Updates: 1. One-way audio is in 95% tries. But how the rest 5% works?? 2. Strange FS logging after the channels are bridged (user A talk to user B) 2009-06-26 02:16:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1005@192.168.147.1 entering state [ready] 2009-06-26 02:16:07 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP: v=0 o=FreeSWITCH 3898736844884231 105343281763004076 IN IP4 192.168.147.130 s=FreeSWITCH c=IN IP4 192.168.147.130 t=0 0 m=audio 31134 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 m=video 0 RTP/AVP 34 a=rtpmap:34 H263/90000 2009-06-26 02:16:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1000000000@192.168.147.130:5060 entering state [ready] freeswitch@localhost.localdomain> 2009-06-26 02:17:09 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/ 1005@uat.pbx.starpoundtech.net entering state [calling] 2009-06-26 02:17:09 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1005@uat.pbx.starpoundtech.net entering state [ready] 2009-06-26 02:17:09 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 1 3 IN IP4 192.168.147.1 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.147.1 t=0 0 m=audio 47590 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 freeswitch@localhost.localdomain> show calls API CALL [show(calls)] output: created,created_epoch,function,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid 2009-06-26 02:16:05,1245968165,switch_ivr_multi_threaded_bridge,1005,1005,inbound1000000000,sofia/external/ 1005@uat.pbx.starpoundtech.net ,4fa86434-b542-4066-99af-5924c78ddab7,1005,1005, 1000000000@192.168.147.130:5060,sofia/external/ 1000000000@192.168.147.130:5060,73df8735-fee2-464d-aec0-fda886ba2cba 2009-06-26 02:16:07,1245968167,switch_ivr_multi_threaded_bridge,1005,1005,1001,sofia/external/ 1005@192.168.147.1 ,1c2c5f6d-669f-4432-ad04-35a64dbc8a14,1005,1005,sip:1001@192.168.147.1:5060 ;fs_nat=yes,sofia/doublenat5090/sip:1001@192.168.147.1:5060 ;fs_nat=yes,66895f68-70bf-410a-bff7-cda9549c102d 2 total. freeswitch@localhost.localdomain> 2009-06-26 02:18:09 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/ 1005@uat.pbx.starpoundtech.net entering state [calling] 2009-06-26 02:18:10 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1005@uat.pbx.starpoundtech.net entering state [ready] 2009-06-26 02:18:10 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 1 3 IN IP4 192.168.147.1 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.147.1 t=0 0 m=audio 47590 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-06-26 02:19:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1005@uat.pbx.starpoundtech.net entering state [calling] 2009-06-26 02:19:08 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1005@uat.pbx.starpoundtech.net entering state [ready] 2009-06-26 02:19:08 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 1 3 IN IP4 192.168.147.1 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.147.1 t=0 0 m=audio 47590 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Artem On Fri, Jun 26, 2009 at 9:25 PM, Artem Shiyanov <shiyanov@gmail.com> wrote:
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(${dialed_extension}@${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 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