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:

 

openzap.conf.xml :

 

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@njpbx.vicp.net"/>

     </condition> 

   </extension>