Updates:<br>1. One-way audio is in 95% tries. But how the rest 5% works??<br>2. Strange FS logging after the channels are bridged (user A talk to user B)<br><br><br>2009-06-26 02:16:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/<a href="mailto:1005@192.168.147.1">1005@192.168.147.1</a> entering state [ready]<br>
2009-06-26 02:16:07 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP:<br>v=0<br>o=FreeSWITCH 3898736844884231 105343281763004076 IN IP4 192.168.147.130<br>s=FreeSWITCH<br>c=IN IP4 192.168.147.130<br>t=0 0<br>m=audio 31134 RTP/AVP 0 101 13<br>
a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=rtpmap:13 CN/8000<br>a=ptime:20<br>m=video 0 RTP/AVP 34<br>a=rtpmap:34 H263/90000<br><br>2009-06-26 02:16:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/<a href="http://1000000000@192.168.147.130:5060">1000000000@192.168.147.130:5060</a> entering state [ready]<br>
freeswitch@localhost.localdomain> 2009-06-26 02:17:09 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/<a href="mailto:1005@uat.pbx.starpoundtech.net">1005@uat.pbx.starpoundtech.net</a> entering state [calling]<br>
2009-06-26 02:17:09 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/<a href="mailto:1005@uat.pbx.starpoundtech.net">1005@uat.pbx.starpoundtech.net</a> entering state [ready]<br>2009-06-26 02:17:09 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP:<br>
v=0<br>o=- 1 3 IN IP4 192.168.147.1<br>s=CounterPath X-Lite 3.0<br>c=IN IP4 192.168.147.1<br>t=0 0<br>m=audio 47590 RTP/AVP 0 101<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br><br><br>freeswitch@localhost.localdomain> show calls<br>
<br>API CALL [show(calls)] output:<br>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<br>
2009-06-26 02:16:05,1245968165,switch_ivr_multi_threaded_bridge,1005,1005,inbound1000000000,sofia/external/<a href="mailto:1005@uat.pbx.starpoundtech.net">1005@uat.pbx.starpoundtech.net</a>,4fa86434-b542-4066-99af-5924c78ddab7,1005,1005,<a href="http://1000000000@192.168.147.130:5060">1000000000@192.168.147.130:5060</a>,sofia/external/<a href="http://1000000000@192.168.147.130:5060">1000000000@192.168.147.130:5060</a>,73df8735-fee2-464d-aec0-fda886ba2cba<br>
2009-06-26 02:16:07,1245968167,switch_ivr_multi_threaded_bridge,1005,1005,1001,sofia/external/<a href="mailto:1005@192.168.147.1">1005@192.168.147.1</a>,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<br>
<br>2 total.<br><br>freeswitch@localhost.localdomain> 2009-06-26 02:18:09 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/<a href="mailto:1005@uat.pbx.starpoundtech.net">1005@uat.pbx.starpoundtech.net</a> entering state [calling]<br>
2009-06-26 02:18:10 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/<a href="mailto:1005@uat.pbx.starpoundtech.net">1005@uat.pbx.starpoundtech.net</a> entering state [ready]<br>2009-06-26 02:18:10 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP:<br>
v=0<br>o=- 1 3 IN IP4 192.168.147.1<br>s=CounterPath X-Lite 3.0<br>c=IN IP4 192.168.147.1<br>t=0 0<br>m=audio 47590 RTP/AVP 0 101<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br><br>2009-06-26 02:19:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/<a href="mailto:1005@uat.pbx.starpoundtech.net">1005@uat.pbx.starpoundtech.net</a> entering state [calling]<br>
2009-06-26 02:19:08 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/<a href="mailto:1005@uat.pbx.starpoundtech.net">1005@uat.pbx.starpoundtech.net</a> entering state [ready]<br>2009-06-26 02:19:08 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP:<br>
v=0<br>o=- 1 3 IN IP4 192.168.147.1<br>s=CounterPath X-Lite 3.0<br>c=IN IP4 192.168.147.1<br>t=0 0<br>m=audio 47590 RTP/AVP 0 101<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br><br><br><br><br>Artem<br><br><br>
<br><br><br><br><br><div class="gmail_quote">On Fri, Jun 26, 2009 at 9:25 PM, Artem Shiyanov <span dir="ltr"><<a href="mailto:shiyanov@gmail.com">shiyanov@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hello!<br><br>I got a problem with one way audio, symptoms are:<br>firstly play audio file to channel A (A is hears sound)<br>secondly bridge channel B with A (A doesn't hear B).<br><br>Environment:<br>- no NAT<br>- 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<br>
- dialplan:<br><extension name="playback_media_file"><br> <condition field="destination_number" expression="playmedia"><br> <action application="answer"/><br>
<action application="playback" data="test.wav"/><br> </condition><br> </extension><br><br><extension name="Local_Extension_from_SP"><br> <condition field="destination_number" expression="^([0-9]{2,9})$"><br>
<action application="set" data="dialed_extension=$1"/><br> <action application="export" data="dialed_extension=$1"/><br> </condition><br> <condition field="${sip_to_host}" expression="^([^.]*)\..*$"><br>
<action application="set" data="orgname=$1"/><br> </condition><br> <condition field="destination_number" expression="^${caller_id_number}$"><br>
<anti-action application="set" data="ringback=${us-ring}"/><br>
<anti-action application="set" data="transfer_ringback=${us-ring}"/><br> <anti-action application="set" data="call_timeout=10"/><br> <anti-action application="set" data="hangup_after_bridge=true"/><br>
<anti-action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/>
<br> <anti-action application="set" data="continue_on_fail=true"/><br> <anti-action application="db" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/><br>
<anti-action application="db" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/><br>
<anti-action application="set"
data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}"/><br> <anti-action application="db" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/><br> <anti-action application="bridge" data="user/${dialed_extension}@${domain_name}"/><br>
<anti-action application="answer"/><br>
<anti-action application="export"
data="sip_h_X-SPFrom=&quote;${sip_from_user}&quote;&lt;${sip_from_uri}&gt;"/><br> <anti-action application="export" data="sip_h_X-SPTo=&lt;${sip_to_uri}&gt;"/><br>
<anti-action application="export" data="sip_h_X-SPCallId=${sip_call_id}"/><br> <anti-action application="bridge" data="sofia/external/${orgname}send2voicemail@$${starpound_sip_app_server}"/><br>
</condition><br> </extension> <br>- Call routing scheme:<br>user calls to FS, FS calls to my B2BUA which manage call with SIP 3pcc<br>Exact description what's going on is:<br>user A -> FS -(bridge)-> my B2BUA<br>
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.<br>
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.<br><br><br>What I've tried:<br>- set parameter "inbound-proxy-media" to "true" in Sofia profile<br>
- set parameter "disable_rtp_auto_adjust to "true" in Sofia profile<br>Nothing helps.<br><br><br>Any help or thoughts would be MUCH appreciated!<br><font color="#888888">Artem<br><br>
</font></blockquote></div><br>