- 重点说明freeswitch的配置
-
- 我们假设asterisk的IP为210.134.185.9,有个sip号码为60006
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- 1、asterisk配置
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- 修改sip.conf,添加如下内容:
-
- [fs_zmrh]
- username=fs_zmrh
- secret=123
- host=dynamic
- type=peer
- nat=yes
- context=from-internal
-
-
- 2、配置domain
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- 修改freeswitch安装目录下conf/drectory/default.xml,添加如下内容:
- <domain name="210.134.185.9">
- <params>
- <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
- </params>
- <variables>
- <variable name="record_stereo" value="true"/>
- <variable name="default_areacode" value="$${default_areacode}"/>
- <variable name="transfer_fallback_extension" value="operator"/>
- </variables>
- <user id="210.134.185.9">
- <gateways>
- <X-PRE-PROCESS cmd="include" data="gateway/*.xml"/>
- </gateways>
- </user>
- </domain>
-
-
- 3、配置网关(gateway)
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- 在freeswtich的conf/directory/目录下新建文件夹gateway,在gateway文件夹下新建一个xml文件,内容如下:
- <include>
- <gateway name="asterisk">
- <param name="username" value="fs_zmrh"/>
- <param name="password" value="123"/>
- <param name="realm" value="210.134.185.9"/>
- <param name="from-domain" value="210.134.185.9"/>
- <param name="expire-seconds" value="600"/>
- <param name="register" value="false"/>
- </gateway>
- </include>
-
-
- 4、配置呼叫规则
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- 修改freeswtich安装目录下的conf/dialplan/default.xml,添加内容如下:
- <extension name="extension-asterisk">
- <condition field="destination_number" expression="^(6[01][01][0-9][0-9])$">
- <action application="set" data="dialed_extension=$1"/>
- <action application="bridge" data="sofia/gateway/asterisk/$1"/>
- </condition>
- </extension>
-
- 配置完毕,启动freeswitch即可进行呼叫
-
- 注意:
- 如果freeswitch和asterisk都在内网,请修改freeswtich安装目录下的conf/sip_profiles下的external.xml,如下,原来为:
- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
- <param name="ext-sip-ip" value="$${external_sip_ip}"/>
-
- 修改为:
- <param name="ext-rtp-ip" value="$${local_ip_v4}"/>
- <param name="ext-sip-ip" value="$${local_ip_v4}"/>

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