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