[Freeswitch-users] busy tone detect issue

Michael Jerris mike at jerris.com
Thu Jun 4 02:05:43 PDT 2009


Are you having this issue on your analog or pri lines? what does your  
openzap.conf look like?

Mike

On Jun 4, 2009, at 4:28 AM, god.nirvana wrote:

> hi all
>      i am new to freeswitch.
> there are some busy tone detect issues,i hope someone could help me.
> i installed freeswitch from trunk,openzap,zaptel....
> but i found some busy tone isuues
>
> my tones.conf:
> [us]
> generate-dial => v=-7;%(1000,0,350,440)
> detect-dial => 350,440
> generate-ring => v=-7;%(2000,4000,440,480)
> detect-ring => 440,480
> generate-busy => v=-7;%(500,500,450,340)
> detect-busy =>450,340
> generate-attn => v=0;%(100,100,1400,2060,2450,2600)
> detect-attn => 1400,2060,2450,2600
> generate-callwaiting-sas => v=0;%(300,0,440)
> detect-callwaiting-sas => 440
> generate-callwaiting-cas => v=0;%(80,0,2750,2130)
> detect-callwaiting-cas => 2750,2130
> detect-fail1 => 913.8
> detect-fail2 => 1370.6
> detect-fail3 => 776.7
>
>
> openzap.conf.xml :
> <configuration name="openzap.conf" description="OpenZAP  
> Configuration">
>   <settings>
>     <param name="debug" value="0"/>
>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>     <!--<param name="enable-analog-option" value="3-way"/>-->
>   </settings>
>    <pri_spans>
>      <span name="PRI_1">
>        <!-- Log Levels: none, alert, crit, err, warning, notice,  
> info, debug -->
>        <param name="q921loglevel" value="alert"/>
>        <param name="q931loglevel" value="alert"/>
>        <param name="mode" value="user"/>
>        <param name="dialect" value="5ess"/>
>        <param name="dialplan" value="XML"/>
>        <param name="context" value="default"/>
>      </span>
>      <span name="PRI_2">
>        <param name="q921loglevel" value="alert"/>
>        <param name="q931loglevel" value="alert"/>
>        <param name="mode" value="user"/>
>        <param name="dialect" value="5ess"/>
>        <param name="dialplan" value="XML"/>
>        <param name="context" value="default"/>
>      </span>
>    </pri_spans>
>   <!-- one entry here per openzap span -->
>   <analog_spans>
>     <span id="1">
>       <!--<param name="hold-music" value="$${moh_uri}"/>-->
>       <!--<param name="enable-analog-option" value="call-swap"/>-->
>       <!--<param name="enable-analog-option" value="3-way"/>-->
>       <param name="tonegroup" value="us"/>
>       <param name="digit-timeout" value="2000"/>
>       <param name="max-digits" value="11"/>
>       <param name="dialplan" value="XML"/>
>       <param name="context" value="default"/>
>       <param name="enable-callerid" value="true"/>
>       <!-- regex to stop dialing when it matches -->
>       <!--<param name="dial-regex" value="5555"/>-->
>       <!-- regex to stop dialing when it does not match -->
>       <!--<param name="fail-dial-regex" value="^5"/>-->
>     </span>
>     <span id="2">
>       <!--<param name="hold-music" value="$${moh_uri}"/>-->
>       <!--<param name="enable-analog-option" value="call-swap"/>-->
>       <!--<param name="enable-analog-option" value="3-way"/>-->
>       <param name="tonegroup" value="us"/>
>       <param name="digit-timeout" value="2000"/>
>       <param name="max-digits" value="11"/>
>       <param name="dialplan" value="XML"/>
>       <param name="context" value="default"/>
>       <param name="enable-callerid" value="true"/>
>       <!-- regex to stop dialing when it matches -->
>       <!--<param name="dial-regex" value="5555"/>-->
>       <!-- regex to stop dialing when it does not match -->
>       <!--<param name="fail-dial-regex" value="^5"/>-->
>     </span>
>     <span id="3">
>       <!--<param name="hold-music" value="$${moh_uri}"/>-->
>       <!--<param name="enable-analog-option" value="call-swap"/>-->
>       <!--<param name="enable-analog-option" value="3-way"/>-->
>       <param name="tonegroup" value="us"/>
>       <param name="digit-timeout" value="2000"/>
>       <param name="max-digits" value="11"/>
>       <param name="dialplan" value="XML"/>
>       <param name="context" value="default"/>
>       <param name="enable-callerid" value="true"/>
>       <!-- regex to stop dialing when it matches -->
>       <!--<param name="dial-regex" value="5555"/>-->
>       <!-- regex to stop dialing when it does not match -->
>       <!--<param name="fail-dial-regex" value="^5"/>-->
>     </span>
>     <span id="4">
>       <!--<param name="hold-music" value="$${moh_uri}"/>-->
>       <!--<param name="enable-analog-option" value="call-swap"/>-->
>       <!--<param name="enable-analog-option" value="3-way"/>-->
>       <param name="tonegroup" value="us"/>
>       <param name="digit-timeout" value="2000"/>
>       <param name="max-digits" value="11"/>
>       <param name="dialplan" value="XML"/>
>       <param name="context" value="default"/>
>       <param name="enable-callerid" value="true"/>
>       <!-- regex to stop dialing when it matches -->
>       <!--<param name="dial-regex" value="5555"/>-->
>       <!-- regex to stop dialing when it does not match -->
>       <!--<param name="fail-dial-regex" value="^5"/>-->
>     </span>
>   </analog_spans>
> </configuration>
>
> when i call the pstn phone from a ip phone,if the pstn call hangup  
> first,the ip phone will hear the busy tone,but the system does not  
> handle the busytone ,the channel does not erase. so i have to add
> <action application="tone_detect" data="busy 450,340 w +25000 hangup  
> 'normal_clearing' 3"/> in the dialplan.and it works.the channel  
> erased.
>
> but in the conference case,pstn phone call in,hangup. all  
> participants hear the tone,"do ~,do~~".freeswitch doest handle it.
> so i change the conference dialplan.
>
>
> <extension name="incoming-fxo-channel-1">
> <condition field="source" expression="mod_openzap">
> <action application="tone_detect" data="busy 450,340 w +25000 hangup  
> 'normal_clearing' 3"/>
> <action application="bridge" data="sofia/maqian/6789 at njpbx.vicp.net"/>
>      </condition>
>    </extension>
>
> restart freeswitch,try again,freeswitch not handle the hangup tone  
> still,all participants hear the tone.
> how to solve it?could some one help me ???
> thx!
> BR
>
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