- FreeSWITCH-users - FreeSWITCH Mailing Lists

patch for sofia_sip_i_invite to support replaces with action
by Luis Azedo 22 Apr '15

22 Apr '15

Hi, i need a inline custom action passed with an INVITE msg with a replaces header or a better to use this scenario. right now, for calls with replaces header there is support for sla, intercept and attendant transfer. my problem is that intercept requires the bridge_uuid which is not set on one leg calls (on hold playing music) so the intercept condition won't be hit. the attendant transfer takes app/data applies and then hangs up the call. i need a simple answer,intercept:(sip->sip_replaces->rp_call_id) instead of bridge_uuid. i tried to set signal_bond and originate_signal_bound on the channel to see if i could bypass this, but with no luck. i created a small patch that i am attaching here with the hope that FreeSWITCH will look and possibly merge it. Best

4 3

att_xfer + leg A
by Денис 22 Apr '15

22 Apr '15

How to make att_xfer from leg A? It makes for B only...

1 0

ClueCon Weekly Today at 1PM EST!
by Ken Rice 22 Apr '15

22 Apr '15

Greetings FreeSWITCHers! Today Wed, April 22nd @ 1PM EST Join us at https://conference.freeswitch.org/ for ClueCon Weekly. Click the ³Call 888² button to join us. This week Chris Rienze (@crienzo), author of mod_rayo, and regular contributor to FreeSWITCH, will be showing off some good things with lua, graylog, and pcapsipdump. Also have your Headset and WebCams ready and use Chrome when calling 888 to get in on the Video side of the conf! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

1 0

Challenges
by Tony Mendez 22 Apr '15

22 Apr '15

Hi all, I have installed the latest version of freeswitch, with fusionpbx. It is an upgraded install. However, some of the extensions do not generate a ring tone when calling in. Specifically, it is one extension and one ring group. Everything else is fine ( I think). Would be good if someone can help with 1) direct me to the relevant sections in the freeswitch manual, or 2) just tell me how to make it ring. In fusionpbx, it is set to “default” for that specific ring group, same as the other ring groups. Gracias, Tony

5 7

Duplicate request from FreeSWITCH in the radius server
by Густаво Силва 22 Apr '15

22 Apr '15

I am getting duplicate request from my freeswitch: Info: rlm_radutmp: Login entry for NAS localhost port 0 duplicate I am using freeswitch and freeradius in the same machine. The radius is authenticating the request but why the freeswitch is sending twice? Is something in the radiusclient that I shoud change? Regards, Gustavo

1 0

Trunk Calls
by Ali Jibran 22 Apr '15

22 Apr '15

Hi all. I have voipfone trunked to freeswitch. Everything works fine. But now I've bought another number from the same provider. The call comes in but executes the same IVR at 1234. How do I change the dialplan that the call executes different IVR depending on the number dialed. here is the gateway conf: <include> <gateway name="voipfone"> <param name="username" value="XXXXX"/> <param name="realm" value="sip.voipfone.net"/> <param name="password" value="XXXXX"/> <param name="extension" value="1234"/> <param name="expire-seconds" value="60"/> </gateway> </include>

3 5

Unable to load mod_java.
by John 21 Apr '15

21 Apr '15

Hi All I have built mod_java and have mod_java.so in /usr/local/freeswitch/mod/ but when I am trying to load mod_java from freeswitch terminal like : load mod_java getting following error : 2015-02-07 15:01:13.175331 [INFO] mod_enum.c:880 ENUM Reloaded 2015-02-07 15:01:13.175331 [NOTICE] modjava.c:366 Java Framework Loading... 2015-02-07 15:01:13.175331 [INFO] switch_time.c:1411 Timezone reloaded 1781 definitions +OK Reloading XML -ERR [module load file routine returned an error] 2015-02-07 15:01:13.215310 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** I am using OpenJDK version : java version "1.6.0_33" OpenJDK Runtime Environment (IcedTea6 1.13.5) (6b33-1.13.5-1ubuntu0.14.04) OpenJDK 64-Bit Server VM (build 23.25-b01, mixed mode) and my conf/autoload_configs/java. conf.xml reads as : <configuration name="java.conf" description="Java Plug-Ins"> <javavm path="/usr/lib/jvm/java-6- openjdk-amd64/jre/lib/amd64/jamvm/libjvm.so"/> <options> <option value="-Djava.class.path=/usr/local/freeswitch/scripts/freeswitch.jar"/> <option value="-agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address=0.0.0.0:8000"/> </options> <startup class="org/freeswitch/example/ApplicationLauncher" method="startup"/> </configuration> Can someone help me to resolve this error? I tried googling this issue and accordingly made above config changes. Thanks, John

1 1

perl ESL outbound socket, no DTMF if not async ???
by Giovanni Maruzzelli 21 Apr '15

21 Apr '15

Ciao FreeSWITCHers, on FS 1.4.18: using perl ESL outbound socket, I see FS receives DTMF 2833 only when socket is declared "async" in dialplan. NB: not only the outbound socket does not receives DTMF events, is FreeSWITCH itself that does not receives DTMFs. Is this correct? Is this the expected behavior? If I delete "async" from dialplan, DTMFs are not received es: <action application="socket" data="127.0.0.1:8040 async"/> works, while with <action application="socket" data="127.0.0.1:8040"/> no DTMFs at all. ==== btw, this is an example of outbound_socket in perl ESL #!/usr/bin/perl require ESL; use IO::Socket::INET; my $ip = "127.0.0.1"; my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8040', Proto => 'tcp', Listen => 1, Reuse => 1 ); die "Could not create socket: $!\n" unless $sock; for(;;) { my $new_sock = $sock->accept(); my $pid = fork(); if ($pid) { print "New child pid $pid created...\n"; close($new_sock); next; } my $fd = fileno($new_sock); my $con = new ESL::ESLconnection($fd); my $info = $con->getInfo(); my $uuid = $info->getHeader("unique-id"); printf "Connected call %s, from %s\n", $uuid, $info->getHeader("caller-caller-id-number"); $con->sendRecv("myevents $uuid"); $con->setEventLock("1"); # need this for forcing execution order on async, has no effect if no "async" in dialplan $con->execute("sleep","500"); $con->execute("answer"); $con->execute("sleep","500"); $con->execute("playback", "ivr/ivr-welcome_to_freeswitch.wav"); $con->execute("sleep","500"); $con->execute("playback", "ivr/ivr-finished_pound_hash_key.wav"); while($con->connected()) { my $e = $con->recvEvent(); if ($e) { my $name = $e->getHeader("event-name"); print "EVENT [$name]\n"; if ($name eq "DTMF") { my $digit = $e->getHeader("dtmf-digit"); my $duration = $e->getHeader("dtmf-duration"); print "DTMF digit $digit ($duration)\n"; $con->execute("hangup"); } } } print "BYE\n"; close($new_sock); } -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

1 0

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 . 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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

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