- FreeSWITCH-users - FreeSWITCH Mailing Lists

GUI
by Ilan Perez 03 Sep '08

03 Sep '08

Anyone developing a GUI for freeswitch Ilan Perez Webmaster 0432 326 017

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Re: [Freeswitch-users] DTMF problem
by Luis Manuel Zuccolo 03 Sep '08

03 Sep '08

Hi, I increase the sleep time and I can see the dtmf digit in cosnole now. How can I capture this digits and send it to voicemail account? An example, please. Thanks in advance Luis Zuccolo ----- Mensaje original ---- De: Michael Collins <mcollins(a)fcnetwork.com> Para: freeswitch-users(a)lists.freeswitch.org Enviado: martes 2 de septiembre de 2008, 14:58:28 Asunto: Re: [Freeswitch-users] DTMF problem Start here: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read There are a few examples. I don’t know if this is the app you need for production but it will definitely let you read a DTMF and thus is sufficient for testing whether or not your system is seeing DTMFs from callers. -MC ________________________________ From:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Luis Manuel Zuccolo Sent: Tuesday, September 02, 2008 10:43 AM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] DTMF problem Hi, Yes, I'm sure that the calls is marking the freeSWITCH extension. In asterisk I can see the digits sended. I'll go to increase the sleep time and see the results. Anyway, how can I use the app read? Would somebody show an example about this? Thanks in advance. Luis Zuccolo ----- Mensaje original ---- De: Michael S Collins < msc(a)freeswitch.org > Para : " freeswitch-users(a)lists.freeswitch.org " < freeswitch-users(a)lists.freeswitch.org > Enviado: martes 2 de septiembre de 2008, 2:09:18 Asunto: Re: [Freeswitch-users] DTMF problem Are you sure that the call is making it to your extension? Also, you sleep for only 250 ms after start_dtmf and then issue a hangup. That's not a whole lot of time to send a digit. Perhaps your test would work better with the dialplan app "read" but that's just a guess on my part. -MC Sent from my iPhone On Sep 1, 2008, at 8:01 PM, "Luis M. Zuccolo" <luismzuccolo(a)yahoo.com.ar> wrote: > Hello, > > I've connected an analog pbx to FreeSWITCH throught a fxo port. > In this setup only need the voicemail functionality of freeSWITCH who > recived a dtmf from a pbx. > To test dtmf app i've created the > extension .../conf/dialplan/extensions/2.xml: > > <include> > <extension name="incoming-fx0-channel-2"> > <condition field="source" expression="mod_openzap"/> > <condition field="destination_number" expression=="^(2)$"> > <action application="answer"/> > <action application="start_dtmf" data="true"/> > <action application="sleep" data="250"/> > <action application="hangup"/> > </condition> > </extension> > </include> > > At console I don't see any reference to dtmf. > > Does anybody have any recommendations about this? > > Thanks in advance > Luis Zuccolo > > __________________________________________________ > Correo Yahoo! > Espacio para todos tus mensajes, antivirus y antispam ¡gratis! > ¡Abrí tu cuenta ya! - http://correo.yahoo.com.ar > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ Yahoo! Cocina Recetas prácticas y comida saludable Visitá http://ar.mujer.yahoo.com/cocina/ Yahoo! Cocina Recetas prácticas y comida saludable http://ar.mujer.yahoo.com/cocina/

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Application playback Requires media! ???
by Birgit Arkesteijn 03 Sep '08

03 Sep '08

Hi all, We've developed an application internally using version 498:8901. After deploying on the clients' system (using version 9288), playback doesn't work any more and the call is bridged straight away. *** I notice the following error in freeswitch.log: 2008-09-03 05:35:00 [DEBUG] switch_core_state_machine.c:140 switch_core_standard_on_execute() sofia/external/18288363200(a)jnctn.net Execute playback(westhawk/Jingle.wav) 2008-09-03 05:35:00 [DEBUG] switch_core_session.c:1049 switch_core_session_execute_application() Application playback Requires media! pre_anyswering channel sofia/external/18288363200(a)jnctn.net *** The dialplan: <document type="freeswitch/xml"> <section name="dialplan" description="FreeSwitch dialplan"> <context name="default"> <extension name="westhawk"> <condition field="destination_number" expression="^8007788821$"> <action application="set" data="call_id=63"/> <action application="playback" data="westhawk/Jingle.wav"/> <action application="bridge" data="sofia/gateway/westhawk/16262435322"/> </condition> </extension> </context> </section> </document> I searched for the error message, but couldn't find anything. The wav file exists. Playback works on our internal system. On our internal system (Suse 10.0, x86_64 GNU/Linux), calls come in via Asterisk over SIP. On the clients' system (Suse 10 enterprise, i386 GNU/Linux ), calls come in via JunctionNetworks over SIP. I have two other scenarios where playback and bridging is done in javascript. This still works, see: <js> var bridge = null; var play_jingle = null; var recordfile = null; function dummy(session, type, data, arg) { } if (session.ready()) { bridge = argv[0]; play_jingle = argv[1]; recordfile = argv[2]; session.answer(); // start recording session.execute("record_session", recordfile) if (play_jingle == "true") { // play 'general' jingle session.streamFile("westhawk/Jingle.wav", dummy); } // play 'record' jingle session.streamFile("westhawk/call_recording_tag.wav", dummy); // bridge the call to the merchant session.execute("bridge", bridge); // hangs up and closes session exit(); } </js> (Oh, before any one asks or if interested, this js was created because I couldn't get recording working for the b-leg by using a dialplan without js. It only recorded the incoming leg, but recorded silence after bridging.) Does anyone know what the "Application playback Requires media!" error message means and how to fix this problem so playback works? Thanks, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

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Remote Phone Routing Update - Now no RTP
by Shawn Lewis 03 Sep '08

03 Sep '08

Ok, I posted previously asking how to figure out the registered endpoint of extensions registering. I used the doublenat.xml sip profile solution, then applied the following the dialplan: <extension name="public_extensions"> <condition field="destination_number" expression="1010"> <action application="set" data="dialed_ext=${destination_number}"/> <action application="set" data="contact=${sofia_contact(doublenat/${dialed_ext}(a)X.X.X.X)}"/> <action application="bridge" data="sofia/doublenat/${contact}"/> </condition> </extension> where X.X.X.X is the EXTERNIP, calls now ring the outside extension, but get no voice/rtp. I have enable the proxy-media setting in my double-nat profile as well. What happens is the internal extension dials external NAT'd extention 2001. sofia passes my extrernal IP and PORT successfully in the SDP of the Invite to the remote extension. I receive my 180-ringing. I receive my 200 OK when answered, in the SDP of the message it is providing the 192.168.1.5 address of the IP telephone. So now FS starts to send media from my IP to 192.168.1.5........ of course which will never reach endpoint registered phone... Here is my doublenat.xml in my sip_profiles, am i missing something?? I would think FS would know not to send there.....  <profile name="doublenat">  <domains> <domain name="doublenat" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="force-register-domain" value=${domain}/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="inbound-late-negotiation" value="true"/>  <param name="NDLB-force-rport" value="true"/> <param name="inbound-proxy-media" value="true"/>  </settings> </profile> ~ Any pointers would be greatly appreciated of course.. Thanks Shawn ----- -- The information contained in this e-mail and any attached documents may be privileged, confidential and protected from disclosure. If you are not the intended recipient you may not read, copy, distribute or use this information. If you have received this communication in error, please notify the sender immediately by replying to this message and then delete it from your system.

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Where is libfreeswitch .dll?
by Lito Manansala 03 Sep '08

03 Sep '08

Hi, Where can i find libfreeswitch .dll files so I can test embedding to my Applications. Thank you

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Sound files/prompts
by Marc Lewis 03 Sep '08

03 Sep '08

I've been working on having a new/alternative set of sound files to the callie files created. One problem that I've run into, though is the amount of leading and trailing silence that each sound file should have. If there is no trailing silence, the next sound in a sequence will run over the currently playing one. In other words, if the following sounds did not have any sort of "padding" sound-b.wav would start playing before sound-a.wav was finished, effectively chopping off a portion of it. <action application="playback" data="sound-a.wav"/> <action application="playback" data="sound-b.wav"/> If there is too much silence in a file, it sounds very choppy. Is there a standard amount of silence that should be added to files or a way to adjust the buffers so that the playback app actually finishes playing a file before it moves to the next one? Thanks. - Marc -- Marc Lewis Avvatel Corporation

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INVITE no SDP
by hemant kumar 03 Sep '08

03 Sep '08

Hi All, I have registered freeswitch to an external gateway as a softphone, and it has registered nicely. Now, i had given extension 1000 to freeswitch. When i try to call from external gateway to freeswitch, it always gives Temporarily Unavailable. However, if i register a phone with freeswitch i am able to dial that extension. Can someone please help me out in this ? Thanks, Hemant

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API info
by Gayatri Kulkarni 03 Sep '08

03 Sep '08

Hi all, Is there a way to find out what all API are accepted through the XML-RPC module and the list of parameters that each API would require? Is there a way to find out the syntax expected by the XML-RPC module for invoking the API I tried a code in Java to find out as follows: <code> getSipServices(XmlRpcClientConfigImpl conf, PrintStream services_file) { try { //create client: XmlRpcClient registryClient = new XmlRpcClient(conf.getServerURL()); String url_str = conf.getServerURL().toString(); //create collection RemoteCollection Switch = new RemoteCollection(registryClient, getUri(url_str)); //actually obtain services: .services = Switch.getServices(); .props = Switch.getProperties(); .props.list(services_file); Enumeration prop_enum = props.elements(); System.out.println("--------------Properties----------------"); while (prop_enum.hasMoreElements()){ System.out.println(prop_enum.nextElement()); } System.out.println("----------------------------------------"); String[] resources = Switch.listResources(); int i = 0; System.out.println("--------------Resources----------------"); while(i < resources.length){ System.out.println(resources[i]); } System.out.println("----------------------------------------"); } catch (Exception e) { ....} return (services); } </code> Unfortunately, the it does not return any properties! for resources, it returns an exception: java.lang.InstantiationError: org.apache.xmlrpc.XmlRpcRequest and services returned are: -----List of available services-------- XPathQueryService CollectionManagementService UserManagementService DatabaseInstanceManager IndexQueryService XUpdateQueryService ValidationService ---------------------------------------- with these services, how do I originate a call from within my application Regards, Gayatri Kulkarni ----- Whenever you find yourself on the side of the majority, it is time to pause and reflect.

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Re: [Freeswitch-users] How to originate a session from outside
by Gayatri Kulkarni 03 Sep '08

03 Sep '08

Guys I made a stupid mistake again guess what - I didn't specify the port number in the URL !!! anyways, the code worked the exception was because I was using an older version of XMLRPC API. Anyways, Adeel you can modify the Java example on the wiki with this one. I think this one's better than what you already have posted there - if you don't mind that is! I don't mean to offend you. Thanks a lot anyways! Regards, Gayatri Kulkarni ----- Whenever you find yourself on the side of the majority, it is time to pause and reflect. From: Gayatri Kulkarni Sent: Tuesday, September 02, 2008 4:29 AM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] How to originate a session from outside Hey Adeel, Remember, originate sofia/internal/1001 &bridge(sofia/internal/1003) had worked the other day? Well, now it isn't working! FS is still saying No_ROUTE_DESTINATION And, it is also giving me a SAXParseException! You want to have a look at the code? Here it is: /** * */ package localcall; import java.lang.String; import java.net.*; import org.apache.xmlrpc.XmlRpcException; import org.apache.xmlrpc.client.XmlRpcClient; import org.apache.xmlrpc.client.XmlRpcClientConfigImpl; /** * @author Gayatri Kulkarni * */ public class LocalCall { /** * */ public LocalCall() { // TODO Auto-generated constructor stub } public XmlRpcClientConfigImpl setConfig(){ XmlRpcClientConfigImpl conf = new XmlRpcClientConfigImpl(); String server = "http://192.168.10.36/RPC2"; String uname = "freeswitch"; String pswd = "works"; try { URL server_url = new URL(server); conf.setServerURL(server_url); } catch (MalformedURLException e){ System.out.println("Cannot generate server URL"); } conf.setBasicPassword(pswd); conf.setBasicUserName(uname); return (conf); } public void placeCall(XmlRpcClientConfigImpl conf){ XmlRpcClient rpcClient = new XmlRpcClient(); String api = "freeswitch.api"; rpcClient.setConfig(conf); Object[] params = new Object[]{ "originate", "sofia/internal/1001 &bridge(sofia/internal/1002)" }; try { rpcClient.execute(api, params ); } catch (XmlRpcException excpn){ System.out.println("Could not execute service. \n"); excpn.printStackTrace(); } } /** * @param args */ public static void main(String[] args) { // TODO Auto-generated method stub LocalCall call = new LocalCall(); call.placeCall(call.setConfig()); } } I have done load mod_xml_rpc on the CLI of FS. Thanks a lot, Regards, Gayatri Kulkarni ----- Whenever you find yourself on the side of the majority, it is time to pause and reflect. From: Adeel Ansari Sent: Wednesday, August 27, 2008 1:20 AM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] How to originate a session from outside Show your program. Further, try the example given in here after appropriate modification. http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC#Java_Example On Wed, Aug 27, 2008 at 3:56 PM, Gayatri Kulkarni <xtpl.gayatri(a)gmail.com> wrote: Thanks a lot! it worked - but why? When I did the same thing through the program it gave me : at RMIServer.main(RMIServer.java:49) Caused by: org.xml.sax.SAXParseException: Unknown type: string at org.apache.xmlrpc.parser.RecursiveTypeParserImpl.startElement(Recursi veTypeParserImpl.java:115) at org.apache.xmlrpc.parser.XmlRpcResponseParser.startElement(XmlRpcResp onseParser.java:126) at com.sun.org.apache.xerces.internal.parsers.AbstractSAXParser.startEle ment(Unknown Source) at com.sun.org.apache.xerces.internal.impl.XMLNSDocumentScannerImpl.scan StartElement(Unknown Source) at com.sun.org.apache.xerces.internal.impl.XMLDocumentFragmentScannerImp l$FragmentContentDriver.next(Unknown Source) at com.sun.org.apache.xerces.internal.impl.XMLDocumentScannerImpl.next(U nknown Source) at com.sun.org.apache.xerces.internal.impl.XMLNSDocumentScannerImpl.next (Unknown Source) at com.sun.org.apache.xerces.internal.impl.XMLDocumentFragmentScannerImp l.scanDocument(Unknown Source) at com.sun.org.apache.xerces.internal.parsers.XML11Configuration.parse(U nknown Source) at com.sun.org.apache.xerces.internal.parsers.XML11Configuration.parse(U nknown Source) at com.sun.org.apache.xerces.internal.parsers.XMLParser.parse(Unknown So urce) at com.sun.org.apache.xerces.internal.parsers.AbstractSAXParser.parse(Un known Source) at com.sun.org.apache.xerces.internal.jaxp.SAXParserImpl$JAXPSAXParser.p arse(Unknown Source) at org.apache.xmlrpc.client.XmlRpcStreamTransport.readResponse(XmlRpcStr eamTransport.java:265) at org.apache.xmlrpc.client.XmlRpcStreamTransport.sendRequest(XmlRpcStre amTransport.java:216) at org.apache.xmlrpc.client.XmlRpcClientWorker.execute(XmlRpcClientWorke r.java:53) at org.apache.xmlrpc.client.XmlRpcClient.execute(XmlRpcClient.java:166) at org.apache.xmlrpc.client.XmlRpcClient.execute(XmlRpcClient.java:136) at org.apache.xmlrpc.client.XmlRpcClient.execute(XmlRpcClient.java:125) at RMIServer.makeACall(RMIServer.java:36) at RMIServer.main(RMIServer.java:49) From: Adeel Ansari Sent: Wednesday, August 27, 2008 12:21 AM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] How to originate a session from outside Try this, originate sofia/internal/1001 &bridge(sofia/internal/1003) Cheers.

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mod_perl won't load
by Nic Wolff 03 Sep '08

03 Sep '08

I'm new to Freeswitch and trying to start a clean install (no flags or options) with a stock modules.conf except mod_perl is enabled (on an EC2 Fedora instance if that matters!) and getting this error on startup: 2008-09-02 13:02:27 [CRIT] switch_loadable_module.c:756 switch_loadable_module_load_file() Error Loading module /usr/local/ freeswitch/mod/mod_perl.so **/usr/local/freeswitch/mod/mod_perl.so: undefined symbol: Perl_Tstack_base_ptr** Has anyone else seen this? Is anyone else running Freeswitch with mod_perl in EC2? Or, generally, what should I do next? Thanks - Nic

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