- FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Country Codes + Area codes
by Seysan 01 Sep '08

01 Sep '08

Hello, Does anyone have or Can help me find a Complete list of All the Countries in the world with the Countrycode? Also, is there a list with cities breakdown o at least important cities of each country plus their codes? I hoping an Excel or CSV file, but if this list exist it doesn't matter what format, I will be able to use it. thank you in advance, Seysan

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