August 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

git fsck
by Sergey Safarov 25 Aug '17

25 Aug '17

Could team developers fix multiple 'author' for commits 487128950df6ee433c131b5feaafe81ee86629f4 and 8574988c3a378b4d5861ecaeb0e958657635703b. Without this to to difficult fork repo and late to track changes on GitHub home$ git fsck Checking object directories: 100% (256/256), done. error in commit 487128950df6ee433c131b5feaafe81ee86629f4: multipleAuthors: invalid format - multiple 'author' lines error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: multipleAuthors: invalid format - multiple 'author' lines Checking objects: 100% (306371/306371), done.

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FS not able to listen on IPv6
by sagar malam 25 Aug '17

25 Aug '17

Hello, I want FS on listen on IPv6 address but facing issue. =======FS CLI output =========== 2017-08-24 17:25:44.786068 [ERR] sofia.c:3146 Error Creating SIP UA for profile: register (sip:mod_sofia@[fe80::e063:e1a0:4d41:6a50]:5070;transport=udp,tcp) ATTEMPT 2 (RETRY IN 5 SEC) 2017-08-24 17:25:49.786024 [ERR] sofia.c:3146 Error Creating SIP UA for profile: register (sip:mod_sofia@[fe80::e063:e1a0:4d41:6a50]:5070;transport=udp,tcp) ATTEMPT 3 (RETRY IN 5 SEC) 2017-08-24 17:25:49.786024 [ERR] sofia.c:3156 Error Creating SIP UA for profile: register (sip:mod_sofia@ [fe80::e063:e1a0:4d41:6a50]:5070;transport=udp,tcp) The likely causes for this are: 1) Another application is already listening on the specified address. 2) The IP the profile is attempting to bind to is not local to this system. ================================================= ==========IFCONFIG output============================ [root@localhost ~]# ifconfig enp0s3: flags=4163<UP,BROADCAST,RUNNING,MULTICAST> mtu 1500 inet 192.168.1.89 netmask 255.255.255.0 broadcast 192.168.1.255 inet6 fe80::e063:e1a0:4d41:6a50 prefixlen 64 scopeid 0x20<link> ether 08:00:27:dc:70:f7 txqueuelen 1000 (Ethernet) RX packets 68251 bytes 9598853 (9.1 MiB) RX errors 0 dropped 28 overruns 0 frame 0 TX packets 46138 bytes 12084174 (11.5 MiB) TX errors 0 dropped 0 overruns 0 carrier 0 collisions 0 ====================================================== There is no other application using 5070 port. Please help me understand issue here.

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XUI Install error
by Andres Gomez 25 Aug '17

25 Aug '17

Hi XUI Guys I saw your presentation in cluecon and it was excellent. Congratulations. I am trying to install xui, but I get to a point that generates error. I am following the instructions of github, and I am installing on debian 8. (https://github.com/seven1240/xui) after this cd db/schema && make make setup I see this error root@debian:/usr/src/xui-master/db/schema# make setup make: *** No rule to make target 'setup'. Stop. Can you help me? Regards

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Building Freeswitch on Pine64+ - AArch64
by William Colls 25 Aug '17

25 Aug '17

Trying to build 1.6.19 on above board, and running into problems with OpenSSL. The problem appears to be related to the implementation of Debian jesse, and I will peruse it with the debain team. In the mean time, what modules do I need to disable so that I can build FreeSWITCH without OpenSSL? Thanks for your time. William.

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Software Echo Canceller for Sangoma E1/T1 Cards
by Nandy Dagondon 24 Aug '17

24 Aug '17

Hi, I encountered echo when connecting FS both Alcatel S12 exchange and NEAX PABX (FS in the middle) using Sangoma A102 (w/o hw echo canceller) using R2 signalling on Wanpipe drivers - even if the call bridges the 2 E1 ports. I can't understand why since this is almost a pass-through connection (didn't add media_bypass=true on the dialplan). My option is to use Oslec but I have to switch to DAHDI-mode. Is there any other recommended echo canceller for Sangoma cards using Wanpipe? Tks, /Nandy

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Issue of access JNDI from java application through mod_java
by Lin, Xiaoping 24 Aug '17

24 Aug '17

I ran into error when trying to bring in a third party package into Java Frame work. It failed in resolving a JNDI name. It works as a stand alone application, but does not work inside java application through mod_java I appreciate any help or input. Many Thanks! The configuration file: <configuration name="java.conf" description="Java Plug-Ins"> <javavm path="/usr/java/jdk1.8.0_31/jre/lib/amd64/server/libjvm.so"/> <options> <option value="-server"/> <option value="-XDebug"/> <option vlaue="-Xrunjdwp:transport=dt_socket,server=y,suspend=n,address=14400"/> <option value="-Dcom.soleo.utils.datafill.DatafillDirectory=/usr/local/XMLNetworkGateway/VoiceService/datafill"/> <option value="-Dcom.sun.management.jmxremote=true" /> <option value="-Dcom.sun.management.jmxremote.port=8000"/> <option value="-Dcom.sun.management.jmxremote.authenticate=false"/> <option value="-Dcom.sun.management.jmxremote.ssl=false"/> <option value="-Dcom.sun.management.jmxremote.local.only=false"/> <option value="-Djava.class.path=$${script_dir}/freeswitch.jar:$${script_dir}/example.jar:/usr/local/freeswitch/java_libs/java_websocket.jar:/usr/local/freeswitch/java_libs/gson-2.8.0.jar:/usr/local/freeswitch/VoiceClient/SessionFactory.jar:/usr/local/XMLNetworkGateway/lib/SRPBaseCommonAPI.jar:/usr/local/XMLNetworkGateway/lib/SoleoDSARSAccess.jar:/usr/local/XMLNetworkGateway/lib/SoleoUtils.jar:/usr/local/XMLNetworkGateway/lib/soleo-counterlib.jar:/usr/local/XMLNetworkGateway/lib/joda-time.jar:/usr/local/XMLNetworkGateway/lib/java-sdk-3.8.0-jar-with-dependencies.jar:/usr/local/XMLNetworkGateway/lib/ListingUtils.jar:/usr/local/XMLNetworkGateway/lib/TTSTextProcessor.jar:/usr/local/XMLNetworkGateway/lib/FSVoiceClient.jar"/> <option value="-Xmx512m"/> <option value="-XX:+HeapDumpOnOutOfMemoryError"/>  <option value="-XX:+UseConcMarkSweepGC -XX:+CMSParallelRemarkEnabled"/> <option value="-XX:+ScavengeBeforeFullGC -XX:+CMSScavengeBeforeRemark"/> </options> <startup class="org/freeswitch/example/ApplicationLauncher" method="startup"/> </configuration> The error from my log Aug 24, 2017 12:35:24.688 com.ibm.watson.developer_cloud.util.CredentialUtils getKeyUsingJNDI FINE: JNDI watson-developer-cloud/speech_to_text/credentials not found. Checking the code It failed at private static String getKeyUsingJNDI(String serviceName) { if (!isClassAvailable("javax.naming.Context") || !isClassAvailable("javax.naming.InitialContext")) { log.info("JNDI string lookups is not available."); return null; } String lookupName = "watson-developer-cloud/" + serviceName + "/credentials"; try { Context context = new InitialContext(); return (String) context.lookup(lookupName); } catch (Exception e) { log.fine("JNDI " + lookupName + " not found."); return null; } }

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Scripts (v8/js)
by Rick Jarvis 24 Aug '17

24 Aug '17

Hello all We’re using scripts now, in particular with the v8 engine. So far so good, but a couple of questions I’m hoping someone can answer: 1. How do we terminate the script early? I’ve tried ‘return;’ but that doesn’t seem to do it? 2. What’s the best way to invoke a delay? ...see the second part of the script below, the standard js ways of waiting generate syntax errors, and using ‘session.execute’ with sleep isn’t fixing it… the problem we’re getting is that either the outgoing message is part of the way through by the time it connects, or if the call is going to a person, it’s possible to answer the call before the audio has established (this might be two separate issues though I guess)... re = session.destination.match(/^555555555$/); if(re) { if(office_open) { session.setVariable("call_timeout", 8); session.setVariable("hangup_after_bridge", true); session.setVariable("continue_on_fail", true); session.execute("export", "alert_info=Bellcore-dr4"); session.execute("db", "insert/call_pickup_uuid/201/" + session.uuid); session.answer(); session.streamFile(“custom/open.wav"); session.execute("bridge", "user/201@" + domain); session.setVariable("call_timeout", 30); session.execute("bridge", "group/mainq@" + domain); } else { session.answer(); await sleep(1000); // tried this setTimeout(function() { // and this session.streamFile(“custom/closed.wav”); // but this has already started session.setVariable("skip_greeting", true); session.setVariable("skip_instructions", true); session.execute("voicemail", "default " + domain + " 201"); }, 1000); } }

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say module with male vocie?
by jungle Boogie 24 Aug '17

24 Aug '17

Hi All, I would like to have the phrase below spoken in a male voice, but when doing so, I receive this error in the fs logs: 2017-08-23 13:36:29.619893 [ERR] mod_say_en.c:627 Unknown Say type=[19] phrase: <action function="say" gender="masculine" data="2600" method="pronounced" type="number"/> If I do it this way: <action function="say" gender="male" data="2600" method="pronounced" type="number"/> It reads it but not in the male voice: 2017-08-23 13:44:11.539900 [DEBUG] switch_ivr_play_say.c:250 Handle say:[2600] (en:en) 2017-08-23 13:44:11.539900 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16@48000hz 1 channels 20ms 2017-08-23 13:44:13.659902 [DEBUG] switch_ivr_play_say.c:1942 done playing file file_string://digits/2.wav!digits/thousand.wav!digits/6.wav!digits/hundred.wav Any help would be appreciated! Thanks, j.b. -- ------- inum: 883510009027723 sip: jungleboogie(a)sip2sip.info

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Re: [Freeswitch-users] Customs fields on Channels table with FS on ODBC in the core.
by Miguel Jesús López Valverde 23 Aug '17

23 Aug '17

Hello, good afternoon: Could anyone offer me any clue about this? Thankyou very much. <http://www.smartic.es/> Miguel J. López Valverde Dpto. Técnico. _____ Pº de la Castellana, 135 7ª pl 28046 Madrid España Tel: 900 900 368 | Móvil: (+34) 667 772 911 E-mail: <mailto:mjlopez@smartic.es> mjlopez(a)smartic.es | Web: <http://www.smartic.es/> http://www.smartic.es _____ <https://www.linkedin.com/company/consulting-smartic-solutions-s.l.> <https://twitter.com/smartic_spain> <https://www.facebook.com/cosutiesmatisolutio?ref=ts> Sus datos de carácter personal (nombre, apellidos, dirección postal y de correo electrónico, etc.) son tratados para la gestión de su relación con la Entidad, así como para el envío de información sobre nuestra actividad y la de terceros relacionadas con la actividad de Consulting Smartic Solutions, S.L., CIF: B85130037, Pº de la Castellana, 135 7ª pl 28046 Madrid. Usted puede ejercer sus derechos de acceso, rectificación, cancelación y oposición dirigiéndose por escrito, con copia de un documento que acredite su identidad, a la dirección info (arroba) smartic.es. Este mensaje puede contener información confidencial. Si usted no es su destinatario, no debe leerlo, copiarlo, distribuirlo, ni hacer uso de la información que contiene. En este caso, por favor, llámenos o comuníquenoslo por escrito y borre este mensaje de su sistema De: Miguel Jesús López Valverde [mailto:mjlopez@smartic.es] Enviado el: martes, 22 de agosto de 2017 14:07 Para: freeswitch-users(a)lists.freeswitch.org Asunto: Customs fields on Channels table with FS on ODBC in the core. Good afternoon: I send this email to transmit a query, in case someone can help me. I was looking for different forums and sources, but I have not found anything about it. I currently have a demo Freeswitch server using ODBC in the core, (as described at https://freeswitch.org/confluence/display/FREESWITCH/Using+ODBC+in+the+core) , and working with a MySQL server As a database engine. This generates tables called Channels and Calls, from which the basic_calls and detailed_calls example views are obtained and, through these tables and views, the information of the calls in progress can be obtained. I wanted to add fields in the Channels table where to add information that comes from channel variables included in the dialplan using XML statements like <variable name = "variable_channel" value = "value" /> or from LUA scripts, in the form session: setVariable ("Variable_channel", "value"). I have noticed that only the variables already contained in the field names are actually updated, as in the case of "accountcode". Is it possible to transmit custom channel variables to added fields in the Channels table? Thank you very much in advance. A cordial greeting. <http://www.smartic.es/> Miguel J. López Valverde Dpto. Técnico. _____ Pº de la Castellana, 135 7ª pl 28046 Madrid España Tel: 900 900 368 | Móvil: (+34) 667 772 911 E-mail: <mailto:mjlopez@smartic.es> mjlopez(a)smartic.es | Web: <http://www.smartic.es/> http://www.smartic.es _____ <https://www.linkedin.com/company/consulting-smartic-solutions-s.l.> <https://twitter.com/smartic_spain> <https://www.facebook.com/cosutiesmatisolutio?ref=ts> Sus datos de carácter personal (nombre, apellidos, dirección postal y de correo electrónico, etc.) son tratados para la gestión de su relación con la Entidad, así como para el envío de información sobre nuestra actividad y la de terceros relacionadas con la actividad de Consulting Smartic Solutions, S.L., CIF: B85130037, Pº de la Castellana, 135 7ª pl 28046 Madrid. Usted puede ejercer sus derechos de acceso, rectificación, cancelación y oposición dirigiéndose por escrito, con copia de un documento que acredite su identidad, a la dirección info (arroba) smartic.es. Este mensaje puede contener información confidencial. Si usted no es su destinatario, no debe leerlo, copiarlo, distribuirlo, ni hacer uso de la información que contiene. En este caso, por favor, llámenos o comuníquenoslo por escrito y borre este mensaje de su sistema <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai gn=sig-email&utm_content=emailclient> Libre de virus. <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai gn=sig-email&utm_content=emailclient> www.avast.com --- El software de antivirus Avast ha analizado este correo electrónico en busca de virus. https://www.avast.com/antivirus

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Pause Voicemail 'beep' when using skip instructions
by Jonathan Hunter 23 Aug '17

23 Aug '17

Hi Guys, Hope all are well? I am using using skip_instructions with the voicemail module, so that we dont have any instructions and just the audio file is played (voicemail_greeting_number=1), which I am doing in dialplan as below; <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="skip_instructions=true"/> <action application="set" data="voicemail_greeting_number=1"/> <action application="voicemail" data="default Domain user"/> This works fine but some people are reporting the 'beep' kicks in too quickly after the audio file has stopped playing. Is there a way to pause/sleep the 'beep' so I can wait for a few seconds before it is generated? Many thanks Jon

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