April 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

play_and_detect_speech not working
by Bemigho Eyimofe 21 Apr '15

21 Apr '15

Hello, I am trying to run play_and_detect_speech for async playing and listening. The console doesn't show any errors but neither a soundfile nor speech will be played to the client. Can anyone confirm that play_and_detect_speech is still working and give a valid example? Here is my Extension: <include> <extension name="play_and_detect_speech example"> <condition field="destination_number" expression="^(414)$"> <action application="answer"/> <action application="set" data="tts_engine=unimrcp"/> <action application="set" data="tts_voice=donna"/> <action application="log" data="INFO PLAY AND DETECT SPEECH"/> <action application="play_and_detect_speech" data="ivr/test.wavdetect:unimrcp {start-input-timers=false,no-input timeout=5000,recognition-timeout=5000}builtin:grammar/boolean?language=en-US;y=1;n=2"/> <action application="log" data="INFO !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!"/> <action application="log" data="CRIT ${detect_speech_result}"/> </condition> </extension> </include> Freeswitch console output: FreeSWITCH Started Max Sessions [1000] Session Rate [30] SQL [Enabled] 2015-04-21 13:45:03.542985 [CONSOLE] switch_core.c:2238 [This app Best viewed at 160x60 or more..] freeswitch@debian-queuex> 2015-04-21 13:45:03.551482 [INFO] switch_time.c:607 Clock synchronized to system time. 2015-04-21 13:45:18.032188 [CONSOLE] mod_voicemail.c:4066 Event Thread Started 2015-04-21 13:45:26.392280 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/queuex_user1(a)10.1.11.115 [e6ed7d96-e81b-11e4-ac47-a729e6b18938] 2015-04-21 13:45:26.412163 [INFO] mod_dialplan_xml.c:635 Processing GDWOMSLK <queuex_user1>->414 in context default 2015-04-21 13:45:26.432225 [INFO] switch_core_media.c:5364 Activating RTCP PORT 4117 2015-04-21 13:45:26.432225 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/queuex_user1(a)10.1.11.115! 2015-04-21 13:45:26.432225 [NOTICE] mod_dptools.c:1268 Channel [sofia/internal/queuex_user1(a)10.1.11.115] has been answered 2015-04-21 13:45:26.432225 [INFO] mod_dptools.c:1638 PLAY AND DETECT SPEECH 2015-04-21 13:45:26.432225 [INFO] mod_unimrcp.c:3129 asr_handle: name = unimrcp, codec = (null), rate = 8000, grammar = (null), param = (null) 2015-04-21 13:45:26.432225 [INFO] mod_unimrcp.c:3131 codec = L16, rate = 8000, dest = (null) 2015-04-21 13:45:26.432225 [NOTICE] mrcp_application.c:96 (ASR-0) Create MRCP Handle 0x837cbc0 [unimrcpserver-mrcp1] 2015-04-21 13:45:26.432225 [INFO] mrcp_client_session.c:133 (ASR-0) Create Channel ASR-0 <new> 2015-04-21 13:45:26.432225 [INFO] mrcp_client_session.c:387 (ASR-0) Receive App Request ASR-0 <new> [2] 2015-04-21 13:45:26.432225 [NOTICE] rtsp_client.c:298 () Create RTSP Handle 0x837ebc8 2015-04-21 13:45:26.432225 [INFO] mrcp_client.c:700 (ASR-0) Add MRCP Handle ASR-0 <new> 2015-04-21 13:45:26.432225 [NOTICE] mrcp_client_session.c:719 (ASR-0) Add Control Channel ASR-0 <new@speechrecog> 2015-04-21 13:45:26.451886 [INFO] mrcp_client_session.c:411 (ASR-0) Send Offer ASR-0 <new> [c:0 a:1 v:0] to 127.0.1.1:1554 2015-04-21 13:45:26.451886 [NOTICE] rtsp_client.c:374 () Established RTSP Connection 127.0.0.1:41506 <-> 127.0.1.1:1554 2015-04-21 13:45:26.451886 [INFO] rtsp_client.c:573 () Add RTSP Handle 0x837ebc8 2015-04-21 13:45:26.451886 [INFO] rtsp_client.c:839 () Send RTSP Data 127.0.0.1:41506 <-> 127.0.1.1:1554 [347 bytes] SETUP rtsp://127.0.1.1:1554/speechrecognizer RTSP/1.0 CSeq: 1 Transport: RTP/AVP;unicast;client_port=4000-4001 Content-Type: application/sdp Content-Length: 179 v=0 o=FreeSWITCH 0 0 IN IP4 127.0.1.1 s=- c=IN IP4 127.0.1.1 t=0 0 m=audio 4000 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 L16/8000 a=sendonly 2015-04-21 13:45:26.451886 [INFO] rtsp_client.c:526 () Push RTSP Request to In-Progress Queue 0x837ebc8 <new> CSeq:1 2015-04-21 13:45:26.451886 [INFO] rtsp_client.c:937 () Receive RTSP Data 127.0.0.1:41506 <-> 127.0.1.1:1554 [312 bytes] RTSP/1.0 200 OK CSeq: 1 Transport: RTP/AVP;unicast;client_port=4000-4001;server_port=5008-5009 Session: e6f516a0e81b11e4 Content-Type: application/sdp Content-Length: 133 v=0 o=UniMRCPServer 0 0 IN IP4 127.0.1.1 s=- c=IN IP4 127.0.1.1 t=0 0 m=audio 5008 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=recvonly 2015-04-21 13:45:26.451886 [INFO] rtsp_client.c:551 () Pop In-Progress RTSP Request 0x837ebc8 CSeq:1 2015-04-21 13:45:26.451886 [INFO] rtsp_client.c:688 () Add RTSP Session 0x837ebc8 <e6f516a0e81b11e4> 2015-04-21 13:45:26.451886 [INFO] mrcp_client_session.c:151 (ASR-0) Receive Answer ASR-0 <e6f516a0e81b11e4> [c:0 a:1 v:0] Status 200 2015-04-21 13:45:26.451886 [INFO] mpf_rtp_stream.c:331 () Enable RTP Session 127.0.1.1:4000 2015-04-21 13:45:26.451886 [INFO] mpf_rtp_stream.c:921 () Open RTP Transmitter 127.0.1.1:4000 -> 127.0.1.1:5008 2015-04-21 13:45:26.451886 [INFO] mpf_bridge.c:93 () Media Path ASR-0 Source->[LPCM/8000/1]->Bridge->[LPCM/8000/1]->Encoder->[PCMU/8000/1]->Sink 2015-04-21 13:45:26.451886 [INFO] mrcp_client_session.c:455 (ASR-0) Raise App Response ASR-0 <e6f516a0e81b11e4> [2] SUCCESS [0] 2015-04-21 13:45:26.451886 [INFO] mrcp_client_session.c:392 (ASR-0) Receive App MRCP Request ASR-0 <e6f516a0e81b11e4> 2015-04-21 13:45:26.451886 [INFO] mrcp_client_session.c:622 (ASR-0) Send MRCP Request ASR-0 <e6f516a0e81b11e4@speechrecog> [1] 2015-04-21 13:45:26.451886 [INFO] rtsp_client.c:839 () Send RTSP Data 127.0.0.1:41506 <-> 127.0.1.1:1554 [378 bytes] ANNOUNCE rtsp://127.0.1.1:1554/speechrecognizer RTSP/1.0 CSeq: 2 Session: e6f516a0e81b11e4 Content-Type: application/mrcp Content-Length: 229 RECOGNIZE 1 MRCP/1.0 Content-Type: text/uri-list Recognition-Timeout: 5000 Recognizer-Start-Timers: false Vendor-Specific-Parameters: no-input timeout=5000 Content-Length: 46 builtin:grammar/boolean?language=en-US;y=1;n=2 2015-04-21 13:45:26.451886 [INFO] rtsp_client.c:526 () Push RTSP Request to In-Progress Queue 0x837ebc8 <e6f516a0e81b11e4> CSeq:2 2015-04-21 13:45:26.451886 [INFO] rtsp_client.c:937 () Receive RTSP Data 127.0.0.1:41506 <-> 127.0.1.1:1554 [137 bytes] RTSP/1.0 200 OK CSeq: 2 Session: e6f516a0e81b11e4 Content-Type: application/mrcp Content-Length: 30 MRCP/1.0 1 200 IN-PROGRESS 2015-04-21 13:45:26.451886 [INFO] rtsp_client.c:551 () Pop In-Progress RTSP Request 0x837ebc8 CSeq:2 2015-04-21 13:45:26.451886 [INFO] mrcp_client_session.c:500 (ASR-0) Raise App MRCP Response ASR-0 <e6f516a0e81b11e4> 2015-04-21 13:45:26.451886 [WARNING] switch_core_file.c:230 File has 2 channels, muxing to 1 channel will occur. 2015-04-21 13:45:26.531882 [WARNING] sofia.c:5692 Ping succeeded sipgate with code 200 - count 1/1/1, state UP 2015-04-21 13:45:26.872152 [INFO] rtsp_client.c:937 () Receive RTSP Data 127.0.0.1:41506 <-> 127.0.1.1:1554 [190 bytes] ANNOUNCE rtsp://127.0.1.1:1554/speechrecognizer RTSP/1.0 CSeq: 2 Session: e6f516a0e81b11e4 Content-Type: application/mrcp Content-Length: 42 START-OF-SPEECH 1 IN-PROGRESS MRCP/1.0 2015-04-21 13:45:26.872152 [INFO] rtsp_client.c:839 () Send RTSP Data 127.0.0.1:41506 <-> 127.0.1.1:1554 [55 bytes] RTSP/1.0 200 OK CSeq: 2 Session: e6f516a0e81b11e4 2015-04-21 13:45:26.872152 [INFO] mrcp_client_session.c:516 (ASR-0) Raise App MRCP Event ASR-0 <e6f516a0e81b11e4> 2015-04-21 13:45:26.891692 [INFO] switch_ivr_async.c:3884 (sofia/internal/queuex_user1(a)10.1.11.115) START OF SPEECH 2015-04-21 13:45:26.891692 [INFO] switch_ivr_async.c:3943 (sofia/internal/queuex_user1(a)10.1.11.115) WAITING FOR RESULT 2015-04-21 13:45:35.692016 [NOTICE] switch_rtp.c:5185 Receiving an RTCP packet[2015-21-03 11:45:35.700029] SSRC[723361390]RTT[0.013550] A[3068113717] - DLSR[273063] - LSR[3067839766] 2015-04-21 13:45:40.312178 [NOTICE] switch_rtp.c:5185 Receiving an RTCP packet[2015-21-03 11:45:40.320207] SSRC[723361390]RTT[0.013275] A[3068416505] - DLSR[248184] - LSR[3068167451] 2015-04-21 13:45:44.932184 [NOTICE] switch_rtp.c:5185 Receiving an RTCP packet[2015-21-03 11:45:44.939794] SSRC[723361390]RTT[0.012268] A[3068719254] - DLSR[223320] - LSR[3068495130] 2015-04-21 13:45:49.551974 [NOTICE] switch_rtp.c:5185 Receiving an RTCP packet[2015-21-03 11:45:49.559918] SSRC[723361390]RTT[0.010956] A[3069022038] - DLSR[198496] - LSR[3068822824] 2015-04-21 13:45:54.171909 [NOTICE] switch_rtp.c:5185 Receiving an RTCP packet[2015-21-03 11:45:54.180123] SSRC[723361390]RTT[0.010818] A[3069324828] - DLSR[173622] - LSR[3069150497] 2015-04-21 13:45:58.791859 [NOTICE] switch_rtp.c:5185 Receiving an RTCP packet[2015-21-03 11:45:58.800101] SSRC[723361390]RTT[0.010559] A[3069627603] - DLSR[148737] - LSR[3069478174] 2015-04-21 13:46:03.411703 [NOTICE] switch_rtp.c:5185 Receiving an RTCP packet[2015-21-03 11:46:03.420106] SSRC[723361390]RTT[0.010025] A[3069930380] - DLSR[123880] - LSR[3069805843] 2015-04-21 13:46:08.031655 [NOTICE] switch_rtp.c:5185 Receiving an RTCP packet[2015-21-03 11:46:08.39750] SSRC[723361390]RTT[0.008957] A[3070233133] - DLSR[99038] - LSR[3070133508] 2015-04-21 13:46:12.651812 [NOTICE] switch_rtp.c:5185 Receiving an RTCP packet[2015-21-03 11:46:12.660125] SSRC[723361390]RTT[0.008652] A[3070535933] - DLSR[74161] - LSR[3070461205] 2015-04-21 13:46:17.271783 [NOTICE] switch_rtp.c:5185 Receiving an RTCP packet[2015-21-03 11:46:17.280116] SSRC[723361390]RTT[0.008118] A[3070838709] - DLSR[49271] - LSR[3070788906] 2015-04-21 13:46:21.891722 [NOTICE] switch_rtp.c:5185 Receiving an RTCP packet[2015-21-03 11:46:21.900002] SSRC[723361390]RTT[0.007355] A[3071141478] - DLSR[24416] - LSR[3071116580] 2015-04-21 13:46:26.511739 [NOTICE] switch_rtp.c:5185 Receiving an RTCP packet[2015-21-03 11:46:26.520157] SSRC[723361390]RTT[0.006851] A[3071444265] - DLSR[327236] - LSR[3071116580] 2015-04-21 13:46:31.132034 [NOTICE] switch_rtp.c:5185 Receiving an RTCP packet[2015-21-03 11:46:31.139830] SSRC[723361390]RTT[0.005707] A[3071747019] - DLSR[302385] - LSR[3071444260] 2015-04-21 13:46:34.811605 [NOTICE] sofia.c:952 Hangup sofia/internal/queuex_user1(a)10.1.11.115 [CS_EXECUTE] [NORMAL_CLEARING] 2015-04-21 13:46:34.811605 [INFO] mrcp_client_session.c:392 (ASR-0) Receive App MRCP Request ASR-0 <e6f516a0e81b11e4> 2015-04-21 13:46:34.811605 [INFO] mrcp_client_session.c:622 (ASR-0) Send MRCP Request ASR-0 <e6f516a0e81b11e4@speechrecog> [2] 2015-04-21 13:46:34.811605 [INFO] rtsp_client.c:839 () Send RTSP Data 127.0.0.1:41506 <-> 127.0.1.1:1554 [167 bytes] ANNOUNCE rtsp://127.0.1.1:1554/speechrecognizer RTSP/1.0 CSeq: 3 Session: e6f516a0e81b11e4 Content-Type: application/mrcp Content-Length: 19 STOP 2 MRCP/1.0 2015-04-21 13:46:34.811605 [INFO] rtsp_client.c:526 () Push RTSP Request to In-Progress Queue 0x837ebc8 <e6f516a0e81b11e4> CSeq:3 2015-04-21 13:46:34.811605 [INFO] rtsp_client.c:937 () Receive RTSP Data 127.0.0.1:41506 <-> 127.0.1.1:1554 [170 bytes] RTSP/1.0 200 OK CSeq: 3 Session: e6f516a0e81b11e4 Content-Type: application/mrcp Content-Length: 63 MRCP/1.0 2 200 COMPLETE Active-Request-Id-List: 1 2015-04-21 13:46:34.811605 [INFO] rtsp_client.c:551 () Pop In-Progress RTSP Request 0x837ebc8 CSeq:3 2015-04-21 13:46:34.811605 [INFO] mrcp_client_session.c:500 (ASR-0) Raise App MRCP Response ASR-0 <e6f516a0e81b11e4> 2015-04-21 13:46:34.811605 [INFO] mrcp_client_session.c:387 (ASR-0) Receive App Request ASR-0 <e6f516a0e81b11e4> [1] 2015-04-21 13:46:34.811605 [INFO] mrcp_client_session.c:833 (ASR-0) Terminate Session ASR-0 <e6f516a0e81b11e4> 2015-04-21 13:46:34.811605 [INFO] rtsp_client.c:839 () Send RTSP Data 127.0.0.1:41506 <-> 127.0.1.1:1554 [96 bytes] TEARDOWN rtsp://127.0.1.1:1554/speechrecognizer RTSP/1.0 CSeq: 4 Session: e6f516a0e81b11e4 2015-04-21 13:46:34.811605 [INFO] rtsp_client.c:526 () Push RTSP Request to In-Progress Queue 0x837ebc8 <e6f516a0e81b11e4> CSeq:4 2015-04-21 13:46:34.831763 [INFO] mpf_rtp_stream.c:935 () Close RTP Transmitter 127.0.1.1:4000 -> 127.0.1.1:5008 [s:3418 o:546880] 2015-04-21 13:46:34.831763 [INFO] mpf_rtp_stream.c:418 () Remove RTP Session 127.0.1.1:4000 2015-04-21 13:46:34.831763 [INFO] rtsp_client.c:937 () Receive RTSP Data 127.0.0.1:41506 <-> 127.0.1.1:1554 [55 bytes] RTSP/1.0 200 OK CSeq: 4 Session: e6f516a0e81b11e4 2015-04-21 13:46:34.831763 [INFO] rtsp_client.c:551 () Pop In-Progress RTSP Request 0x837ebc8 CSeq:4 2015-04-21 13:46:34.831763 [INFO] rtsp_client.c:706 () Remove RTSP Session 0x837ebc8 <e6f516a0e81b11e4> 2015-04-21 13:46:34.831763 [INFO] rtsp_client.c:442 () Remove RTSP Handle 0x837ebc8 2015-04-21 13:46:34.831763 [NOTICE] rtsp_client.c:305 () Destroy RTSP Handle 0x837ebc8 2015-04-21 13:46:34.831763 [INFO] rtsp_client.c:382 () Close RTSP Connection 127.0.0.1:41506 <-> 127.0.1.1:1554 2015-04-21 13:46:34.831763 [NOTICE] rtsp_client.c:432 () Destroy RTSP Connection 127.0.0.1:41506 <-> 127.0.1.1:1554 2015-04-21 13:46:34.831763 [INFO] mrcp_client_session.c:209 (ASR-0) Session Terminated ASR-0 <e6f516a0e81b11e4> 2015-04-21 13:46:34.831763 [INFO] mrcp_client.c:710 (ASR-0) Remove MRCP Handle ASR-0 <e6f516a0e81b11e4> 2015-04-21 13:46:34.831763 [INFO] mrcp_client_session.c:455 (ASR-0) Raise App Response ASR-0 <e6f516a0e81b11e4> [1] SUCCESS [0] 2015-04-21 13:46:34.831763 [NOTICE] mrcp_application.c:182 (ASR-0) Destroy MRCP Handle ASR-0 2015-04-21 13:46:34.831763 [NOTICE] switch_core_session.c:1641 Session 1 (sofia/internal/queuex_user1(a)10.1.11.115) Ended 2015-04-21 13:46:34.831763 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/queuex_user1(a)10.1.11.115 [CS_DESTROY] Freundliche Grüße, Bemigho Eyimofe Softwareentwickler . Netempire AG | Lüghauser Straße 16 | D - 51503 Rösrath . Fon: 0221 99 11 12 - 22 | Fax: 0221 99 11 12 - 99 . mailto:bee@netempire.de <mailto:bee@netempire.de> | http://www.netempire.de <http://www.netempire.de/> __________________________________________________________ . Vorstandsvorsitzender: Andreas Nettesheim . Vorstand: Dr. Guido Gimmler | Johannes Krause | Kim Bauer . Vorsitzender des Aufsichtsrates: Dr. Dierk Wehrmeister . Handelsregister: Amtsgericht Köln, HRB 35407 . USt-IdNr.: DE 215404683 __________________________________________________________

1 0

mod_xml_radius is not creating the variables in the directory
by Густаво Силва 21 Apr '15

21 Apr '15

I am not able to create the variables with the attribute values replied from radius in the directory. The directory is being created as bellow: <document type="freeswitch/xml"> <section name="directory"> <domain name="xxxxxxxx"> <user id="1008"> <variables> <variable name="radius_auth_result" value="0"></variable> </variables> </user> </domain> </section> </document> My radius configuration is like: 1008 Cleartext-Password := "1234" Acct-Interim-Interval := "30", h323-call-origin := "Answer from 1008", Freeswitch-AVPair := "315" What should I have to do to get the values from the attributes replied from radius server? Regards, Gustavo

1 0

How to receive call using gsmopen
by Anurag Rana 21 Apr '15

21 Apr '15

Hi All, I have micromax usb dongle with a gsm sim in it attached to my system. mod_gsmopen has been compiled and has been loaded. Freeswitch is running. *This is the output of lsusb command -* ---------------------------------------------------------- Bus 002 Device 003: ID 064e:e263 Suyin Corp. Bus 002 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 001 Device 006: ID 1c9e:9605 OMEGA TECHNOLOGY Bus 001 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 003 Device 011: ID 15d9:0a4e Trust International B.V. Bus 003 Device 010: ID 1a2c:0c21 China Resource Semico Co., Ltd Bus 003 Device 009: ID 1a40:0101 Terminus Technology Inc. 4-Port HUB Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub --------------------------------------------------------------------------------------- *This is the output of dmesg | grep tty command* ---------------------------------------------------------------------- [ 8114.018211] usb 1-1.2: GSM modem (1-port) converter now attached to ttyUSB0 [ 8114.018840] usb 1-1.2: GSM modem (1-port) converter now attached to ttyUSB1 [ 8114.019458] usb 1-1.2: GSM modem (1-port) converter now attached to ttyUSB2 [ 8114.019961] usb 1-1.2: GSM modem (1-port) converter now attached to ttyUSB3 --------------------------------------------------------------------- *This is the content of gsmopen.conf.xml file* *-----*------------------------------------------------------------ <configuration name="gsmopen.conf" description="GSMopen Configuration"> <global_settings> <param name="debug" value="8"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="hold-music" value="$${moh_uri}"/> <param name="destination" value="5000"/> </global_settings>  <per_interface_settings> <interface id="1" name="gsm01"> <param name="controldevice_name" value="/dev/ttyUSB3"/> <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>      </interface>  </per_interface_settings> </configuration> ------------------------------------------------------------------------ I searched internet for "how to receive a call using FS" or "How to make an inbound call using gsmopen" but could not find detailed stepwise explanation. Can someone please tell me what I need to do next so that when I call the sim number in dongle, call is received by FS and actions are performed based on dialplan. OR direct me to some link which explains this. (apart from this https://freeswitch.org/confluence/display/FREESWITCH/mod_gsmopen) Thanks Anurag Rana <https://sites.google.com/site/homepagerana/> M.Tech CSE, IIIT-Delhi, https://sites.google.com/site/homepagerana/

4 7

loglevel per caller_id
by Sergey Safarov 21 Apr '15

21 Apr '15

Whether there is a possibility of default logging level to expose NOTICE, and definitely calls from one caller_id set to debug?

4 5

What's wrong with Fax ECM and T.38?
by Victor Chukalovskiy 21 Apr '15

21 Apr '15

Good day, My appologies as this is not specific to FS per say. I'm looking if someone has feedback on a broader topic. I noticed that when T.30 DIS / DCS signals between fax machines negotiate error correction mode (ECM) it often causes issues to various T.38 implementations. I also noticed that certain soft-switches or media gateway would intercept DIS signal and overwrite ECM bit to "false". Thus making far-end fax machine believe call should proceed as non-ECM. In FS there is a channel variable for it. Some Cisco, Metaswitch gear do that as well. Would someone know what is it about ECM Fax mode that makes it less "t.38 friendly"? What forces vendors to implement tricks to prevent Fax machines from negotiating ECM? Many thanks, -Victor

2 1

Dynamic local_stream?
by Ali Pey 21 Apr '15

21 Apr '15

Hello, Is it possible to dynamically set different local_stream files for different users/bridges? For instance I need a different custom MOH for call1 than call2 and they dynamically change as users can upload their own custom moh and I can't change the local_stream.conf.xml config file. Thanks, Ali Pey

4 4

mod_conference CDR format
by Free Switcher 20 Apr '15

20 Apr '15

Hello FreeSwitchers, Does anyone know if it is possible to specify CDR format for conferences? mod_conference outputs XML CDRs by default and I'd like it in CSV format. I can't seem to find any way to specify format anywhere. Thanks for any tips. Regards, Andy

2 1

FreeSWITCH Week in Review (Master Branch) April 11th-17th
by Ken Rice 20 Apr '15

20 Apr '15

New Post on freeswitch.org from Kathleen check it out at http://ift.tt/1JnvNj3 FreeSWITCH Week in Review (Master Branch) April 11th-17th Hello, again. This passed week in the FreeSWITCH master branch we had 14 commits. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. Improvements in build system, cross platform support, and packaging: FS-7060 Added amqp.conf.xml to freeswitch.spec to fix packaging FS-7428 Freeswitch.spec needs to copy mongo driver tarball to libs so that packaging doesn’t fetch from files.freeswitch.org for every build. The following bugs were squashed: FS-7407 Fixed a sip presence issue where the sip proxy changes the transport between the phone and freeswitch and then freeswitch replies back to the proxy over the wrong transport FS-7431 Fixed the parsing of silence streams to be less forgiving of syntax errors FS-7434 Reset jitter buffer when SSRC changes FS-7440 Fix bug in SDP parser, when video a= lines are above the audio a= lines, that could lead to a crash

1 0

skypopen call disconnection after some time ?
by Shabbir abbasi 20 Apr '15

20 Apr '15

this call was disconnected dont know why ?? 2015-04-20 00:27:50.233170 [DEBUG] skypopen_protocol.c:207 [1.4.6~32bit ] [DEBUG_SKYPE 207 ][skype105 ][UP,INPROGRS] READING: |||CALL 5005997 DURATION 602||| skypopen_protocol.c:1082 [1.4.6~32bit ] [DEBUG_SKYPE 1082 ][skype105 ][UP,INPROGRS] CLOSED skypopen_protocol.c:1097 [1.4.6~32bit ] [DEBUG_SKYPE 1097 ][skype105 ][UP,INPROGRS] Skype incoming audio GONE [skype105 ][UP,INPROGRS] len=-1, error: Broken pipe [skype105 ][UP,INC_HNG] Skype outbound audio GONE [skype105 ][UP,INC_HNG] outbound audio server (I am it) EXITING [skype105 ][UP,INC_HNG] incoming audio (read) server (I am it) EXITING [DEBUG_SKYPE 207 ][skype105 ][UP,INC_HNG] READING: |||CALL 5005997 DURATION 603||| [DEBUG_SKYPE 1441 ][skype105 ][UP,INC_HNG] skype call ended any idea ???

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P-asserted-identity in Responses
by Tomáš Boros 20 Apr '15

20 Apr '15

Hi, Is it possible to to enable P-Asserted-Identity in the 18x/200 Responses? I need to copy info from P-asserted identity from Leg-B back to Leg-A. So its like: Invite comes to my freeswitch and the call is bridged to Leg-B. Leg-B answers the call and places a P-Asserted-Identity header into the 200 OK response 200 OK response on the Leg-A should containt P-Asserted-Identity of Leg-B (callee). Thank you, -- Tomáš Boros

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