- FreeSWITCH-users - FreeSWITCH Mailing Lists

ringnack does not working
by Farrukh Ahmed 02 Aug '07

02 Aug '07

Dear Sir, I was trying to get ringback working but its not working. I used exactly what docs said. can some one tell me what does it means http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready ? i could not find any thing regarding this on fs example config files but its on wiki my FS Configuration File FYI, http://pastebin.freeswitch.org/3155 Best Regards. -- Färrükh Ähmëd

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Information about freeswitch and skypho
by Sauro Cesaretti 31 Jul '07

31 Jul '07

Hello everyone, I'd like to know if it's possible to use freeswitch to make calls and receive calls using skypho. I'm sure that skypho use a sip protocol and I even configured it with ekiga but I don't know how can I configure it with freeswitch. beside, I'd like to know if it'll be possible use freeswitch under text-console. Thaks for all information in advance Best regards, Sauro

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Re: [Freeswitch-users] segfaults after 24-36 hours
by Anthony Minessale 30 Jul '07

30 Jul '07

We believe there is a bug inside the actual sofia library. We need to deal with the sofia developers so we can figure out what's wrong. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Ivan C Myrvold <ivan(a)myrvold.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, July 30, 2007 10:59:54 AM Subject: Re: [Freeswitch-users] segfaults after 24-36 hours Using it at home, maybe 2-3 calls per day :) Ivan Den 30. jul. 2007 kl. 14:02 skrev Brian West: Weird we have boxes that don't have this issue... what kind of call load are you putting on it? freeswitch@pacman> status API CALL [status()] output: UP 0 years, 5 days, 17 hours, 50 minutes, 47 seconds, 241 milliseconds, 637 microseconds /b On Jul 29, 2007, at 3:51 PM, Ivan C Myrvold wrote: Hope that someone can fix this, because I am getting tired of restarting FreeSwitch every 1-2 days. _______________________________________________ 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 ____________________________________________________________________________________ Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more. http://mobile.yahoo.com/go?refer=1GNXIC

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freeswitch and enum
by Yogesh Chaturvedi 30 Jul '07

30 Jul '07

Hi, I am trying to make call through enum lookup and using bind 9.4.1 as a dns server. My zone file look like this : $ORIGIN . $TTL 21600 ; 6 hours 91.e164.arpa IN SOA ns1.91.e164.arpa. hostmaster.ns1.91.e164.arpa. ( 2005022250 ; serial 10800 ; refresh (3 hours) 3600 ; retry (1 hour) 604800 ; expire (1 week) 86400 ; minimum (1 day) ) NS ns1.91.e164.arpa. A 192.168.96.57 $TTL 60 ; 1 minute NAPTR 100 10 "u" "SIP+E2U" "!^.*$!sip:100@192.168.96.122!i" . $ORIGIN 91.e164.arpa. 1 NAPTR 100 10 "u" "E2U+SIP" "!^.*$!sip:100@192.168.96.122!i" . athir A 192.168.96.57 here i have two xlite phone register with freeswitch and i want to make call between them using enum. I have given name to them as 1 and 2 now i want to call to 1 using 2. Freeswitch give me error dns error and do not resolve 1.91.e164.arpa. Even when i test the naptr record using nslookup utility of bind it give me right answer but freeswitch fail to resolve the address. Thanks Yogesh

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Re: [Freeswitch-users] how to detect dtmf wave files and convert it to DTMF tones in RFC 2833
by Anthony Minessale 27 Jul '07

27 Jul '07

The easy way is the start_dtmf application you can call from your dialplan. Once you call it any inband dtmf that is detected in later apps will be interpreted. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Dinesh Dialani <ddialani(a)devfoundry.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, July 27, 2007 1:31:08 AM Subject: [Freeswitch-users] how to detect dtmf wave files and convert it to DTMF tones in RFC 2833  Hi All, There exists a library libteletone. I know that it has functionality to detect and generate DTMF tones, but I didn’t get any help as to how can I test it with my FreeSwitch configuration, as no document is available on http://wiki.freeswitch.org/wiki Is there anybody who used detection and generation of DTMF tones using libteletone library in FreeSwitch and can help me to use it? Thanks in advance. Dinesh ____________________________________________________________________________________ Choose the right car based on your needs. Check out Yahoo! Autos new Car Finder tool. http://autos.yahoo.com/carfinder/

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Cut a Bridge with a DTMF
by Stéphane Lasserre 27 Jul '07

27 Jul '07

Hello Is it possible to cut a call made with a bridge with a DTMF ? Does I need to made a conference instead ? Thanks _________________________________________________________________ David Guetta a réuni les sons les plus connus de Messenger dans le Mix Messenger, le son de l’été ! Téléchargez-le gratuitement ! http://specials.divertissements.fr.msn.com/mixmessenger

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Play the Ring Tone on a Javascript Bridge
by Stéphane Lasserre 27 Jul '07

27 Jul '07

Hello Is it possible to open play the ring tones on a Javascript on a bridge because this script do not play the ringtones new_session.originate(session, "sofia/default/"+Number2Dial); new_session.ring_ready; new_session.setCallerData("ringback","%(2000, 4000, 440.0, 480.0)"); new_session.waitForAnswer(60000); if (session.ready() && new_session.ready()) { bridge(session, new_session); } I try on a xml file he work but not on the javascript <action application="ring_ready"/> <action application="set" data="ringback=%(2000, 4000, 440.0, 480.0)"/> <action application="bridge" data="sofia/default/003"/Number2Dial> Thanks _________________________________________________________________ David Guetta a réuni les sons les plus connus de Messenger dans le Mix Messenger, le son de l’été ! Téléchargez-le gratuitement ! http://specials.divertissements.fr.msn.com/mixmessenger

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how to detect dtmf wave files and convert it to DTMF tones in RFC 2833
by Dinesh Dialani 27 Jul '07

27 Jul '07

Hi All, There exists a library libteletone. I know that it has functionality to detect and generate DTMF tones, but I didn't get any help as to how can I test it with my FreeSwitch configuration, as no document is available on http://wiki.freeswitch.org/wiki Is there anybody who used detection and generation of DTMF tones using libteletone library in FreeSwitch and can help me to use it? Thanks in advance. Dinesh

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ODBC.Getdata function do not work on Windows
by Stéphane Lasserre 26 Jul '07

26 Jul '07

Hello The javascript function odbc.getdate do not work on windows plateform because the C function SQLRowCount do not return the number of row in case of select. If you remove this verification it return the values. Stephane _________________________________________________________________ Windows Live Messenger vous offre 30 nouvelles émoticônes gratuites, installées directement dans votre Messenger ! http://www.emoticones-messenger.fr/

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how to configure the *.xml to make sip re gistration
by cake918＠126.com 23 Jul '07

23 Jul '07

hello, first thanks those people who have given advice to me. this time i shall register the special sip caller ,but not have anything registered. at the begin,i rename the freeswitch_combined.xml to freeswitch.xml,so i can configure all things in one .xml file. i have made the first configuration to test stuff out. then,when i want to register a special sip caller,though i have read the sofia.xml reference to configure it,the UA cant't reach it. is there some file else need to configure? because the freewitch_combined.xml which i have renamed to freeswitch.xml have not the #include to include sofia.xml, so i have to use #include to incude this file again.does this make sense? another question, in the freeswitch.xml it uses #include,and in freeswitch_combined.xml it uses <load module=...>.what's the difference betweem the two ? thank you very much! 百万同时在线，人山人海同玩梦幻西游

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