music}"/>
<!-- File to play when you join the conference -->
<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
<!-- File to play when you leave the conference -->
<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
<!-- File to play when you are ejected from the conference -->
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<!-- File to play when the conference is locked -->
<param name="locked-sound" value="conference/conf-locked.wav"/>
<!-- File to play when the conference is locked during the call-->
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<!-- File to play when the conference is unlocked during the call-->
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<!-- File to play to prompt for a pin -->
<param name="pin-sound" value="conference/conf-pin.wav"/>
<!-- File to play to when the pin is invalid -->
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<!-- Conference pin -->
<!-- <param name="pin" value="12345"/> -->
<!-- <param name="moderator-pin" value="54321"/> -->
<!-- Max number of times the user can be prompted for PIN -->
<!-- <param name="pin-retries" value="3"/> -->
<!-- Default Caller ID Name for outbound calls -->
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<!-- Default Caller ID Number for outbound calls -->
<param name="caller-id-number" value="$${outbound_caller_id}"/>
<!-- Suppress start and stop talking events -->
<!-- <param name="suppress-events" value="start-talking,stop-talking"/> -->
<!-- enable comfort noise generation -->
<param name="comfort-noise" value="true"/>
<!-- Uncomment auto-record to toggle recording every conference call. -->
<!-- Another valid value is shout://user:pass@server.com/live.mp3 -->
<!--
<param name="auto-record" value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
-->
<!-- IVR digit machine timeouts -->
<!-- How much to wait between DTMF digits to match caller-controls -->
<!-- <param name="ivr-dtmf-timeout" value="500"/> -->
<!-- How much to wait for the first DTMF, 0 forever -->
<!-- <param name="ivr-input-timeout" value="0" /> -->
<!-- Delay before a conference is asked to be terminated -->
<!-- <param name="endconf-grace-time" value="120" /> -->
<!-- Can be | delim of wait-mod|audio-always|video-bridge|video-floor-only
wait_mod will wait until the moderator in,
audio-always will always mix audio from all members regardless they are talking or not -->
<!-- <param name="conference-flags" value="audio-always"/> -->
<!-- Allow live array sync for Verto -->
<!-- <param name="conference-flags" value="livearray-sync"/> -->
</profile>
<profile name="wideband">
<param name="domain" value="$${domain}"/>
<param name="rate" value="16000"/>
<param name="interval" value="20"/>
<param name="energy-level" value="100"/>
<!-- <param name="sound-prefix" value="$${sound_prefix}"/> -->
<param name="muted-sound" value="conference/conf-muted.wav"/>
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<param name="alone-sound" value="conference/conf-alone.wav"/>
<param name="moh-sound" value="$${hold_music}"/>
<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<param name="locked-sound" value="conference/conf-locked.wav"/>
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<param name="pin-sound" value="conference/conf-pin.wav"/>
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<param name="caller-id-number" value="$${outbound_caller_id}"/>
<param name="comfort-noise" value="true"/>
<!-- <param name="tts-engine" value="flite"/> -->
<!-- <param name="tts-voice" value="kal16"/> -->
</profile>
<profile name="ultrawideband">
<param name="domain" value="$${domain}"/>
<param name="rate" value="32000"/>
<param name="interval" value="20"/>
<param name="energy-level" value="100"/>
<!-- <param name="sound-prefix" value="$${sound_prefix}"/> -->
<param name="muted-sound" value="conference/conf-muted.wav"/>
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<param name="alone-sound" value="conference/conf-alone.wav"/>
<param name="moh-sound" value="$${hold_music}"/>
<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<param name="locked-sound" value="conference/conf-locked.wav"/>
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<param name="pin-sound" value="conference/conf-pin.wav"/>
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<param name="caller-id-number" value="$${outbound_caller_id}"/>
<param name="comfort-noise" value="true"/>
<!-- <param name="conference-flags" value="video-floor-only|rfc-4579|livearray-sync|auto-3d-position|transcode-video|minimize-video-encoding"/> -->
<!-- <param name="video-mode" value="mux"/> -->
<!-- <param name="video-layout-name" value="3x3"/> -->
<!-- <param name="video-layout-name" value="group:grid"/> -->
<!-- <param name="video-canvas-size" value="1280x720"/> -->
<!-- <param name="video-canvas-bgcolor" value="#333333"/> -->
<!-- <param name="video-layout-bgcolor" value="#000000"/> -->
<!-- <param name="video-codec-bandwidth" value="2mb"/> -->
<!-- <param name="video-fps" value="15"/> -->
<!-- <param name="video-auto-floor-msec" value="100"/> -->
<!-- <param name="tts-engine" value="flite"/> -->
<!-- <param name="tts-voice" value="kal16"/> -->
</profile>
<profile name="cdquality">
<param name="domain" value="$${domain}"/>
<param name="rate" value="48000"/>
<param name="interval" value="20"/>
<param name="energy-level" value="100"/>
<!-- <param name="sound-prefix" value="$${sound_prefix}"/> -->
<param name="muted-sound" value="conference/conf-muted.wav"/>
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<param name="alone-sound" value="conference/conf-alone.wav"/>
<param name="moh-sound" value="$${hold_music}"/>
<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<param name="locked-sound" value="conference/conf-locked.wav"/>
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<param name="pin-sound" value="conference/conf-pin.wav"/>
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<param name="caller-id-number" value="$${outbound_caller_id}"/>
<param name="comfort-noise" value="true"/>
<!-- <param name="conference-flags" value="video-floor-only|rfc-4579|livearray-sync|auto-3d-position|minimize-video-encoding"/> -->
<!-- <param name="video-mode" value="mux"/> -->
<!-- <param name="video-layout-name" value="3x3"/> -->
<!-- <param name="video-layout-name" value="group:grid"/> -->
<!-- <param name="video-canvas-size" value="1920x1080"/> -->
<!-- <param name="video-canvas-bgcolor" value="#333333"/> -->
<!-- <param name="video-layout-bgcolor" value="#000000"/> -->
<!-- <param name="video-codec-bandwidth" value="2mb"/> -->
<!-- <param name="video-fps" value="15"/> -->
</profile>
<profile name="video-mcu-stereo">
<param name="domain" value="$${domain}"/>
<param name="rate" value="48000"/>
<param name="channels" value="2"/>
<param name="interval" value="20"/>
<param name="energy-level" value="200"/>
<!-- <param name="tts-engine" value="flite"/> -->
<!-- <param name="tts-voice" value="kal16"/> -->
<param name="muted-sound" value="conference/conf-muted.wav"/>
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<param name="alone-sound" value="conference/conf-alone.wav"/>
<param name="moh-sound" value="$${hold_music}"/>
<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<param name="locked-sound" value="conference/conf-locked.wav"/>
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<param name="pin-sound" value="conference/conf-pin.wav"/>
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<param name="caller-id-number" value="$${outbound_caller_id}"/>
<param name="comfort-noise" value="false"/>
<param name="conference-flags" value="livearray-json-status|json-events|video-floor-only|rfc-4579|livearray-sync|minimize-video-encoding|manage-inbound-video-bitrate|video-required-for-canvas|video-mute-exit-canvas|mute-detect"/>
<param name="video-auto-floor-msec" value="1000"/>
<param name="video-mode" value="mux"/>
<param name="video-layout-name" value="3x3"/>
<param name="video-layout-name" value="group:grid"/>
<param name="video-canvas-size" value="1920x1080"/>
<param name="video-canvas-bgcolor" value="#333333"/>
<param name="video-layout-bgcolor" value="#000000"/>
<param name="video-codec-bandwidth" value="3mb"/>
<param name="video-fps" value="30"/>
<!-- <param name="video-codec-config-profile-name" value="conference"/> -->
</profile>
<profile name="video-mcu-stereo-720">
<param name="domain" value="$${domain}"/>
<param name="rate" value="48000"/>
<param name="channels" value="2"/>
<param name="interval" value="20"/>
<param name="energy-level" value="200"/>
<!-- <param name="tts-engine" value="flite"/> -->
<!-- <param name="tts-voice" value="kal16"/> -->
<param name="muted-sound" value="conference/conf-muted.wav"/>
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<param name="alone-sound" value="conference/conf-alone.wav"/>
<param name="moh-sound" value="$${hold_music}"/>
<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<param name="locked-sound" value="conference/conf-locked.wav"/>
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<param name="pin-sound" value="conference/conf-pin.wav"/>
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<param name="caller-id-number" value="$${outbound_caller_id}"/>
<param name="comfort-noise" value="false"/>
<param name="conference-flags" value="livearray-json-status|json-events|video-floor-only|rfc-4579|livearray-sync|minimize-video-encoding|manage-inbound-video-bitrate|video-required-for-canvas|video-mute-exit-canvas|mute-detect"/>
<param name="video-auto-floor-msec" value="1000"/>
<param name="video-mode" value="mux"/>
<param name="video-layout-name" value="3x3"/>
<param name="video-layout-name" value="group:grid"/>
<param name="video-canvas-size" value="1280x720"/>
<param name="video-canvas-bgcolor" value="#333333"/>
<param name="video-layout-bgcolor" value="#000000"/>
<param name="video-codec-bandwidth" value="3mb"/>
<param name="video-fps" value="30"/>
</profile>
<profile name="video-mcu-stereo-480">
<param name="domain" value="$${domain}"/>
<param name="rate" value="48000"/>
<param name="channels" value="2"/>
<param name="interval" value="20"/>
<param name="energy-level" value="200"/>
<!-- <param name="tts-engine" value="flite"/> -->
<!-- <param name="tts-voice" value="kal16"/> -->
<param name="muted-sound" value="conference/conf-muted.wav"/>
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<param name="alone-sound" value="conference/conf-alone.wav"/>
<param name="moh-sound" value="$${hold_music}"/>
<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<param name="locked-sound" value="conference/conf-locked.wav"/>
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<param name="pin-sound" value="conference/conf-pin.wav"/>
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<param name="caller-id-number" value="$${outbound_caller_id}"/>
<param name="comfort-noise" value="false"/>
<param name="conference-flags" value="livearray-json-status|json-events|video-floor-only|rfc-4579|livearray-sync|minimize-video-encoding|manage-inbound-video-bitrate|video-required-for-canvas|video-mute-exit-canvas|mute-detect"/>
<param name="video-auto-floor-msec" value="1000"/>
<param name="video-mode" value="mux"/>
<param name="video-layout-name" value="3x3"/>
<param name="video-layout-name" value="group:grid"/>
<param name="video-canvas-size" value="640x480"/>
<param name="video-canvas-bgcolor" value="#333333"/>
<param name="video-layout-bgcolor" value="#000000"/>
<param name="video-codec-bandwidth" value="3mb"/>
<param name="video-fps" value="30"/>
</profile>
<profile name="video-mcu-stereo-320">
<param name="domain" value="$${domain}"/>
<param name="rate" value="48000"/>
<param name="channels" value="2"/>
<param name="interval" value="20"/>
<param name="energy-level" value="200"/>
<!-- <param name="tts-engine" value="flite"/> -->
<!-- <param name="tts-voice" value="kal16"/> -->
<param name="muted-sound" value="conference/conf-muted.wav"/>
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<param name="alone-sound" value="conference/conf-alone.wav"/>
<param name="moh-sound" value="$${hold_music}"/>
<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<param name="locked-sound" value="conference/conf-locked.wav"/>
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<param name="pin-sound" value="conference/conf-pin.wav"/>
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<param name="caller-id-number" value="$${outbound_caller_id}"/>
<param name="comfort-noise" value="false"/>
<param name="conference-flags" value="livearray-json-status|json-events|video-floor-only|rfc-4579|livearray-sync|minimize-video-encoding|manage-inbound-video-bitrate|video-required-for-canvas|video-mute-exit-canvas|mute-detect"/>
<param name="video-auto-floor-msec" value="1000"/>
<param name="video-mode" value="mux"/>
<param name="video-layout-name" value="3x3"/>
<param name="video-layout-name" value="group:grid"/>
<param name="video-canvas-size" value="480x320"/>
<param name="video-canvas-bgcolor" value="#333333"/>
<param name="video-layout-bgcolor" value="#000000"/>
<param name="video-codec-bandwidth" value="3mb"/>
<param name="video-fps" value="30"/>
</profile>
<profile name="sla">
<param name="domain" value="$${domain}"/>
<param name="rate" value="16000"/>
<param name="interval" value="20"/>
<param name="caller-controls" value="none"/>
<param name="energy-level" value="200"/>
<param name="moh-sound" value="silence"/>
<param name="comfort-noise" value="true"/>
</profile>
</profiles>
</configuration>

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@ -0,0 +1,393 @@
<configuration name="conference_layouts.conf" description="Audio Conference">
<layout-settings>
<layouts>
<layout name="1x1">
<image x="0" y="0" scale="360" floor="true"/>
</layout>
<layout name="1x2" auto-3d-position="true">
<image x="90" y="0" scale="180"/>
<image x="90" y="180" scale="180"/>
</layout>
<layout name="2x1" auto-3d-position="true">
<image x="0" y="90" scale="180"/>
<image x="180" y="90" scale="180"/>
</layout>
<layout name="2x1-zoom" auto-3d-position="true">
<image x="0" y="0" scale="180" hscale="360" zoom="true"/>
<image x="180" y="0" scale="180" hscale="360" zoom="true"/>
</layout>
<layout name="3x1-zoom" auto-3d-position="true">
<image x="0" y="0" scale="120" hscale="360" zoom="true"/>
<image x="120" y="0" scale="120" hscale="360" zoom="true"/>
<image x="240" y="0" scale="120" hscale="360" zoom="true"/>
</layout>
<layout name="5-grid-zoom" auto-3d-position="true">
<image x="0" y="0" scale="180"/>
<image x="180" y="0" scale="180"/>
<image x="0" y="180" scale="120" hscale="180" zoom="true"/>
<image x="120" y="180" scale="120" hscale="180" zoom="true"/>
<image x="240" y="180" scale="120" hscale="180" zoom="true"/>
</layout>
<layout name="3x2-zoom" auto-3d-position="true">
<image x="0" y="0" scale="120" hscale="180" zoom="true"/>
<image x="120" y="0" scale="120" hscale="180" zoom="true"/>
<image x="240" y="0" scale="120" hscale="180" zoom="true"/>
<image x="0" y="180" scale="120" hscale="180" zoom="true"/>
<image x="120" y="180" scale="120" hscale="180" zoom="true"/>
<image x="240" y="180" scale="120" hscale="180" zoom="true"/>
</layout>
<layout name="7-grid-zoom" auto-3d-position="true">
<image x="0" y="0" scale="120" hscale="180" zoom="true"/>
<image x="120" y="0" scale="120" hscale="180" zoom="true"/>
<image x="240" y="0" scale="120" hscale="180" zoom="true"/>
<image x="0" y="180" scale="90" hscale="180" zoom="true"/>
<image x="90" y="180" scale="90" hscale="180" zoom="true"/>
<image x="180" y="180" scale="90" hscale="180" zoom="true"/>
<image x="270" y="180" scale="90" hscale="180" zoom="true"/>
</layout>
<layout name="4x2-zoom" auto-3d-position="true">
<image x="0" y="0" scale="90" hscale="180" zoom="true"/>
<image x="90" y="0" scale="90" hscale="180" zoom="true"/>
<image x="180" y="0" scale="90" hscale="180" zoom="true"/>
<image x="270" y="0" scale="90" hscale="180" zoom="true"/>
<image x="0" y="180" scale="90" hscale="180" zoom="true"/>
<image x="90" y="180" scale="90" hscale="180" zoom="true"/>
<image x="180" y="180" scale="90" hscale="180" zoom="true"/>
<image x="270" y="180" scale="90" hscale="180" zoom="true"/>
</layout>
<layout name="1x1+2x1" auto-3d-position="true">
<image x="90" y="0" scale="180"/>
<image x="0" y="180" scale="180"/>
<image x="180" y="180" scale="180"/>
</layout>
<layout name="2x2" auto-3d-position="true">
<image x="0" y="0" scale="180"/>
<image x="180" y="0" scale="180"/>
<image x="0" y="180" scale="180"/>
<image x="180" y="180" scale="180"/>
</layout>
<layout name="3x3" auto-3d-position="true">
<image x="0" y="0" scale="120"/>
<image x="120" y="0" scale="120"/>
<image x="240" y="0" scale="120"/>
<image x="0" y="120" scale="120"/>
<image x="120" y="120" scale="120"/>
<image x="240" y="120" scale="120"/>
<image x="0" y="240" scale="120"/>
<image x="120" y="240" scale="120"/>
<image x="240" y="240" scale="120"/>
</layout>
<layout name="4x4" auto-3d-position="true">
<image x="0" y="0" scale="90"/>
<image x="90" y="0" scale="90"/>
<image x="180" y="0" scale="90"/>
<image x="270" y="0" scale="90"/>
<image x="0" y="90" scale="90"/>
<image x="90" y="90" scale="90"/>
<image x="180" y="90" scale="90"/>
<image x="270" y="90" scale="90"/>
<image x="0" y="180" scale="90"/>
<image x="90" y="180" scale="90"/>
<image x="180" y="180" scale="90"/>
<image x="270" y="180" scale="90"/>
<image x="0" y="270" scale="90"/>
<image x="90" y="270" scale="90"/>
<image x="180" y="270" scale="90"/>
<image x="270" y="270" scale="90"/>
</layout>
<layout name="5x5" auto-3d-position="true">
<image x="0" y="0" scale="72"/>
<image x="72" y="0" scale="72"/>
<image x="144" y="0" scale="72"/>
<image x="216" y="0" scale="72"/>
<image x="288" y="0" scale="72"/>
<image x="0" y="72" scale="72"/>
<image x="72" y="72" scale="72"/>
<image x="144" y="72" scale="72"/>
<image x="216" y="72" scale="72"/>
<image x="288" y="72" scale="72"/>
<image x="0" y="144" scale="72"/>
<image x="72" y="144" scale="72"/>
<image x="144" y="144" scale="72"/>
<image x="216" y="144" scale="72"/>
<image x="288" y="144" scale="72"/>
<image x="0" y="216" scale="72"/>
<image x="72" y="216" scale="72"/>
<image x="144" y="216" scale="72"/>
<image x="216" y="216" scale="72"/>
<image x="288" y="216" scale="72"/>
<image x="0" y="288" scale="72"/>
<image x="72" y="288" scale="72"/>
<image x="144" y="288" scale="72"/>
<image x="216" y="288" scale="72"/>
<image x="288" y="288" scale="72"/>
</layout>
<layout name="6x6" auto-3d-position="true">
<image x="0" y="0" scale="60"/>
<image x="60" y="0" scale="60"/>
<image x="120" y="0" scale="60"/>
<image x="180" y="0" scale="60"/>
<image x="240" y="0" scale="60"/>
<image x="300" y="0" scale="60"/>
<image x="0" y="60" scale="60"/>
<image x="60" y="60" scale="60"/>
<image x="120" y="60" scale="60"/>
<image x="180" y="60" scale="60"/>
<image x="240" y="60" scale="60"/>
<image x="300" y="60" scale="60"/>
<image x="0" y="120" scale="60"/>
<image x="60" y="120" scale="60"/>
<image x="120" y="120" scale="60"/>
<image x="180" y="120" scale="60"/>
<image x="240" y="120" scale="60"/>
<image x="300" y="120" scale="60"/>
<image x="0" y="180" scale="60"/>
<image x="60" y="180" scale="60"/>
<image x="120" y="180" scale="60"/>
<image x="180" y="180" scale="60"/>
<image x="240" y="180" scale="60"/>
<image x="300" y="180" scale="60"/>
<image x="0" y="240" scale="60"/>
<image x="60" y="240" scale="60"/>
<image x="120" y="240" scale="60"/>
<image x="180" y="240" scale="60"/>
<image x="240" y="240" scale="60"/>
<image x="300" y="240" scale="60"/>
<image x="0" y="300" scale="60"/>
<image x="60" y="300" scale="60"/>
<image x="120" y="300" scale="60"/>
<image x="180" y="300" scale="60"/>
<image x="240" y="300" scale="60"/>
<image x="300" y="300" scale="60"/>
</layout>
<layout name="8x8" auto-3d-position="true">
<image x="0" y="0" scale="45"/>
<image x="45" y="0" scale="45"/>
<image x="90" y="0" scale="45"/>
<image x="135" y="0" scale="45"/>
<image x="180" y="0" scale="45"/>
<image x="225" y="0" scale="45"/>
<image x="270" y="0" scale="45"/>
<image x="315" y="0" scale="45"/>
<image x="0" y="45" scale="45"/>
<image x="45" y="45" scale="45"/>
<image x="90" y="45" scale="45"/>
<image x="135" y="45" scale="45"/>
<image x="180" y="45" scale="45"/>
<image x="225" y="45" scale="45"/>
<image x="270" y="45" scale="45"/>
<image x="315" y="45" scale="45"/>
<image x="0" y="90" scale="45"/>
<image x="45" y="90" scale="45"/>
<image x="90" y="90" scale="45"/>
<image x="135" y="90" scale="45"/>
<image x="180" y="90" scale="45"/>
<image x="225" y="90" scale="45"/>
<image x="270" y="90" scale="45"/>
<image x="315" y="90" scale="45"/>
<image x="0" y="135" scale="45"/>
<image x="45" y="135" scale="45"/>
<image x="90" y="135" scale="45"/>
<image x="135" y="135" scale="45"/>
<image x="180" y="135" scale="45"/>
<image x="225" y="135" scale="45"/>
<image x="270" y="135" scale="45"/>
<image x="315" y="135" scale="45"/>
<image x="0" y="180" scale="45"/>
<image x="45" y="180" scale="45"/>
<image x="90" y="180" scale="45"/>
<image x="135" y="180" scale="45"/>
<image x="180" y="180" scale="45"/>
<image x="225" y="180" scale="45"/>
<image x="270" y="180" scale="45"/>
<image x="315" y="180" scale="45"/>
<image x="0" y="225" scale="45"/>
<image x="45" y="225" scale="45"/>
<image x="90" y="225" scale="45"/>
<image x="135" y="225" scale="45"/>
<image x="180" y="225" scale="45"/>
<image x="225" y="225" scale="45"/>
<image x="270" y="225" scale="45"/>
<image x="315" y="225" scale="45"/>
<image x="0" y="270" scale="45"/>
<image x="45" y="270" scale="45"/>
<image x="90" y="270" scale="45"/>
<image x="135" y="270" scale="45"/>
<image x="180" y="270" scale="45"/>
<image x="225" y="270" scale="45"/>
<image x="270" y="270" scale="45"/>
<image x="315" y="270" scale="45"/>
<image x="0" y="315" scale="45"/>
<image x="45" y="315" scale="45"/>
<image x="90" y="315" scale="45"/>
<image x="135" y="315" scale="45"/>
<image x="180" y="315" scale="45"/>
<image x="225" y="315" scale="45"/>
<image x="270" y="315" scale="45"/>
<image x="315" y="315" scale="45"/>
</layout>
<layout name="1up_top_left+5" auto-3d-position="true">
<image x="0" y="0" scale="240" floor="true"/>
<image x="240" y="0" scale="120"/>
<image x="240" y="120" scale="120"/>
<image x="0" y="240" scale="120"/>
<image x="120" y="240" scale="120"/>
<image x="240" y="240" scale="120"/>
</layout>
<layout name="1up_top_left+7" auto-3d-position="true">
<image x="0" y="0" scale="270" floor="true"/>
<image x="270" y="0" scale="90"/>
<image x="270" y="90" scale="90"/>
<image x="270" y="180" scale="90"/>
<image x="0" y="270" scale="90"/>
<image x="90" y="270" scale="90"/>
<image x="180" y="270" scale="90"/>
<image x="270" y="270" scale="90"/>
</layout>
<layout name="1up_top_left+9" auto-3d-position="true">
<image x="0" y="0" scale="288" floor="true"/>
<image x="288" y="0" scale="72"/>
<image x="288" y="72" scale="72"/>
<image x="288" y="144" scale="72"/>
<image x="288" y="216" scale="72"/>
<image x="0" y="288" scale="72"/>
<image x="72" y="288" scale="72"/>
<image x="144" y="288" scale="72"/>
<image x="216" y="288" scale="72"/>
<image x="288" y="288" scale="72"/>
</layout>
<layout name="2up_top+8" auto-3d-position="true">
<image x="0" y="0" scale="180" floor="true"/>
<image x="180" y="0" scale="180" reservation_id="secondary"/>
<image x="0" y="180" scale="90"/>
<image x="90" y="180" scale="90"/>
<image x="180" y="180" scale="90"/>
<image x="270" y="180" scale="90"/>
<image x="0" y="270" scale="90"/>
<image x="90" y="270" scale="90"/>
<image x="180" y="270" scale="90"/>
<image x="270" y="270" scale="90"/>
</layout>
<layout name="2up_middle+8" auto-3d-position="true">
<image x="0" y="90" scale="180" floor="true"/>
<image x="180" y="90" scale="180" reservation_id="secondary"/>
<image x="0" y="0" scale="90"/>
<image x="90" y="0" scale="90"/>
<image x="180" y="0" scale="90"/>
<image x="270" y="0" scale="90"/>
<image x="0" y="270" scale="90"/>
<image x="90" y="270" scale="90"/>
<image x="180" y="270" scale="90"/>
<image x="270" y="270" scale="90"/>
</layout>
<layout name="2up_bottom+8" auto-3d-position="true">
<image x="0" y="180" scale="180" floor="true"/>
<image x="180" y="180" scale="180" reservation_id="secondary"/>
<image x="0" y="0" scale="90"/>
<image x="90" y="0" scale="90"/>
<image x="180" y="0" scale="90"/>
<image x="270" y="0" scale="90"/>
<image x="0" y="90" scale="90"/>
<image x="90" y="90" scale="90"/>
<image x="180" y="90" scale="90"/>
<image x="270" y="90" scale="90"/>
</layout>
<layout name="3up+4" auto-3d-position="true">
<image x="0" y="0" scale="180" floor="true"/>
<image x="180" y="0" scale="180" reservation_id="secondary"/>
<image x="0" y="180" scale="180" reservation_id="third"/>
<image x="180" y="180" scale="90"/>
<image x="270" y="180" scale="90"/>
<image x="180" y="270" scale="90"/>
<image x="270" y="270" scale="90"/>
</layout>
<layout name="3up+9" auto-3d-position="true">
<image x="0" y="0" scale="180" floor="true"/>
<image x="180" y="0" scale="180" reservation_id="secondary"/>
<image x="0" y="180" scale="180" reservation_id="third"/>
<image x="180" y="180" scale="60"/>
<image x="240" y="180" scale="60"/>
<image x="300" y="180" scale="60"/>
<image x="180" y="240" scale="60"/>
<image x="240" y="240" scale="60"/>
<image x="300" y="240" scale="60"/>
<image x="180" y="300" scale="60"/>
<image x="240" y="300" scale="60"/>
<image x="300" y="300" scale="60"/>
</layout>
<layout name="2x1-presenter-zoom" auto-3d-position="true">
<image x="0" y="0" scale="180" hscale="360" zoom="true" floor="true"/>
<image x="180" y="0" scale="180" hscale="360" zoom="true" reservation_id="presenter"/>
</layout>
<layout name="presenter-dual-vertical">
<image x="90" y="0" scale="180" floor-only="true"/>
<image x="90" y="180" scale="180" reservation_id="presenter"/>
</layout>
<layout name="presenter-dual-horizontal">
<image x="0" y="90" scale="180" floor-only="true"/>
<image x="180" y="90" scale="180" reservation_id="presenter"/>
</layout>
<layout name="presenter-overlap-small-top-right">
<image x="0" y="0" scale="360" floor-only="true"/>
<image x="300" y="0" scale="60" overlap="true" reservation_id="presenter"/>
</layout>
<layout name="presenter-overlap-small-bot-right">
<image x="0" y="0" scale="360" floor-only="true"/>
<image x="300" y="300" scale="60" overlap="true" reservation_id="presenter"/>
</layout>
<layout name="presenter-overlap-large-top-right">
<image x="0" y="0" scale="360" floor-only="true"/>
<image x="180" y="0" scale="180" overlap="true" reservation_id="presenter"/>
</layout>
<layout name="presenter-overlap-large-bot-right">
<image x="0" y="0" scale="360" floor-only="true"/>
<image x="180" y="180" scale="180" overlap="true" reservation_id="presenter"/>
</layout>
<layout name="overlaps" auto-3d-position="true">
<image x="0" y="0" scale="360" floor-only="true"/>
<image x="300" y="300" scale="60" overlap="true"/>
<image x="240" y="300" scale="60" overlap="true"/>
<image x="180" y="300" scale="60" overlap="true"/>
<image x="120" y="300" scale="60" overlap="true"/>
<image x="60" y="300" scale="60" overlap="true"/>
<image x="0" y="300" scale="60" overlap="true"/>
</layout>
</layouts>
<groups>
<group name="grid">
<layout>1x1</layout>
<layout>2x1</layout>
<layout>1x1+2x1</layout>
<layout>2x2</layout>
<layout>3x3</layout>
<layout>4x4</layout>
<layout>5x5</layout>
<layout>6x6</layout>
<layout>8x8</layout>
</group>
<group name="grid-zoom">
<layout>1x1</layout>
<layout>2x1-zoom</layout>
<layout>3x1-zoom</layout>
<layout>2x2</layout>
<layout>5-grid-zoom</layout>
<layout>3x2-zoom</layout>
<layout>7-grid-zoom</layout>
<layout>4x2-zoom</layout>
<layout>3x3</layout>
</group>
<group name="1up_top_left_plus">
<layout>1up_top_left+5</layout>
<layout>1up_top_left+7</layout>
<layout>1up_top_left+9</layout>
</group>
<group name="3up_plus">
<layout>3up+4</layout>
<layout>3up+9</layout>
</group>
</groups>
</layout-settings>
</configuration>

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<configuration name="console.conf" description="Console Logger">
<!-- pick a file name, a function name or 'all' -->
<!-- map as many as you need for specific debugging -->
<mappings>
<!--
name can be a file name, function name or 'all'
value is one or more of debug,info,notice,warning,err,crit,alert,all
See examples below
The following map is the default, which is all debug levels enabled:
<map name="all" value="debug,info,notice,warning,err,crit,alert"/>
Example: the following turns on debugging for error and critical levels only
<map name="all" value="err,crit"/>
NOTE: using map name="all" will override any other settings! If you
want a more specific set of console messages then you will need
to specify which files and/or functions you want to have debug
messages. One option is to turn on just the more critical
messages with map name="all", then specify the other types of
console messages you want to see for various files and functions.
Example: turn on ERROR, CRIT, ALERT for all modules, then specify other
levels for various modules and functions
<map name="all" value="err,crit,alert"/>
<map name="switch_loadable_module_process" value="all"/>
<map name="mod_local_stream.c" value="warning,debug"/>
<map name="mod_sndfile.c" value="warning,info,debug"/>
-->
<map name="all" value="console,debug,info,notice,warning,err,crit,alert"/>
<!--
You can use or modify this sample set of mappings. It turns on higher
level messages for all modules and then specifies extra lower level
messages for freetdm, Sofia, and switch core messages.
<map name="all" value="warning,err,crit,alert"/>
<map name="zap_analog.c" value="all"/>
<map name="zap_io.c" value="all"/>
<map name="zap_isdn.c" value="all"/>
<map name="zap_zt.c" value="all"/>
<map name="mod_freetdm" value="all"/>
<map name="sofia.c" value="notice"/>
<map name="switch_core_state_machine.c" value="all"/>
-->
</mappings>
<settings>
<!-- comment or set to false for no color logging -->
<param name="colorize" value="true"/>
<param name="loglevel" value="$${console_loglevel}"/>
</settings>
</configuration>

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<configuration name="curl.conf" description="cURL module">
<settings>
<param name="max-bytes" value="64000"/>
</settings>
</configuration>

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<configuration name="db.conf" description="LIMIT DB Configuration">
<settings>
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
</settings>
</configuration>

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<configuration name="dialplan_directory.conf" description="Dialplan Directory">
<settings>
<param name="directory-name" value="ldap"/>
<param name="host" value="ldap.freeswitch.org"/>
<param name="dn" value="cn=Manager,dc=freeswitch,dc=org"/>
<param name="pass" value="test"/>
<param name="base" value="dc=freeswitch,dc=org"/>
</settings>
</configuration>

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<configuration name="directory.conf" description="Directory">
<settings>
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
<!--<param name="dbname" value="directory"/>-->
</settings>
<profiles>
<profile name="default">
<param name="max-menu-attempts" value="3"/>
<param name="min-search-digits" value="3"/>
<param name="terminator-key" value="#"/>
<param name="digit-timeout" value="3000"/>
<param name="max-result" value="5"/>
<param name="next-key" value="6"/>
<param name="prev-key" value="4"/>
<param name="switch-order-key" value="*"/>
<param name="select-name-key" value="1"/>
<param name="new-search-key" value="3"/>
<param name="search-order" value="last_name"/>
</profile>
</profiles>
</configuration>

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<configuration name="distributor.conf" description="Distributor Configuration">
<lists>
<!-- every 10 calls to test you will get foo1 once and foo2 9 times...yes NINE TIMES! -->
<!-- this is not the same as 100 with 10 and 90 that would do foo1 10 times in a row then foo2 90 times in a row -->
<list name="test">
<node name="foo1" weight="1"/>
<node name="foo2" weight="9"/>
</list>
</lists>
</configuration>

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<configuration name="easyroute.conf" description="EasyRoute Module">
<settings>
<!-- These are kind Obvious -->
<param name="db-username" value="root"/>
<param name="db-password" value="password"/>
<param name="db-dsn" value="easyroute"/>
<!-- Default Technology and profile -->
<param name="default-techprofile" value="sofia/default"/>
<!-- IP or Hostname of Default Route -->
<param name="default-gateway" value="192.168.66.6"/>
<!-- Number of times to retry ODBC connection on connection problems, default is 120 -->
<param name="odbc-retries" value="120"/>
<!-- Customer Query. Use this with Care!!! We are not responsible if you mess
This up!!! Query *MUST* return columns in the following order!
gateway varchar(128) - contains destination gateway host:port pair (ex: 192.168.1.1:5060 )
group varchar(128) - contains optional group name
call_limit varchar(16) - contains optional call limit
tech_prefix varchar(128) - tech prefix used to build dial string (ex: sofia/default )
acctcode varchar(128) - used to set channel variable acctcode for logging into the CDRs
destination_number varchar(16) - Number returning for the query for building the dial string. (ex: 18005551212)
See Documentation on the Wiki for further information -->
<!-- <param name="custom-query" value="call FS_GET_SIP_LOCATION(%s);"/> -->
</settings>
</configuration>

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<configuration name="enum.conf" description="ENUM Module">
<settings>
<param name="default-root" value="e164.org"/>
<param name="default-isn-root" value="freenum.org"/>
<param name="auto-reload" value="true"/>
<param name="query-timeout-ms" value="200"/>
<param name="query-timeout-retry" value="2"/>
<param name="random-nameserver" value="false"/>
<!-- If you have specific (non-recursive) servers for your enum queries, specify them here ( up to 10 ) -->
<!-- <param name="nameserver" value="x.x.x.x"/> -->
<!-- <param name="nameserver" value="y.y.y.y"/> -->
</settings>
<routes>
<route service="E2U+SIP" regex="sip:(.*)" replace="sofia/${use_profile}-ipv6/$1;transport=udp|sofia/${use_profile}/$1;transport=udp"/>
<route service="E2T+SIP" regex="sip:(.*)" replace="sofia/${use_profile}-ipv6/$1;transport=tcp|sofia/${use_profile}/$1;transport=tcp"/>
<route service="E2T+SIPS" regex="sip:(.*)" replace="sofia/${use_profile}-ipv6/$1;transport=tls|sofia/${use_profile}/$1;transport=tls"/>
</routes>
</configuration>

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<configuration name="erlang_event.conf" description="Erlang Socket Client">
<settings>
<param name="listen-ip" value="0.0.0.0"/>
<param name="listen-port" value="8031"/>
<!-- Specify the first part of the node name
(the host part after the @ will be autodetected)
OR pass a complete nodename to avoid autodetection
eg. freeswitch@example or freeswitch@example.com.
If you pass a complete node name, the 'shortname' parameter has no effect. -->
<param name="nodename" value="freeswitch"/>
<!-- Specify this OR 'cookie-file' or $HOME/.erlang.cookie will be read -->
<param name="cookie" value="ClueCon"/>
<!-- Read a cookie from an arbitary erlang cookie file instead -->
<!--<param name="cookie-file" value="/$${temp_dir}/erlang.cookie"/>-->
<param name="shortname" value="true"/>
<!-- in additon to cookie, optionally restrict by ACL -->
<!--<param name="apply-inbound-acl" value="lan"/>-->
<!-- alternative is "binary" -->
<!--<param name="encoding" value="string"/>-->
<!-- provide compatability with previous OTP release (use with care) -->
<!--<param name="compat-rel" value="12"/> -->
</settings>
</configuration>

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<configuration name="event_multicast.conf" description="Multicast Event">
<settings>
<param name="address" value="225.1.1.1"/>
<param name="port" value="4242"/>
<param name="bindings" value="all"/>
<param name="ttl" value="1"/>
<!-- <param name="loopback" value="no"/>-->
<!-- Uncomment this to enable pre-shared key encryption on the packets. -->
<!-- For this option to work, you'll need to have the openssl development -->
<!-- headers installed when you ran ./configure -->
<!-- <param name="psk" value="ClueCon"/> -->
</settings>
</configuration>

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<configuration name="event_socket.conf" description="Socket Client">
<settings>
<param name="nat-map" value="false"/>
<param name="listen-ip" value="0.0.0.0"/>
<param name="listen-port" value="62221"/>
<param name="password" value="305852127a3c0a6c"/>
<!--<param name="apply-inbound-acl" value="loopback.auto"/>-->
<!--<param name="stop-on-bind-error" value="true"/>-->
</settings>
</configuration>

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<configuration name="fax.conf" description="FAX application configuration">
<settings>
<param name="use-ecm" value="true"/>
<param name="verbose" value="false"/>
<param name="disable-v17" value="false"/>
<param name="ident" value="SpanDSP Fax Ident"/>
<param name="header" value="SpanDSP Fax Header"/>
<param name="spool-dir" value="$${temp_dir}"/>
<param name="file-prefix" value="faxrx"/>
</settings>
</configuration>

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<configuration name="fifo.conf" description="FIFO Configuration">
<settings>
<param name="delete-all-outbound-member-on-startup" value="false"/>
</settings>
<fifos>
<fifo name="cool_fifo@$${domain}" importance="0">
<!--<member timeout="60" simo="1" lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
</fifo>
</fifos>
</configuration>

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<configuration name="format_cdr.conf" description="Multi Format CDR CURL logger">
<!-- You can have multiple profiles, to allow logging to both json and cdr simultaneously, or to
different paths or servers with different settings, just be sure to use different name for
each profile. -->
<profiles>
<profile name="default">
<settings>
<!-- the format of data to send, defaults to xml -->
<!-- <param name="format" value="json|xml"/> -->
<param name="format" value="xml"/>
<!-- the url to post to if blank web posting is disabled -->
<!-- <param name="url" value="http://localhost/cdr_curl/post.php"/> -->
<!-- optional: credentials to send to web server -->
<!-- <param name="cred" value="user:pass"/> -->
<!-- the total number of retries (not counting the first 'try') to post to webserver incase of failure -->
<!-- <param name="retries" value="2"/> -->
<!-- delay between retries in seconds, default is 5 seconds -->
<!-- <param name="delay" value="1"/> -->
<!-- Log via http and on disk, default is false -->
<!-- <param name="log-http-and-disk" value="true"/> -->
<!-- optional: if not present we do not log every record to disk -->
<!-- either an absolute path, a relative path assuming ${prefix}/logs or a blank value will default to ${prefix}/logs/format_cdr -->
<param name="log-dir" value=""/>
<!-- optional: if not present we do log the b leg -->
<!-- true or false if we should create a cdr for the b leg of a call-->
<param name="log-b-leg" value="false"/>
<!-- optional: if not present, all filenames are the uuid of the call -->
<!-- true or false if a leg files are prefixed "a_" -->
<param name="prefix-a-leg" value="true"/>
<!-- encode the post data may be 'true' for url encoding, 'false' for no encoding, 'base64' for base64 encoding or 'textxml' for text/xml -->
<param name="encode" value="true"/>
<!-- optional: set to true to disable Expect: 100-continue lighttpd requires this setting -->
<!--<param name="disable-100-continue" value="true"/>-->
<!-- optional: full path to the error log dir for failed web posts if not specified its the same as log-dir -->
<!-- either an absolute path, a relative path assuming ${prefix}/logs or a blank or omitted value will default to ${prefix}/logs/format_cdr -->
<!-- <param name="err-log-dir" value="$${temp_dir}"/> -->
<!-- which auhtentification scheme to use. Supported values are: basic, digest, NTLM, GSS-NEGOTIATE or "any" for automatic detection -->
<!--<param name="auth-scheme" value="basic"/>-->
<!-- optional: this will enable the CA root certificate check by libcurl to
verify that the certificate was issued by a major Certificate Authority.
note: default value is disabled. only enable if you want this! -->
<!--<param name="enable-cacert-check" value="true"/>-->
<!-- optional: verify that the server is actually the one listed in the cert -->
<!-- <param name="enable-ssl-verifyhost" value="true"/> -->
<!-- optional: these options can be used to specify custom SSL certificates
to use for HTTPS communications. Either use both options or neither.
Specify your public key with 'ssl-cert-path' and the private key with
'ssl-key-path'. If your private key has a password, specify it with
'ssl-key-password'. -->
<!-- <param name="ssl-cert-path" value="$${certs_dir}/public_key.pem"/> -->
<!-- <param name="ssl-key-path" value="$${certs_dir}/private_key.pem"/> -->
<!-- <param name="ssl-key-password" value="MyPrivateKeyPassword"/> -->
<!-- optional: use a custom CA certificate in PEM format to verify the peer
with. This is useful if you are acting as your own certificate authority.
note: only makes sense if used in combination with "enable-cacert-check." -->
<!-- <param name="ssl-cacert-file" value="$${certs_dir}/cacert.pem"/> -->
<!-- optional: specify the SSL version to force HTTPS to use. Valid options are
"SSLv3" and "TLSv1". Otherwise libcurl will auto-negotiate the version. -->
<!-- <param name="ssl-version" value="TLSv1"/> -->
<!-- optional: enables cookies and stores them in the specified file. -->
<!-- <param name="cookie-file" value="/$${temp_dir}/cookie-mod_format_cdr_curl.txt"/> -->
<!-- Whether to URL encode the individual JSON values. Defaults to true, set to false for standard JSON. -->
<param name="encode-values" value="true"/>
</settings>
</profile>
</profiles>
</configuration>

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<configuration name="graylog2.conf" description="Graylog2 Logger">
<!-- emerg - system is unusable -->
<!-- alert - action must be taken immediately -->
<!-- crit - critical conditions -->
<!-- err - error conditions -->
<!-- warning - warning conditions -->
<!-- notice - normal, but significant, condition -->
<!-- info - informational message -->
<!-- debug - debug-level message -->
<settings>
<param name="server-host" value="192.168.0.69"/>
<param name="server-port" value="12201"/>
<param name="loglevel" value="warning"/>
<!-- Uncomment if using logstash w/ gelf.rb -->
<!--param name="send-uncompressed-header" value="true"/-->
<!-- fields to add to every log associated w/ a session -->
<fields>
<!-- for example: channel variable "customer_account_number" will be the data source for the customer field in graylog2 -->
<!--field name="customer" variable="customer_account_number"/-->
</fields>
</settings>
</configuration>

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<configuration name="hash.conf" description="Hash Configuration">
<remotes>
<!-- List of hosts from where to pull usage data -->
<!-- <remote name="Test1" host="10.0.0.10" port="8021" password="ClueCon" interval="1000" /> -->
</remotes>
</configuration>

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<configuration name="hiredis.conf" description="mod_hiredis">
<profiles>
<profile name="default">
<connections>
<connection name="primary">
<param name="hostname" value="172.18.101.101"/>
<param name="password" value="redis"/>
<param name="port" value="6379"/>
<param name="timeout_ms" value="500"/>
</connection>
<connection name="secondary">
<param name="hostname" value="localhost"/>
<param name="password" value="redis"/>
<param name="port" value="6380"/>
<param name="timeout_ms" value="500"/>
</connection>
</connections>
<params>
<param name="ignore-connect-fail" value="true"/>
</params>
</profile>
</profiles>
</configuration>

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<configuration name="httapi.conf" description="HT-TAPI Hypertext Telephony API">
<settings>
<!-- print xml on the console -->
<param name="debug" value="true"/>
<!-- time to keep audio files when discovered they were deleted from the http server -->
<param name="file-not-found-expires" value="300"/>
<!-- how often to re-check the server to make sure the remote file has not changed -->
<param name="file-cache-ttl" value="300"/>
</settings>
<profiles>
<profile name="default">
<!-- default params for conference action tags -->
<conference>
<param name="default-profile" value="default"/>
</conference>
<!-- default params for dial action tags -->
<dial>
<param name="context" value="default"/>
<param name="dialplan" value="XML"/>
</dial>
<!-- permissions -->
<permissions>
<!-- <permission name="all" value="true"/> -->
<!--<permission name="none" value="true"/> -->
<permission name="set-params" value="true"/>
<permission name="set-vars" value="false">
<!-- default to "deny" or "allow" -->
<!-- type attr can be "deny" or "allow" nothing defaults to opposite of the list default so allow in this case -->
<!--
<variable-list default="deny">
<variable name="caller_id_name"/>
<variable name="hangup"/>
</variable-list>
-->
</permission>
<permission name="get-vars" value="false">
<!-- default to "deny" or "allow" -->
<!-- type attr can be "deny" or "allow" nothing defaults to opposite of the list default so allow in this case -->
<!--
<variable-list default="deny">
<variable name="caller_id_name"/>
<variable name="hangup"/>
</variable-list>
-->
</permission>
<permission name="extended-data" value="false"/>
<permission name="execute-apps" value="true">
<!-- default to "deny" or "allow" -->
<application-list default="deny">
<!-- type attr can be "deny" or "allow" nothing defaults to opposite of the list default so allow in this case -->
<application name="info"/>
<application name="hangup"/>
</application-list>
</permission>
<permission name="expand-vars-in-tag-body" value="false">
<!-- default to "deny" or "allow" -->
<!-- type attr can be "deny" or "allow" nothing defaults to opposite of the list default so allow in this case -->
<!--
<variable-list default="deny">
<variable name="caller_id_name"/>
<variable name="hangup"/>
</variable-list>
<api-list default="deny">
<api name="expr"/>
<api name="lua"/>
</api-list>
-->
</permission>
<permission name="dial" value="true"/>
<permission name="dial-set-context" value="false"/>
<permission name="dial-set-dialplan" value="false"/>
<permission name="dial-set-cid-name" value="false"/>
<permission name="dial-set-cid-number" value="false"/>
<permission name="dial-full-originate" value="false"/>
<permission name="conference" value="true"/>
<permission name="conference-set-profile" value="false"/>
</permissions>
<params>
<!-- default url can be overridden by app data -->
<param name="gateway-url" value="http://www.freeswitch.org/api/index.cgi" />
<!-- set this to provide authentication credentials to the server -->
<!--<param name="gateway-credentials" value="muser:mypass"/>-->
<!--<param name="auth-scheme" value="basic"/>-->
<!-- optional: this will enable the CA root certificate check by libcurl to
verify that the certificate was issued by a major Certificate Authority.
note: default value is disabled. only enable if you want this! -->
<!--<param name="enable-cacert-check" value="true"/>-->
<!-- optional: verify that the server is actually the one listed in the cert -->
<!-- <param name="enable-ssl-verifyhost" value="true"/> -->
<!-- optional: these options can be used to specify custom SSL certificates
to use for HTTPS communications. Either use both options or neither.
Specify your public key with 'ssl-cert-path' and the private key with
'ssl-key-path'. If your private key has a password, specify it with
'ssl-key-password'. -->
<!-- <param name="ssl-cert-path" value="$${certs_dir}/public_key.pem"/> -->
<!-- <param name="ssl-key-path" value="$${certs_dir}/private_key.pem"/> -->
<!-- <param name="ssl-key-password" value="MyPrivateKeyPassword"/> -->
<!-- optional timeout -->
<!-- <param name="timeout" value="10"/> -->
<!-- optional: maximum amount of time in seconds that is allowed to make the connection to the server -->
<!-- <param name="connect-timeout" value="2"/> -->
<!-- optional: use a custom CA certificate in PEM format to verify the peer
with. This is useful if you are acting as your own certificate authority.
note: only makes sense if used in combination with "enable-cacert-check." -->
<!-- <param name="ssl-cacert-file" value="$${certs_dir}/cacert.pem"/> -->
<!-- optional: specify the SSL version to force HTTPS to use. Valid options are
"SSLv3" and "TLSv1". Otherwise libcurl will auto-negotiate the version. -->
<!-- <param name="ssl-version" value="TLSv1"/> -->
<!-- optional: enables cookies and stores them in the specified file. -->
<!-- <param name="cookie-file" value="$${temp_dir}/cookie-mod_xml_curl.txt"/> -->
<!-- one or more of these imply you want to pick the exact variables that are transmitted -->
<!--<param name="enable-post-var" value="Caller-Unique-ID"/>-->
</params>
</profile>
</profiles>
</configuration>

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<configuration name="http_cache.conf" description="HTTP GET cache">
<settings>
<!-- set to true if you want to enable http:// and https:// formats. Do not use if mod_httapi is also loaded -->
<param name="enable-file-formats" value="false"/>
<param name="max-urls" value="10000"/>
<param name="location" value="$${cache_dir}"/>
<param name="default-max-age" value="86400"/>
<param name="prefetch-thread-count" value="8"/>
<param name="prefetch-queue-size" value="100"/>
<!-- absolute path to CA bundle file -->
<param name="ssl-cacert" value="$${certs_dir}/cacert.pem"/>
<!-- verify certificates -->
<param name="ssl-verifypeer" value="true"/>
<!-- verify host name matches certificate -->
<param name="ssl-verifyhost" value="true"/>
<!-- default is 300 seconds, override here -->
<!--param name="connect-timeout" value="300"/-->
<!-- default is 300 seconds, override here -->
<!--param name="download-timeout" value="300"/-->
</settings>
</configuration>

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<configuration name="ivr.conf" description="IVR menus">
<menus>
<X-PRE-PROCESS cmd="include" data="../ivr_menus/*.xml"/>
</menus>
</configuration>

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<configuration name="java.conf" description="Java Plug-Ins">
<javavm path="/opt/jdk1.6.0_04/jre/lib/amd64/server/libjvm.so"/>
<options>
<option value="-Djava.class.path=$${script_dir}/freeswitch.jar:$${script_dir}/example.jar"/>
<option value="-agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address=0.0.0.0:8000"/>
</options>
<startup class="org/freeswitch/example/ApplicationLauncher" method="startup"/>
</configuration>

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<?xml version="1.0" encoding="UTF-8"?>
<configuration name="kazoo.conf" description="General purpose Erlang c-node produced to better fit the Kazoo project">
<settings>
<param name="listen-ip" value="0.0.0.0" />
<param name="listen-port" value="8031" />
<!--<param name="cookie-file" value="/etc/freeswitch/autoload_configs/.erlang.cookie" />-->
<param name="cookie" value="change_me" />
<param name="shortname" value="false" />
<param name="nodename" value="freeswitch" />
<param name="send-msg-batch-size" value="10" />
<param name="receive-timeout" value="1" />
<!--<param name="receive-msg-preallocate" value="0" />-->
<!--<param name="event-stream-preallocate" value="0" />-->
<!--<param name="event-stream-framing" value="2" />-->
<!--<param name="kazoo-var-prefix" value="ecallmgr" />-->
<!--<param name="compat-rel" value="12"/> -->
</settings>
<event-filter type="whitelist">
<header name="Acquired-UUID" />
<header name="action" />
<header name="Action" />
<header name="alt_event_type" />
<header name="Answer-State" />
<header name="Application" />
<header name="Application-Data" />
<header name="Application-Name" />
<header name="Application-Response" />
<header name="att_xfer_replaced_by" />
<header name="Auth-Method" />
<header name="Auth-Realm" />
<header name="Auth-User" />
<header name="Bridge-A-Unique-ID" />
<header name="Bridge-B-Unique-ID" />
<header name="Call-Direction" />
<header name="Caller-Callee-ID-Name" />
<header name="Caller-Callee-ID-Number" />
<header name="Caller-Caller-ID-Name" />
<header name="Caller-Caller-ID-Number" />
<header name="Caller-Context" />
<header name="Caller-Controls" />
<header name="Caller-Destination-Number" />
<header name="Caller-Dialplan" />
<header name="Caller-Network-Addr" />
<header name="Caller-Unique-ID" />
<header name="Call-ID" />
<header name="Channel-Call-State" />
<header name="Channel-Call-UUID" />
<header name="Channel-Presence-ID" />
<header name="Channel-State" />
<header name="Chat-Permissions" />
<header name="Conference-Name" />
<header name="Conference-Profile-Name" />
<header name="Conference-Unique-ID" />
<header name="Conference-Size" />
<header name="New-ID" />
<header name="Old-ID" />
<header name="Detected-Tone" />
<header name="dialog_state" />
<header name="direction" />
<header name="Distributed-From" />
<header name="DTMF-Digit" />
<header name="DTMF-Duration" />
<header name="Event-Date-Timestamp" />
<header name="Event-Name" />
<header name="Event-Subclass" />
<header name="Expires" />
<header name="Ext-SIP-IP" />
<header name="File" />
<header name="FreeSWITCH-Hostname" />
<header name="from" />
<header name="Hunt-Destination-Number" />
<header name="ip" />
<header name="Message-Account" />
<header name="metadata" />
<header name="old_node_channel_uuid" />
<header name="Other-Leg-Callee-ID-Name" />
<header name="Other-Leg-Callee-ID-Number" />
<header name="Other-Leg-Caller-ID-Name" />
<header name="Other-Leg-Caller-ID-Number" />
<header name="Other-Leg-Destination-Number" />
<header name="Other-Leg-Direction" />
<header name="Other-Leg-Unique-ID" />
<header name="Participant-Type" />
<header name="Path" />
<header name="profile_name" />
<header name="Profiles" />
<header name="proto-specific-event-name" />
<header name="Raw-Application-Data" />
<header name="Resigning-UUID" />
<header name="set" />
<header name="sip_auto_answer" />
<header name="sip_auth_method" />
<header name="sip_from_host" />
<header name="sip_from_user" />
<header name="sip_to_host" />
<header name="sip_to_user" />
<header name="sub-call-id" />
<header name="technology" />
<header name="to" />
<header name="Unique-ID" />
<header name="URL" />
<header name="variable_channel_is_moving" />
<header name="variable_collected_digits" />
<header name="variable_current_application" />
<header name="variable_current_application_data" />
<header name="variable_domain_name" />
<header name="variable_effective_caller_id_name" />
<header name="variable_effective_caller_id_number" />
<header name="variable_fax_bad_rows" />
<header name="variable_fax_document_total_pages" />
<header name="variable_fax_document_transferred_pages" />
<header name="variable_fax_ecm_used" />
<header name="variable_fax_result_code" />
<header name="variable_fax_result_text" />
<header name="variable_fax_success" />
<header name="variable_fax_transfer_rate" />
<header name="variable_holding_uuid" />
<header name="variable_hold_music" />
<header name="variable_media_group_id" />
<header name="variable_originate_disposition" />
<header name="variable_playback_terminator_used" />
<header name="variable_presence_id" />
<header name="variable_record_ms" />
<header name="variable_recovered" />
<header name="variable_silence_hits_exhausted" />
<header name="variable_sip_auth_realm" />
<header name="variable_sip_from_host" />
<header name="variable_sip_from_user" />
<header name="variable_sip_h_X-AUTH-IP" />
<header name="variable_sip_received_ip" />
<header name="variable_sip_to_host" />
<header name="variable_sip_to_user" />
<header name="variable_sofia_profile_name" />
<header name="variable_transfer_history" />
<header name="variable_user_name" />
<header name="variable_endpoint_disposition" />
<header name="variable_originate_disposition" />
<header name="variable_bridge_hangup_cause" />
<header name="variable_hangup_cause" />
<header name="variable_last_bridge_proto_specific_hangup_cause" />
<header name="variable_proto_specific_hangup_cause" />
<header name="VM-Call-ID" />
<header name="VM-sub-call-id" />
<header name="whistle_application_name" />
<header name="whistle_application_response" />
<header name="whistle_event_name" />
<header name="sip_auto_answer_notify" />
<header name="eavesdrop_group" />
<header name="origination_caller_id_name" />
<header name="origination_caller_id_number" />
<header name="origination_callee_id_name" />
<header name="origination_callee_id_number" />
<header name="sip_auth_username" />
<header name="sip_auth_password" />
<header name="effective_caller_id_name" />
<header name="effective_caller_id_number" />
<header name="effective_callee_id_name" />
<header name="effective_callee_id_number" />
<!-- Registrations -->
<header name="call-id" />
<header name="profile-name" />
<header name="from-user" />
<header name="from-host" />
<header name="presence-hosts" />
<header name="contact" />
<header name="rpid" />
<header name="status" />
<header name="expires" />
<header name="to-user" />
<header name="to-host" />
<header name="network-ip" />
<header name="network-port" />
<header name="username" />
<header name="realm" />
<header name="user-agent" />
<!-- CDR Headers -->
<header name="Hangup-Cause" />
<header name="Unique-ID" />
<header name="variable_switch_r_sdp" />
<header name="variable_sip_local_sdp_str" />
<header name="variable_sip_to_uri" />
<header name="variable_sip_from_uri" />
<header name="variable_effective_caller_id_number" />
<header name="Caller-Caller-ID-Number" />
<header name="variable_effective_caller_id_name" />
<header name="Caller-Caller-ID-Name" />
<header name="Caller-Callee-ID-Name" />
<header name="Caller-Callee-ID-Number" />
<header name="Other-Leg-Unique-ID" />
<header name="variable_sip_user_agent" />
<header name="variable_duration" />
<header name="variable_billsec" />
<header name="variable_progresssec" />
<header name="variable_progress_uepoch" />
<header name="variable_progress_media_uepoch" />
<header name="variable_start_uepoch" />
<header name="variable_digits_dialed" />
<header name="variable_sip_cid_type" />
<!-- Conference Headers -->
<header name="Hear" />
<header name="Speak" />
<header name="Video" />
<header name="Talking" />
<header name="Mute-Detect" />
<header name="Member-ID" />
<header name="Member-Type" />
<header name="Energy-Level" />
<header name="Current-Energy" />
<header name="Floor" />
</event-filter>
</configuration>

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<configuration name="lcr.conf" description="LCR Configuration">
<settings>
<param name="odbc-dsn" value="freeswitch-mysql:freeswitch:Fr33Sw1tch"/>
<!-- <param name="odbc-dsn" value="freeswitch-pgsql:freeswitch:Fr33Sw1tch"/> -->
</settings>
<profiles>
<profile name="default">
<param name="id" value="0"/>
<param name="order_by" value="rate,quality,reliability"/>
</profile>
<profile name="qual_rel">
<param name="id" value="1"/>
<param name="order_by" value="quality,reliability"/>
</profile>
<profile name="rel_qual">
<param name="id" value="2"/>
<param name="order_by" value="reliability,quality"/>
</profile>
<!--
Some samples of how to do custom SQL:
=============================================================
PostgreSQL with contrib prefix module which supports fast
prefix queries. Ideal option.
=============================================================
<profile name="pg_prefix">
<param name="custom_sql" value="
SELECT l.digits AS lcr_digits,
c.carrier_name AS lcr_carrier_name,
l.${lcr_rate_field} as lcr_rate_field,
cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix,
l.lead_strip AS lcr_lead_strip, l.trail_strip AS lcr_trail_strip,
l.prefix AS lcr_prefix, l.suffix AS lcr_suffix
FROM lcr l
JOIN carriers c ON l.carrier_id=c.id
JOIN carrier_gateway cg ON c.id=cg.carrier_id
WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
AND digits_prefix @> %q
AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end
ORDER BY digits DESC, ${lcr_rate_field}, random();
"/>
</profile>
=============================================================
PostgreSQL with contrib prefix module which supports fast
prefix queries. Ideal option. Alternate syntax which requies
a session but allows variable substitution.
=============================================================
<profile name="pg_prefix2">
<param name="custom_sql" value="
SELECT l.digits AS lcr_digits,
c.carrier_name AS lcr_carrier_name,
l.${lcr_rate_field} as lcr_rate_field,
cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix,
l.lead_strip AS lcr_lead_strip, l.trail_strip AS lcr_trail_strip,
l.prefix AS lcr_prefix, l.suffix AS lcr_suffix
FROM lcr l
JOIN carriers c ON l.carrier_id=c.id
JOIN carrier_gateway cg ON c.id=cg.carrier_id
WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
AND digits_prefix @> '${lcr_query_digits}'
AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end
ORDER BY digits DESC, ${lcr_rate_field}, random();
"/>
</profile>
=============================================================
Demonstrates use of computed inlist.
=============================================================
<profile name="inlist">
<param name="custom_sql" value="
SELECT l.digits AS lcr_digits,
c.carrier_name AS lcr_carrier_name,
l.${lcr_rate_field} as lcr_rate_field,
cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix,
l.lead_strip AS lcr_lead_strip, l.trail_strip AS lcr_trail_strip,
l.prefix AS lcr_prefix, l.suffix AS lcr_suffix
FROM lcr l
JOIN carriers c ON l.carrier_id=c.id
JOIN carrier_gateway cg ON c.id=cg.carrier_id
WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
AND digits IN (${lcr_query_expanded_digits})
AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end
ORDER BY digits DESC, ${lcr_rate_field}, random();
"/>
</profile>
-->
</profiles>
</configuration>

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<configuration name="local_stream.conf" description="stream files from local dir">
<!-- fallback to default if requested moh class isn't found -->
<directory name="default" path="$${sounds_dir}/music/8000">
<param name="rate" value="8000"/>
<param name="shuffle" value="true"/>
<param name="channels" value="1"/>
<param name="interval" value="20"/>
<param name="timer-name" value="soft"/>
<!-- list of short files to break in with every so often -->
<!--<param name="chime-list" value="file1.wav,file2.wav"/>-->
<!-- frequency of break-in (seconds)-->
<!--<param name="chime-freq" value="30"/>-->
<!-- limit to how many seconds the file will play -->
<!--<param name="chime-max" value="500"/>-->
</directory>
<directory name="moh/8000" path="$${sounds_dir}/music/8000">
<param name="rate" value="8000"/>
<param name="shuffle" value="true"/>
<param name="channels" value="1"/>
<param name="interval" value="20"/>
<param name="timer-name" value="soft"/>
</directory>
<directory name="moh/16000" path="$${sounds_dir}/music/16000">
<param name="rate" value="16000"/>
<param name="shuffle" value="true"/>
<param name="channels" value="1"/>
<param name="interval" value="20"/>
<param name="timer-name" value="soft"/>
</directory>
<directory name="moh/32000" path="$${sounds_dir}/music/32000">
<param name="rate" value="32000"/>
<param name="shuffle" value="true"/>
<param name="channels" value="1"/>
<param name="interval" value="20"/>
<param name="timer-name" value="soft"/>
</directory>
<directory name="moh/48000" path="$${sounds_dir}/music/48000">
<param name="rate" value="48000"/>
<param name="shuffle" value="true"/>
<param name="channels" value="1"/>
<param name="interval" value="10"/>
<param name="timer-name" value="soft"/>
</directory>
</configuration>

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<configuration name="logfile.conf" description="File Logging">
<settings>
<!-- true to auto rotate on HUP, false to open/close -->
<param name="rotate-on-hup" value="true"/>
</settings>
<profiles>
<profile name="default">
<settings>
<!-- File to log to -->
<!--<param name="logfile" value="/var/log/freeswitch.log"/>-->
<!-- At this length in bytes rotate the log file (0 for never) -->
<param name="rollover" value="1048576000"/>
<!-- Maximum number of log files to keep before wrapping -->
<!-- If this parameter is enabled, the log filenames will not include a date stamp -->
<param name="maximum-rotate" value="32"/>
<!-- Prefix all log lines by the session's uuid -->
<param name="uuid" value="true" />
</settings>
<mappings>
<!--
name can be a file name, function name or 'all'
value is one or more of debug,info,notice,warning,err,crit,alert,all
Please see comments in console.conf.xml for more information
-->
<map name="all" value="console,debug,info,notice,warning,err,crit,alert"/>
</mappings>
</profile>
</profiles>
</configuration>

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<configuration name="lua.conf" description="LUA Configuration">
<settings>
<!--
Specify local directories that will be searched for LUA modules
These entries will be pre-pended to the LUA_CPATH environment variable
-->
<!-- <param name="module-directory" value="/usr/lib/lua/5.1/?.so"/> -->
<!-- <param name="module-directory" value="/usr/local/lib/lua/5.1/?.so"/> -->
<!--
Specify local directories that will be searched for LUA scripts
These entries will be pre-pended to the LUA_PATH environment variable
-->
<!-- <param name="script-directory" value="/usr/local/lua/?.lua"/> -->
<!-- <param name="script-directory" value="$${script_dir}/?.lua"/> -->
<!--<param name="xml-handler-script" value="/dp.lua"/>-->
<!--<param name="xml-handler-bindings" value="dialplan"/>-->
<!--
The following options identifies a lua script that is launched
at startup and may live forever in the background.
You can define multiple lines, one for each script you
need to run.
-->
<!--<param name="startup-script" value="startup_script_1.lua"/>-->
<!--<param name="startup-script" value="startup_script_2.lua"/>-->
<!--<hook event="CUSTOM" subclass="conference::maintenance" script="catch-event.lua"/>-->
</settings>
</configuration>

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<configuration name="memcache.conf" description="memcache Configuration">
<settings>
<!-- comma sep list of servers: eg: localhost,otherhost:port,anotherone -->
<param name="memcache-servers" value="localhost"/>
</settings>
</configuration>

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<configuration name="modules.conf" description="Modules">
<modules>
<!-- Loggers (I'd load these first) -->
<load module="mod_console"/>
<!-- <load module="mod_graylog2"/> -->
<load module="mod_logfile"/>
<!-- <load module="mod_syslog"/> -->
<!--<load module="mod_yaml"/>-->
<!-- Multi-Faceted -->
<!-- mod_enum is a dialplan interface, an application interface and an api command interface -->
<load module="mod_enum"/>
<!-- XML Interfaces -->
<!-- <load module="mod_xml_rpc"/> -->
<!-- <load module="mod_xml_curl"/> -->
<!-- <load module="mod_xml_cdr"/> -->
<!-- <load module="mod_xml_radius"/> -->
<!-- <load module="mod_xml_scgi"/> -->
<!-- Event Handlers -->
<!-- <load module="mod_amqp"/> -->
<load module="mod_cdr_csv"/>
<!-- <load module="mod_cdr_sqlite"/> -->
<!-- <load module="mod_event_multicast"/> -->
<load module="mod_event_socket"/>
<!-- <load module="mod_event_zmq"/> -->
<!-- <load module="mod_zeroconf"/> -->
<!-- <load module="mod_erlang_event"/> -->
<!-- <load module="mod_smpp"/> -->
<!-- <load module="mod_snmp"/> -->
<!-- Directory Interfaces -->
<!-- <load module="mod_ldap"/> -->
<!-- Endpoints -->
<!-- <load module="mod_dingaling"/> -->
<!-- <load module="mod_portaudio"/> -->
<!-- <load module="mod_alsa"/> -->
<load module="mod_sofia"/>
<load module="mod_loopback"/>
<!-- <load module="mod_woomera"/> -->
<!-- <load module="mod_freetdm"/> -->
<!-- <load module="mod_unicall"/> -->
<!-- <load module="mod_skinny"/> -->
<!-- <load module="mod_khomp"/> -->
<load module="mod_rtc"/>
<!-- <load module="mod_rtmp"/> -->
<load module="mod_verto"/>
<!-- Applications -->
<!-- <load module="mod_signalwire"/> -->
<load module="mod_commands"/>
<load module="mod_conference"/>
<!-- <load module="mod_curl"/> -->
<load module="mod_db"/>
<load module="mod_dptools"/>
<load module="mod_expr"/>
<load module="mod_fifo"/>
<load module="mod_hash"/>
<!--<load module="mod_mongo"/> -->
<load module="mod_voicemail"/>
<!--<load module="mod_directory"/>-->
<!--<load module="mod_distributor"/>-->
<!--<load module="mod_lcr"/>-->
<!--<load module="mod_easyroute"/>-->
<load module="mod_esf"/>
<load module="mod_fsv"/>
<!--<load module="mod_cluechoo"/>-->
<load module="mod_valet_parking"/>
<!--<load module="mod_fsk"/>-->
<!--<load module="mod_spy"/>-->
<!--<load module="mod_sms"/>-->
<!--<load module="mod_sms_flowroute"/>-->
<!--<load module="mod_smpp"/>-->
<!--<load module="mod_random"/>-->
<load module="mod_httapi"/>
<!--<load module="mod_translate"/>-->
<!-- SNOM Module -->
<!--<load module="mod_snom"/>-->
<!-- This one only works on Linux for now -->
<!--<load module="mod_ladspa"/>-->
<!-- Dialplan Interfaces -->
<!-- <load module="mod_dialplan_directory"/> -->
<load module="mod_dialplan_xml"/>
<load module="mod_dialplan_asterisk"/>
<!-- Codec Interfaces -->
<load module="mod_spandsp"/>
<load module="mod_g723_1"/>
<load module="mod_g729"/>
<load module="mod_amr"/>
<!--<load module="mod_ilbc"/>-->
<!--<load module="mod_h26x"/>-->
<load module="mod_b64"/>
<!--<load module="mod_siren"/>-->
<!--<load module="mod_isac"/>-->
<load module="mod_opus"/>
<!-- File Format Interfaces -->
<load module="mod_av"/>
<load module="mod_sndfile"/>
<load module="mod_native_file"/>
<!--<load module="mod_opusfile"/>-->
<load module="mod_png"/>
<!-- <load module="mod_shell_stream"/> -->
<!--For icecast/mp3 streams/files-->
<!--<load module="mod_shout"/>-->
<!--For local streams (play all the files in a directory)-->
<load module="mod_local_stream"/>
<load module="mod_tone_stream"/>
<!-- Timers -->
<!-- <load module="mod_timerfd"/> -->
<!-- <load module="mod_posix_timer"/> -->
<!-- Languages -->
<!-- <load module="mod_v8"/> -->
<!-- <load module="mod_perl"/> -->
<!-- <load module="mod_python"/> -->
<!-- <load module="mod_python3"/> -->
<!-- <load module="mod_java"/> -->
<load module="mod_lua"/>
<!-- ASR /TTS -->
<!-- <load module="mod_flite"/> -->
<!-- <load module="mod_pocketsphinx"/> -->
<!-- <load module="mod_cepstral"/> -->
<!-- <load module="mod_tts_commandline"/> -->
<!-- <load module="mod_rss"/> -->
<!-- Say -->
<load module="mod_say_en"/>
<!-- <load module="mod_say_ru"/> -->
<!-- <load module="mod_say_zh"/> -->
<!-- <load module="mod_say_sv"/> -->
<!-- Third party modules -->
<!--<load module="mod_nibblebill"/>-->
<!--<load module="mod_callcenter"/>-->
<load module="mod_unimrcp"/>
</modules>
</configuration>

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<configuration name="mongo.conf">
<settings>
<!--
mongodb://[username:password@]host1[:port1][,host2[:port2],...[,hostN[:portN]]][/[database][?options]]
-->
<param name="connection-string" value="mongodb://127.0.0.1:27017/?connectTimeoutMS=10000"/>
<!--
<param name="map" value="function() { emit(this.a, 1); }"/>
<param name="reduce" value="function(key, values) { return Array.sum(values); }"/>
<param name="finalize" value="function(key, value) { return value;}"/>
-->
</settings>
</configuration>

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<configuration name="msrp.conf" description="MSRP">
<settings>
<param name="listen-ip" value="$${local_ip_v4}"/>
<!-- <param name="listen-port" value="2855"/> -->
<!-- <param name="listen-ssl-port" value="2856"/> -->
<!-- <param name="message-buffer-size" value="50"/> -->
<!-- <param name="debug" value="true"/> -->
<!-- <param name="secure-cert" value="$${certs_dir}/wss.pem"/> -->
<!-- <param name="secure-key" value="$${certs_dir}/wss.pem"/> -->
</settings>
</configuration>

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<configuration name="nibblebill.conf" description="Nibble Billing">
<settings>
<!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help with these options -->
<!-- Information for connecting to your database -->
<param name="odbc-dsn" value="bandwidth.com"/>
<!-- The database table where your CASH column is located -->
<param name="db_table" value="accounts"/>
<!-- The column name where we store the value of the account -->
<param name="db_column_cash" value="cash"/>
<!-- The column name for the unique ID identifying the account -->
<param name="db_column_account" value="id"/>
<!-- Custom SQL for loading current balance - overrides column names
channel vars are interpreted.
field nibble_balance is used for balance info
<param name="custom_sql_lookup" value="SELECT cash AS nibble_balance FROM accounts WHERE account_code='${nibble_account}'"/>
-->
<!-- Custom SQL for loading current balance - overrides column names
channel vars are interpreted.
nibble_increment is the amount to update
<param name="custom_sql_save" value="UPDATE accounts SET cash=cash-${nibble_increment} WHERE account_code='${nibble_account}'"/>
-->
<!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e. bill only at end of call) -->
<param name="global_heartbeat" value="60"/>
<!-- By default, warn a caller when their balance is at $5.00. You can set this to a negative number. -->
<param name="lowbal_amt" value="5"/>
<param name="lowbal_action" value="play ding"/>
<!-- By default, terminate a caller when their balance hits $0.00. You can set this to a negative number. -->
<param name="nobal_amt" value="0"/>
<param name="nobal_action" value="hangup"/>
<!-- If a call goes beyond a certain dollar amount, flag or terminate it -->
<param name="percall_max_amt" value="100"/>
<param name="percall_action" value="hangup"/>
</settings>
</configuration>

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<configuration name="opal.conf" description="Opal Endpoints">
<settings>
<param name="trace-level" value="3"/>
<param name="context" value="default"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-type" value="signal"/> <!-- string, signal, rfc2833, in-band -->
<param name="jitter-size" value="40,100"/> <!-- Jitter buffer min/max size, milliseconds -->
<!-- <param name="codec-prefs" value="PCMU,PCMA"/> --> <!-- list, and preferecnce order, of codecs -->
<!-- <param name="disable-transcoding" value="true"/> --> <!-- do not transcode, use source channel codec only -->
<param name="gk-address" value=""/> <!-- empty to disable, "*" to search LAN -->
<param name="gk-identifer" value=""/> <!-- optional name of gk -->
<param name="gk-interface" value="$${local_ip_v4}"/> <!-- optional listener interface name -->
</settings>
<listeners>
<listener name="default">
<param name="h323-ip" value="$${local_ip_v4}"/>
<param name="h323-port" value="1720"/>
</listener>
</listeners>
</configuration>

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<configuration name="opus.conf">
<settings>
<param name="use-vbr" value="1"/>
<!--<param name="use-dtx" value="1"/>-->
<param name="complexity" value="10"/>
<!-- Set the initial packet loss percentage 0-100 -->
<!--<param name="packet-loss-percent" value="10"/>-->
<!-- Support asymmetric sample rates -->
<!--<param name="asymmetric-sample-rates" value="1"/>-->
<!-- Enable bitrate negotiation -->
<!--<param name="bitrate-negotiation" value="1"/>-->
<!-- Keep FEC Enabled -->
<param name="keep-fec-enabled" value="1"/>
<!--<param name="use-jb-lookahead" value="true"/> -->
<!--
maxaveragebitrate: the maximum average codec bitrate (values: 6000 to 510000 in bps) 0 is not considered
maxplaybackrate: the maximum codec internal frequency (values: 8000, 12000, 16000, 24000, 48000 in Hz) 0 is not considered
This will set the local encoder and instruct the remote encoder trough specific "fmtp" attibute in the SDP.
Example: if you receive "maxaveragebitrate=20000" from SDP and you have set "maxaveragebitrate=24000" in this configuration
the lowest will prevail in this case "20000" is set on the encoder and the corresponding fmtp attribute will be set
to instruct the remote encoder to do the same.
-->
<param name="maxaveragebitrate" value="0"/>
<param name="maxplaybackrate" value="0"/>
<!-- Max capture rate, 8000, 12000, 16000, 24000 and 48000 are valid options -->
<!--<param name="sprop-maxcapturerate" value="0"/>-->
<!-- Enable automatic bitrate variation during the call based on RTCP feedback -->
<!--<param name="adjust-bitrate" value="1"/>-->
<!-- will enforce mono even if the remote party wants stereo. must be used in conjunction with param "max-audio-channels" set to 1 in switch.conf.xml. -->
<param name="mono" value="0"/>
</settings>
</configuration>

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<!--
To use this application simply install the open source Oreka recorder server (Orkaudio) and point
the sip-server-addr and sip-server-port to the oreka server
-->
<configuration name="oreka.conf" description="Oreka Recorder configuration">
<settings>
<!-- Oreka/Orkaudio recording server address -->
<!-- <param name="sip-server-addr" value="192.168.1.200"/> -->
<!-- Which port to send signaling to in the recording server -->
<!-- <param name="sip-server-port" value="6000"/> -->
</settings>
</configuration>

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<configuration name="osp.conf" description="OSP Module Configuration">
<settings>
<!-- Debug info flag -->
<param name="debug-info" value="disabled"/>
<!-- Log level for debug info -->
<param name="log-level" value="info"/>
<!-- Crypto hareware accelerate is disabled by default -->
<param name="crypto-hardware" value="disabled"/>
<!-- SIP settings -->
<param name="sip" module="sofia" profile="external"/>
<!-- H.323 settings -->
<!-- <param name="h323" module="h323" profile="external"/> -->
<!-- IAX settings -->
<!-- <param name="iax" module="iax" profile="external"/> -->
<!-- Skype settings -->
<!-- <param name="skype" module="skypopen" profile="external"/> -->
<!-- Default destination protocol -->
<param name="default-protocol" value="sip"/>
</settings>
<profiles>
<!-- Default OSP profile -->
<profile name="default">
<!-- Service point URLs, up to 8 allowed -->
<!-- <param name="service-point-url" value="http://osptestserver.transnexus.com:5045/osp"/> -->
<!-- <param name="service-point-url" value="https://127.0.0.1:1443/osp"/> -->
<param name="service-point-url" value="http://127.0.0.1:5045/osp"/>
<!-- FreeSWITCH IP address for OSP -->
<param name="device-ip" value="127.0.0.1:5080"/>
<!-- SSL lifetime in seconds -->
<param name="ssl-lifetime" value="300"/>
<!-- HTTP max connections, 1~1000 -->
<param name="http-max-connections" value="20"/>
<!-- HTTP persistence in seconds -->
<param name="http-persistence" value="60"/>
<!-- HTTP retry delay in seconds, 0~10 -->
<param name="http-retry-delay" value="0"/>
<!-- HTTP retry limit, 0~100 -->
<param name="http-retry-limit" value="2"/>
<!-- HTTP timeout in milliseconds, 200~60000 -->
<param name="http-timeout" value="10000"/>
<!-- OSP work mode, direct or indirect -->
<param name="work-mode" value="direct"/>
<!-- OSP service type, voice or npquery -->
<param name="service-type" value="voice"/>
<!-- Max number of destinations -->
<param name="max-destinations" value="12"/>
</profile>
</profiles>
</configuration>

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<configuration name="perl.conf" description="PERL Configuration">
<settings>
<!--<param name="xml-handler-script" value="$${temp_dir}/xml.pl"/>-->
<!--<param name="xml-handler-bindings" value="dialplan"/>-->
<!--
The following options identifies a perl script that is launched
at startup and may live forever in the background.
You can define multiple lines, one for each script you
need to run.
-->
<!--param name="startup-script" value="startup_script_1.pl"/-->
<!--param name="startup-script" value="startup_script_2.pl"/-->
</settings>
</configuration>

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<configuration name="pocketsphinx.conf" description="PocketSphinx ASR Configuration">
<settings>
<param name="threshold" value="400"/>
<param name="silence-hits" value="25"/>
<param name="listen-hits" value="1"/>
<param name="auto-reload" value="true"/>
<!--<param name="language-weight" value="1"/>-->
<!--<param name="narrowband-model" value="communicator"/>-->
<!--<param name="wideband-model" value="wsj1"/>-->
<!--<param name="dictionary" value="default.dic"/>-->
</settings>
</configuration>

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