September 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] How to originate a session from outside
by Boris Krivonog 02 Sep '08

02 Sep '08

The easiest way would be using XML-RPC (http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC) with Apache XML-RPC (http://ws.apache.org/xmlrpc/client.html) libraries the code would look something like: XmlRpcClientConfigImpl config = new XmlRpcClientConfigImpl(); config.setServerURL( new URL("http://<freeswitchip>:8080/RPC2") ); config.setBasicPassword("freeswitch"); config.setBasicUserName("works"); XmlRpcClient client = new XmlRpcClient(); client.setConfig(config); Object[] params = new Object[]{ "originate", "sofia/gateway/gizmo2/01160176xxxxxx &park()" }; client.execute( "freeswitch.api", params ); Since the code above is written "from my memory", take it for illustrative purpose only, however you should get the idea. If you need more control over the call(s), consider using http://wiki.freeswitch.org/wiki/Event_Socket. Hope this helps, Boris On Fri, Aug 22, 2008 at 12:24 PM, Boris Krivonog <boris.krivonog(a)gmail.com> wrote: > > The easiest way would be using XML-RPC (http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC) with Apache XML-RPC (http://ws.apache.org/xmlrpc/client.html) libraries the code would look something like: > > XmlRpcClientConfigImpl config = new XmlRpcClientConfigImpl(); > config.setServerURL( new URL("http://<freeswitchip>:8080/RPC2") ); > XmlRpcClient client = new XmlRpcClient(); > client.setConfig(config); > Object[] params = new Object[]{ "originate", "sofia/gateway/gizmo2/01160176xxxxxx &park()" }; > client.execute( "freeswitch.api", params ); > > Since the code above is written "from my memory", > > > > Adeel Ansari wrote: > > Hi All, > > I had success in bridging a session, using my Java Program as a hook, and manage to connect one mobile phone to another. It goes like below. > > === > --> originate sofia/gateway/gizmo2/01160176xxxxxx &java(net.digitalapps.freeswitch.CallTest) > === > > In this case, freeswitch call my Java program. Now, My intension is to not use console at all, instead use my java program to originate a session in freeswitch and then bridge the session. So, call freeswitch from my program, not the other way around. > > Is it possible? Need suggestions in any case. I mean if I need to come up with my own client. Actually, I was looking JAIN-SIP to achieve this. But then thought of asking your suggestions. With JAIN I might go like this, place a call, hit the dial plan invoke my Java Hook Program and bridge the call. But I am afraid that it might connect my client with the mobile phone, not one phone to another. > > Thanks. > > -- > Best, > Adeel Ansari > > http://www.linkedin.com/in/adeelansari > > ________________________________ > _______________________________________________ > 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 >

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IVR menu ending with #
by Sheeju Alex 02 Sep '08

02 Sep '08

Hi All, I have created a IVR menu where in the digits is ended with #, could anyone point me why a menu with ending digits (e.g. 12345#) doesn't work? <entry action="menu-exec-app" digits="/^(\d{5})#$/" param="transfer $1 XML default"/> Does Freeswitch IVR doesn't support # in digits? Thanks, Sheeju

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Re: [Freeswitch-users] API info
by Gayatri Kulkarni 02 Sep '08

02 Sep '08

Also, how to originate calls between endpoints registered to 2 different FSs?? Thanks a lot in advance, 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 2:45 AM To: freeswitch-users(a)lists.freeswitch.org Subject: API info 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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Blind/Unconditional , Busy , No Answer Call Forward
by Lee JJ 02 Sep '08

02 Sep '08

Hello : The better way , e.g. user press "*digit" setting no to be forwarded , and press another "*digit" to cancel forward . Where these setting should be stored ? DB , then convert into dialplan ? What info can I process call forward by channel/session state , busy , or no answer ? Thanks !

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Re: [Freeswitch-users] TLS and SRTP between 2 Freeswitch servers
by Peter P GMX 02 Sep '08

02 Sep '08

Hello Brian, any idea, when we can expect a fix for that? Best regards Peter Brian West schrieb: > I just realized this is a gateway call... that will have to be fixed. > > /b > > On Aug 27, 2008, at 3:24 PM, Peter P GMX wrote: > >> Hello Brian, >> >> I tried to set ;transport=tls on the sofia string and it showed me an >> invalid gateway >> >> So I checked the gateway: server1 is not registered on server2 >> freeswitch@freeswitch> sofia status gateway freeswitch2 >> API CALL [sofia(status gateway freeswitch2)] output: >> Invalid Gateway! >> >> When I set the register-transport parameter back from tls to udp on >> server1 I get >> freeswitch@freeswitch> sofia status gateway freeswitch2 >> API CALL [sofia(status gateway freeswitch2)] output: >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> ====================================================================== >> Name freeswitch2 >> Scheme Digest >> Realm xxx.xxx.xxx.56 >> Username 1007 >> Password yes >> From <sip:1007@xxx.xxx.xxx.56;transport=udp> >> Contact <sip:1007@xxx.xxx.xxx.55;transport=udp> >> To sip:1007@xxx.xxx.xxx.56 >> Proxy sip:xxx.xxx.xxx.56 >> Context default >> Expires 3600 >> Freq 3600 >> Ping 0 >> PingFreq 0 >> State REGED >> Status UP >> ============================== >> I ngrepped the traffic between server1 and server2 and could see that >> there was traffic on port 5060. >> >> Switching back to TLS it didn't work again. >> <param name="register-transport" value="tls"/> >> I ngrepped the traffic between server1 and server2 on all ports and >> could see that there was absolutely no traffic between them. >> >> So, as soon as I enter tls instead of upd, the traffic stops. >> Any hint? >> >> Best regards >> Peter >> >> >> >> Brian West schrieb: >>> You Append ;transport=tls on the sofia string which is much easier. >>> >>> /b >>> >>> >>> >>> On Aug 26, 2008, at 4:51 PM, Peter P GMX wrote: >>> >>> >>>> I have tried to set this up, but I need some help to get TLS to work >>>> >>>> What does work: >>>> I setup 2 freeswitch servers with IP xxx.xxx.xxx.55 and xxx.xxx.xxx. >>>> 56. >>>> They are connected via UDP/Port 5060 (SIP) >>>> I have 2 snom phones connected to the servers via TLS (1002@ >>>> xxx.xxx.xxx.55 and 1003@ xxx.xxx.xxx.56) >>>> Server1 registers to Server2 as UA 1007 >>>> I can dial "8001" on 1002@server and reach 1003@server2 and make a >>>> call >>>> >>> >>> >>> _______________________________________________ >>> 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 > > Brian West > sip:brian@freeswitch.org > > >

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ECHO NOT WORKING
by M.Emran 02 Sep '08

02 Sep '08

Hi, i have test my FS box with default config.I can hear ivr if i dial 5000. but echo testing is not working when i dial 9996. -- Regards ---------- M Emran InSpiration Software Ltd. Web: www.inspiresoftbd.com

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How to save RTP-audiostream to a file?
by Евгений Золотов 02 Sep '08

02 Sep '08

We had a question: > With which application and API is possible to keep full content of RTP audiostream to a file? > Our FreeSWITCH is configured in such manner that RTP-traffic directing throw FreeSWITCH, > but not between UAs. And took answer on it: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session ------------------------------------------------------------------- On it HELP we have mase such extension : <extension name="ext-record"> <condition field="destination_number" expression="^4010$"> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) 2008, Altron Inc." /> <action application="set" data="RECORD_SOFTWARE=FreeSwitch" /> <action application="set" data="RECORD_ARTIST=Registration" /> <action application="set" data="RECORD_COMMENT=!!!" /> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> <action application="set" data="ringback=${us-ring}"/> <action application="bridge" data="sofia/nat/${dialed_ext}@localhost:5062" /> <action application="answer" /> </condition> </extension> This draft works already, but there are new questions: 1. It's seems, that record of RTP session can occur only when RTP stream is routed through FS: UA-> FS-> UA, instead of directly (in LAN) UA-> UA. For this purpose, as we understand, it is necessary to set values in variables of FreeSWITCH: <action application="set" data="bypass_media=..."/> <action application="set" data="proxy_media=..."/> <action application="set" data="inbound_late_negotiation=..."/> ( for SIP profile nat for extension ext-record). What values need to be set in variables that all RTP traffic passed through FreeSWITCH? 2. At extension ext-record we set record_session then we do bridge to leg B. As it is possible to configurate registration from another extension, for example: <extension name="XXXX"> <condition field="destination_number" expression="^XXXX$"> ... <action application="execute_extension" data="ext-record XML public "/> ... <action application="bridge" data="sofia/nat/YYYY" /> ... </condition> </extension> - where own extension not call any leg B, but only record session XXXX->YYYY. How we are able to do this??? Thankful in advance, Evgeniy.

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How to obtain information about undocumented applications?
by Jon Bruel 02 Sep '08

02 Sep '08

A number of applications and other functions are not documented in the WIKI. For example, the application displace, is not documented. Where can I search to find the documentation regarding this and other applications?

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Re: [Freeswitch-users] DTMF problem
by Michael S Collins 02 Sep '08

02 Sep '08

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

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DTMF problem
by Luis M. Zuccolo 02 Sep '08

02 Sep '08

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

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