Hi everyone! In my environment I use FreeSwitch as media server and session border controller. SIP routing is mostly done with my private B2BUA. The problem itself is in my "hold" functionality. In details: A is calling to B: <!-- if the calling party is the called party, go to their VM if the calling party is NOT the called party dial the extension (1000-1019) for 30 seconds and go to voicemail if the call fails (continue_on_fail=true), otherwise hang up after a successful bridge (hangup_after-bridge=true) --> <extension name="Local_Extension_from_SP"> <condition field="${sip_contact_uri}" expression="^.*StarPound_.*_B2BUA.*$"/> <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(${dialed_extension}@${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@ $${starpound_sip_app_server}"/> </condition> </extension> Call being answered. Then B wants to hold a call (A should hear music-on-hold, B should hear nothing but still have a call). To accomplish this I: 1. on behalf of B2BUA make call to FreeSwitch "hold" extension, get SDP for music-on-hold (moh_sdp) <extension name="hold"> <condition field="destination_number" expression="hold"> <action application="answer"/> <action application="playback" data="$${hold_music}"/> </condition> </extension> 2. re-invite A with SDP moh_sdp 3. re-invite B with it's own sdp and attribute 'inactive' (also tried re-invite with A's SDP and 'inactive' and with 0.0.0.0 IP in SDP and 'inactive' - same results) Result: both A and B hear music-on-hold. I suspect that the reason is in 'bridge' function.. it seems, that it still bridges media against the SIP. Here is a snippet from FreeSwitch log related to "strange" (from my point of view) behaviour - when FreeSwitch change media for B to the incorrect one. Here 1004 is B. 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2939 sofia_glue_negotiate_sdp() Audio Codec Compare [GSM:3:8000:20]/[GSM:3:8000:20] 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:1863 sofia_glue_tech_set_codec() Already using GSM 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2899 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2908 sofia_glue_negotiate_sdp() Set comfort noise payload to 13 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2139 sofia_glue_activate_rtp() Audio params changed for sofia/external/6782886802@10.249.194.9:5060 from 75.101.166.29:27426 to 75.101.166.29:22922 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/6782886802@10.249.194.9:5060] 10.249.194.9 port 28594 -> 75.101.166.29 port 22922 codec: 3 ms: 20 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2165 sofia_glue_activate_rtp() AUDIO RTP CHANGING DEST TO: [75.101.166.29:22922] 2009-05-26 13:19:14 [DEBUG] sofia.c:2996 sofia_handle_sip_i_state() Channel sofia/external/1005@10.249.194.9 entering state [ready][200] 2009-05-26 13:19:15 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() sofia/doublenat5090/sip:1004@172.16.0.6 <sip%3A1004@172.16.0.6> ;fs_nat=yes;fs_path=sip%3A1004%40193.33.170.63%3A5060 Command Execute playback(local_stream://moh) EXECUTE sofia/doublenat5090/sip:1004@172.16.0.6 <sip%3A1004@172.16.0.6>;fs_nat=yes;fs_path=sip%3A1004%40193.33.170.63%3A5060 playback(local_stream://moh) 2009-05-26 13:19:15 [DEBUG] mod_local_stream.c:346 local_stream_file_open() Opening Stream [moh/8000] 8000hz 2009-05-26 13:19:15 [DEBUG] switch_ivr_play_say.c:1084 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2009-05-26 13:19:15 [DEBUG] switch_core_io.c:649 switch_core_session_write_frame() sofia/doublenat5090/sip:1004@172.16.0.6 <sip%3A1004@172.16.0.6> ;fs_nat=yes;fs_path=sip%3A1004%40193.33.170.63%3A5060 receive message [TRANSCODING_NECESSARY] Any thoughts would be really appreciated! Artem