I am building an ostn node. I followed a recipe, but got caught up with learning FS basics. I had this trouble that calls were being signalled, but no audio was actually put through. I thought it was nat problems, but connectivity was going fine. This is my dialplan snippet: <extension name="Local_Extension"> <condition field="destination_number" expression="^(\d{3})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <!-- <bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <!-- <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> --> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <!-- <action application="answer"/> --> <action application="info"/> <action application="answer"/> <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> <!-- <action application="answer"/> --> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> This works, but you can see i have an answer app right before the bridge. If i don't answer the call, i don't get to the point where i get Activating SRTP RECV/SRTP SEND. Yet both channels get answered and RTP goes both ways, but i have a feeling it's not SRTP and thus is not being decrypted. I have this in the global context: <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_80)$" break="never"> <action application="set" data="sip_secure_media=true"/> <!-- Offer SRTP on outbound legs if we have it on inbound. --> <action application="export" data="sip_secure_media=true"/> </condition> which should activate SRTP on both legs. I have pasted in to the bin a debug log with and without the answer and media sending here : http://pastebin.freeswitch.org/19937