[Freeswitch-users] sip to h323 video call via freeswitch

[Navnath Sonavne]

Anthony,



any domain has the same dial-string:

<param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>



and when an extension calls another extension I use:

<action application="bridge" data="{ignore_early_media=true}sofia/$${domain}/${Called_Exten}"/>


Hi Anthony,


You said any domain has same dial-string so i searched it in various files but i did not find it.
Can you tell me in which file i can find it?

In reply you said  when an extension calls another extension I use:

<action application="bridge" data="{ignore_early_media=true}sofia/$${domain}/${Called_Exten}"/>

I used this action in my default.xml but no success.

FYI I am able to convert sip protocol to h323 protocol for Audio call.
Here is my extension from default.xml to convert sip to h323 : 

<extension name="Local_Extension">
       <condition field="destination_number" expression="^(\d{4})$">
      <action application="export" data="dialed_extension=$1"/>
      <action application="set" data="effective_caller_id_number=${dialed_extension}"/>
      <action application="set" data="PEER_IP=192.168.8.103"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="set" data="proxy_media=true"/>
      <action application="bridge" data="h323/$1@${PEER_IP}"/>
      </condition>     
    </extension>

I am using this for video call also but no success.

I am attaching vars.xml , default.xml , internal.xml , h323.conf.xml , opal.conf.xml and sofia.conf.xml.
Please look in these files and tell me if am I going wrong and what should be done to make sip to h323 video call?

-Navnath
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