这时需要用的continue_on_fail变量
如下设置可实现呼叫sip失败后继续呼叫后面的pstn
引用
continue_on_fail=true
详细配置如下
引用
<extension name="Goip outbound">
<condition field="destination_number" expression="^01(\d+)$">
<action application="log" data="INFO goip outbound to $1"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="ringback=${us-ring}"/>
<action application="bridge" data="user/$1"/>
<action application="bridge" data="sofia/gateway/pstn/11$1"/>
</condition>
</extension>
<condition field="destination_number" expression="^01(\d+)$">
<action application="log" data="INFO goip outbound to $1"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="ringback=${us-ring}"/>
<action application="bridge" data="user/$1"/>
<action application="bridge" data="sofia/gateway/pstn/11$1"/>
</condition>
</extension>
简单解释：
用户拨打 01XXXXXXXX进入
主叫先与user/$1建立桥接失败
因为continue_on_fail=true 所以继续执行
与sofia/gateway/pstn/11$1建立桥接