February 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Suggestion for xml_curl performance
by shehzad p 27 Feb '09

27 Feb '09

Hi all, Recently I faced some performance bottleneck by using Javascript. Now I am testing xml_curl for next setup, In terms of performance and stability will any body give me some information, what are the pros & cons of using xml_curl, what is precaution for using it, or any other recommendations... Thanks, msp -- View this message in context: http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p221181… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Cant get Disposition status in Javascript
by Baskar 26 Feb '09

26 Feb '09

Hi, I am using javascript to store uuid, phone_no, endpoint_disposition and hangup cause in my MYSQL. I can get the session UUID , Phone_no, endpoint disposition but i cant get the originate disposition. Javascript : session.setVariable("session.uuid", "ses_uuid: " + session.uuid); session.setVariable("phone", "phone_no: " +argv[0]); result = session.getVariable("endpoint_disposition") hangup = session.getVariable("originate_disposition") OUTPUT: S_UUID PHONE_NO RESULT HANGUP_STATE f579cb15-5145-4eb1-a080-03b9e53b90f739841799874 39841799874 ANSWER for "originate_disposition" i did not get any value stored in the Table. So how can get the originate_disposition ???? -- Warm Regards, N.Baskar

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Re: [Freeswitch-users] switch voices in ivr menus
by Alexander de Greiff 26 Feb '09

26 Feb '09

brian, demo3 is the main menu. all submenus change voices correctly. when i go to the main menu via menu-sub (7) then the voice is changes correctly. only when i menu-top (9) to main menu the voice is not changed. how do i produce the debug log? in the cli? this i a remote terminal. f8 is not an option. here is the part of my ivr.conf.xml: <menu name="demo3" greet-long="say:guten tag. das ist das hauptmenue. sie koennen eine durchwahl waehlen oder fuer untermenue 1 druecken oder untermenue 2 oder 9 oder stern oder 8" greet-short="say:Hauptmenue. 1=untermenue 1. 2=untermenue2. 9=top. stern=exit. 8=zurueck." invalid-sound="say:ungueltige auswahl" exit-sound="say:auf wiedersehen" timeout ="10000" max-failures="3" tts-engine="cepstral" tts-voice="matthias" phrase_lang="de"> <entry action="menu-sub" digits="1" param="demo3_submenu_1"/> <entry action="menu-sub" digits="2" param="demo3_submenu_2"/> <entry action="menu-exec-app" digits="/^([1-2][0-1][0-9])$/" param="transfer $1 XML features"/> <entry action="menu-top" digits="9"/> <entry action="menu-exit" digits="*"/> <entry action="menu-back" digits="8"/> </menu> <menu name="demo3_submenu_1" greet-long="say:das ist untermenue 1. Bitte treffen sie folgende auswahl. top 9 druecken. eit stern druecken. zurueck 8 druecken." greet-short="say:untermenue 1. top 9. stern exit. 8 zurueck." invalid-sound="say:ungueltige auswahl" timeout ="10000" max-failures="3" tts-engine="cepstral" tts-voice="katrin" phrase_lang="de"> <entry action="menu-sub" digits="2" param="demo3_submenu_2"/> <entry action="menu-sub" digits="7" param="demo3"/> <entry action="menu-top" digits="9"/> <entry action="menu-exit" digits="*"/> <entry action="menu-back" digits="8"/> </menu> <menu name="demo3_submenu_2" greet-long="say:das ist untermenue 2. Bitte treffen sie folgende auswahl. top 9 druecken. exit stern druecken. zurueck 8 druecken." greet-short="say:untermenue 1. top 9. stern exit. 8 zurueck." invalid-sound="say:ungueltige auswahl" timeout ="10000" max-failures="3" tts-engine="cepstral" tts-voice="matthias" phrase_lang="de"> <entry action="menu-sub" digits="1" param="demo3_submenu_1"/> <entry action="menu-sub" digits="7" param="demo3"/> <entry action="menu-top" digits="9"/> <entry action="menu-exit" digits="*"/> <entry action="menu-back" digits="8"/> </menu>

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switch voices in ivr menus
by Alexander de Greiff 26 Feb '09

26 Feb '09

another question: - in the ivr application i have cepstral voice matthias read the main menu ok. - i select submenu1 and cepstral voice katrin reads the submenu1 correctly. - i go back to the main menu and the voice is not switched back to the specified voice matthias. each voice is explicitly specified in each menu. any suggestions? kind regards alex

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Re: [Freeswitch-users] [ANN] Spice Telephony - an open source FreeSWITCH/Erlang callcenter platform
by JCATS 26 Feb '09

26 Feb '09

Have you planned any predictive dialer features ( like VICIDIAL )? -- View this message in context: http://www.nabble.com/-ANN--Spice-Telephony---an-open-source-FreeSWITCH-Erl… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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dialplan condition regex question
by Alexander de Greiff 26 Feb '09

26 Feb '09

hi all, i am dialing the number 123456789 (example) reaching fs via inbound sip gateway and hitting following dialplan: <condition field="${sip_to_user}" expression="^123456(\d+)$"> <action application="set" data="my_dialed_extension=$1"/> </condition> <condition field="${my_dialed_extension}" expression="^789"> ... <action application="bridge" data="{ignore_early_media=true}user/${my_dialed_extension}@$${domain}"/> ... </condition> via info i can see that the variable my_dialed_extension is populated ok with 789 but somehow the second condition is not met. when i change that to match (.*) the actions gets executed and the my_dialed_extension inside is correct. any suggestions? kind regards alex

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Session Timer
by mashudi 26 Feb '09

26 Feb '09

Hi Folks, in case of FreeSwitch sip message response for UPDATE message wih SIP/2.0 200 OK, how to change the session timer value from 120 to 300 ? Session-Expires: 120;refresher=uac. thank's for help and suggestion. mashudi ***************************************** Sekarang Gratis Nelpon SLJJ Flexi diperluas ke Yogya *****************************************

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SheevaPlug Development Kit
by Kristian Kielhofner 25 Feb '09

25 Feb '09

Hello everyone, I just ordered one of these: http://www.marvell.com/products/embedded_processors/developer/kirkwood/shee… Just over $110 with shipping but they are expecting the price to come down quite a bit (perhaps as low as $50): - 1.2Ghz ARM5 - 512MB RAM - Multiple flash storage options - Gigabit ethernet - USB 2.0 - 5 watt power usage They probably won't be shipping until late March but I thought I'd get my order in early. Who knows how practical it will be but needless to say I'm going to get FreeSWITCH to run on it! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

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Re: [Freeswitch-users] Adding an info digit to sip from
by Josh Forman 25 Feb '09

25 Feb '09

The problem here is that what you are showing me produces: From: "1004" <sip:1004@99.185.85.3>;tag=1SparjgraS69m To: <sip:1235@conference.freeswitch.org;this=rocks> when what I need to output would look like this: From: "1004" <sip:1004@99.185.85.3;this=rocks>;tag=1SparjgraS69m To: <sip:1235@conference.freeswitch.org> with the "this=rocks" in the FROM field, not the TO field. I know that you can change parts of the from field by setting effective_caller_id_name and effective_caller_id_number, but I don't know how I would add that bit of data to the end of the SIP URI inside the < > Is there a variable that I could set or perhaps some method similar to overwriting the To header shown at http://wiki.freeswitch.org/wiki/Sofia#Modifying_the_To :_header that can be used to accomplish this? Thanks Josh On Feb 25, 2009, at 2:35 PM, freeswitch-users-request(a)lists.freeswitch.org wrote: >> Message: 3 >> Date: Wed, 25 Feb 2009 09:29:26 -0600 >> From: Brian West <brian(a)freeswitch.org> >> Subject: Re: [Freeswitch-users] Adding an info digit to sip from >> header >> To: freeswitch-users(a)lists.freeswitch.org >> Message-ID: <0E1E9F62-64C5-4B70-9C67-C7B5728DB111(a)freeswitch.org> >> Content-Type: text/plain; charset="us-ascii" >> >> You can do something like this "sofia/blah/somenumber@someip: >> 5060;this=rocks" >> >> /b >> >> On Feb 25, 2009, at 9:22 AM, Josh Forman wrote: >> >>> I'm trying to edit the sip headers to make the from field look like >>> this: >>> >>> From: <sip:SOMENUMBER@SOMEIP:5060;isup-oli=27>;tag=gK0a00d6ea. >>> >>> I know that to read that data on an incoming sip message it is in $ >>> {sip_from_params}, but how can I add the ;isup-oli=27 part on an >>> outgoing message? >>> >>> Thanks >>> >>> Josh > > > > ------------------------------ > > Message: 3 > Date: Wed, 25 Feb 2009 10:53:39 -0600 > From: Brian West <brian(a)freeswitch.org> > Subject: Re: [Freeswitch-users] Adding an info digit to sip from > header > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <D58A3A53-8535-4EAF-BFE5-B86F621C1393(a)freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed > > It will actually add it to both places > > INVITE sip:1235@conference.freeswitch.org;this=rocks SIP/2.0 > Via: SIP/2.0/UDP 99.185.85.3;rport;branch=z9hG4bK0Kaa1322U42eK > Max-Forwards: 69 > From: "1004" <sip:1004@99.185.85.3>;tag=1SparjgraS69m > To: <sip:1235@conference.freeswitch.org;this=rocks> > > I verified it does indeed add it in both places. > > /b > > > > On Feb 25, 2009, at 10:34 AM, Josh Forman wrote: > >> Which variable would I need to set via the dialplan to do this >> though? Your example looks like it would be the dialstring for the >> bridge application but if that works it would probably be added to >> the >> To header instead of the From, right? I can't be sure since nothing >> I've tried has had any affect. >> Between looking at the wiki and random experimenting I haven't found >> anything that works thus far. >

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Re: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu
by Alexander de Greiff 25 Feb '09

25 Feb '09

michael, i googled for "asterisk alternative" and voila... the trigger was that every now and then i renew servers in the infrastructure and the one with asterisk was overdue. i wasn't really unhappy with asterisk, but these things bothered me (maybe i am not up to date): - dialplan gets messy - no conferences without hardware (rented remote server!) - ivr with cepstral voices: sometimes get hickups so far i like the fs approach very much. stable sip channels, no hickups with voices. kind regards alex ----- Ursprüngliche Mail ----- Von: "Michael Collins" <msc(a)freeswitch.org> An: freeswitch-users(a)lists.freeswitch.org Gesendet: Mittwoch, 25. Februar 2009 19:31:24 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu > so far i am a happy camper with freeswitch. this is a different snack bracket than asterisk... If you don't mind telling us, where did you hear about FS and what made you decide to try it? Are you unhappy with Asterisk or are you simply looking for something a bit different? Just curious. Thanks, MC _______________________________________________ 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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