Hello! I got a problem with one way audio, symptoms are: firstly play audio file to channel A (A is hears sound) secondly bridge channel B with A (A doesn't hear B).
Environment: - no NAT - User Agents being used X-Lite, EyeBeam, SJphone - same result for all of them- no audio, Wireshark shows that there is no RTP-flow to A from FreeSwitch - dialplan: <extension name="playback_media_file"> <condition field="destination_number" expression="playmedia"> <action application="answer"/> <action application="playback" data="test.wav"/> </condition> </extension> <extension name="Local_Extension_from_SP"> <condition field="destination_number" expression="^([0-9]{2,9})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> </condition> <condition field="${sip_to_host}" expression="^([^.]*)\..*$"> <action application="set" data="orgname=$1"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <anti-action application="set" data="ringback=${us-ring}"/> <anti-action application="set" data="transfer_ringback=${us-ring}"/> <anti-action application="set" data="call_timeout=10"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="db" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <anti-action application="db" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <anti-action application="set" data="called_party_callgroup=${user_data(${dialed_extensi...@${domain_name} var callgroup)}"/> <anti-action application="db" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <anti-action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> <anti-action application="answer"/> <anti-action application="export" data="sip_h_X-SPFrom="e;${sip_from_user}"e;<${sip_from_uri}>"/> <anti-action application="export" data="sip_h_X-SPTo=<${sip_to_uri}>"/> <anti-action application="export" data="sip_h_X-SPCallId=${sip_call_id}"/> <anti-action application="bridge" data="sofia/external/${orgname}send2voicemail@ $${starpound_sip_app_server}"/> </condition> </extension> - Call routing scheme: user calls to FS, FS calls to my B2BUA which manage call with SIP 3pcc Exact description what's going on is: user A -> FS -(bridge)-> my B2BUA Then my B2BUA transfers (using re-INVITE's), on behalf of user, call to extension "playback_media_file" . After a while B2BUA transfer (re-Inviting) user to extension "Local_Extension_from_SP". This should create a new call to user B. As a result - A doesn't hear B, but B- is OK. On the contrary, if a call is routed (by B2BUA) to the "Local_Extension_from_SP" extension (ommiting "playback_media_file" ext) - everything is OK. What I've tried: - set parameter "inbound-proxy-media" to "true" in Sofia profile - set parameter "disable_rtp_auto_adjust to "true" in Sofia profile Nothing helps. Any help or thoughts would be MUCH appreciated! Artem
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