September 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Checking Dial Status in FS
by Ahmed Munir 18 Sep '09

18 Sep '09

Hi, In FS, which function can be used for checking the dial status (in channel variables as well as mod_perl function/class)? In asterisk $DIALSTATUS is used to check the status i.e. busy, answer, etc. Kindly do let me know. -- Regards, Ahmed Munir

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FreeSWITCH Weekly Conference Call, Sept 18, Starting Shortly
by Michael Collins 18 Sep '09

18 Sep '09

Get ready to call in! Agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_18 Talk to you all shortly! -MC

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mod_lcr and indexes
by Marcelo Sosa - LST 18 Sep '09

18 Sep '09

Hello all, This is my first message on the list, i´m pretty new to FS. I was playing a bit with mod_lcr and found that the sql query for fetching the lowest rate can be changed to a better use of indexes, at least on mysql. Anyone can do some test using other DBs? The change i've made was simple, the original query was something about "... AND digits IN (12345, 1234, 123, 12, 1) ..." and using EXPLAIN i saw that it was using carrier_id as key for the biggest table and not digits. I've changed the code so the query is " AND (digits='12345' OR digits='1234' OR digits='123' OR digits='12' OR digits='1') " and mysql uses the index from the digits row, reducing the returned resultset of the subquery from all the digits from a carrier to the number of "OR" in the query (in my case, from 19850+ to 14). Anyone think that this may be a nice change? or it is just a bad use of indexes by mysql? Regards, Marcelo Sosa

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Nokia N800
by Valentin Doroga 18 Sep '09

18 Sep '09

I'm coming again with the same question: does anybody have any instructions how to build FreeSWITCH for Nokia N800? Or, does anybody have a recently built version? Val we relocated the machine with the build env for that, I'll try to find the time to resurrect it and make a new one. On Wed, Apr 1, 2009 at 4:00 PM, Valentin Doroga <valentin.doroga(a)pronexus.com> wrote: There are some old binaries at: http://www.freeswitch.org/downloads/n800/ Is there a newer version? Any place with instruction to build? Val. _______________________________________________ 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 <http://www.freeswitch.org/> -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net/> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400

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Simple call waiting question
by Mark Campbell-Smith 18 Sep '09

18 Sep '09

HI All, I am trying to create a simple call waiting dialplan as my phone does not have Recall button. The simple scenario is: 1. B calls user A and is answered 2. C calls user A 3. A puts B on hold 4. A answers C 4. A then recalls first call from B I was going to use fifo for step 3. I am doing: <action application="transfer" data="-bleg 5900 XML default"/> and then <condition field="destination_number" expression="^5900$"> <action application="set" data="fifo_music=$${hold_music}"/> <action application="fifo" data="5900@${domain_name} in"/> </condition> My question is then, how do I get A to answer the call from C? Step 4 and then step 5? On another note, if I put <action application="transfer" data="-aleg 5900 XML default"/> I get the following in the console: Transfer sofia/internal_nat/sip:1000@xx.xx.xx.xx:5060 to 5900[-aleg@XML] Shouldn't that be XML[5900@default] ? Thanks!

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Re: [Freeswitch-users] DB table sip_dialogs is always empty
by Martin Dvorak 18 Sep '09

18 Sep '09

Hello, let me apologize for long reply delay, but my f***** mail server "eat" nearly all messages from FS mailing list (I have received just three e-mails in the past five days...). So, lets try another one :-) Now back to the thema: Thank you very much, Brian, Anthony and Michael for your answers! If I catch it right, the recommended way to get current calls info is to use "show channels" command. Am I right? BTW: If this kind of info is stored in the internal SQLite, isn't be useful to make it configurable => SQLite x ODBC? Thanks once more! Best regards kokoska.rokoska anthm wrote: that table is specific to the manage-presence option On Thu, Sep 17, 2009 at 6:08 AM, kokoska.rokoska <kokoska.rokoska at post.cz>wrote: > I'm sorry to resend this post, but even after few hours I can't see it > in the mailing-list... > Thanks. > > > Best regards, > > kokoska.rokoska > > > kokoska rokoska napsal(a): > > Hello, > > > > I have setted-up odbc-dsn on all my FreeSWITCH sofia profiles and based > > on logs FS connected to the dsns correctly. But during the calls I can't > > see any rows in sip_dialogs table. When I run "show channels" from > > console it works fine. > > All other tables are "populated and maintained" as I expect :-) > > > > I'm running FreeSWITCH on 64bit Centos 5.3 with Postgresql 8.1 DB... > > > > Could you, please, point what I'm missing, or where I'm wrong? Thank you > > very much! > > > > > > Best regards, > > > > kokoska.rokoska > > >

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FreeSWITCH Documentation
by Muhammad Shahzad 18 Sep '09

18 Sep '09

Hi, I have observed that one of the major hurdle while writing patches and / or bug fixes is lack of doxygen documentation for FS source code. For example it took me 5+ days to understand mod_dingaling code and its hooks into FS source code to write up soft reload patch, while it could have taken less then 3 days to do so if source code documentation was available. So, since right now i have some human resources including myself available, I would like to document all source code (or at least core FS code i.e. everything that has "switch_" prefix) using doxygen. I know its a huge task and will take a while to complete but at least lets get it started. If anyone else wants to participate as well in this task, then we can team up to complete it quickly. Let me know if you guys are interested. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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Re: [Freeswitch-users] skill-based ACD
by Pete Mueller 18 Sep '09

18 Sep '09

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Event socket library bug on hangup?
by Shameem Shiek 17 Sep '09

17 Sep '09

When a call is hangup, I get SERVER_DISCONNECTED Event over and over again instead of CHANNEL_HANGUP event. Has anyone else experienced this? I am using freeswitch 1.04.

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"barge in" implementation with mod_socket and eavesdrop
by Artem Shiyanov 17 Sep '09

17 Sep '09

Hello! I'm trying to implement "barge in" functionality (see http://www.yourdictionary.com/telecom/barge-in) with "eavesdrop" but still with no success. The situation is: - Person A calls to the extension: <extension name="some_ext"> <condition field="destination_number" expression="^900.$"> <action application="answer"/> <action application="park"/> </condition> </extension> - I bridge him with person B with help of mod_socket: SendMsg <some_uuid> call-command: execute execute-app-name: bridge execute-app-arg: <person B address in form: "user/..."> - A and B talks - Person C decides to barge in the call A<-->B (to become a third participator in the call) a) I send (mod_socket): SendMsg <C_uuid> call-command: execute execute-app-name: eavesdrop execute-app-arg: <A_uuid or B_uuid, result is the same> b) Then, as the spec says ( http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop) I sent DTMF "3" with api uuid_send_dtmf <C_uuid> 3 but it doesn't work. I mean that A can hear B and vice verse, but both A and B can't hear C. C also doesn't hear neither A nor B. If I press "3" on the C's softphone (latest X-Lite) then, really, C becomes a full-capabilities participator of the call. Instead of "uuid_send_dtmf" I tried: 1) sendevent DTMF Unique-ID: <C_uuid> DTMF-Digit: 3 DTMF-Duration: 2000 2) first make queue_dtmf for the <C_uuid>, and then eavesdrop 3) SendMsg <C_uuid> call-command: execute execute-app-name: gentones execute-app-arg: 3 4) SendMsg <C_uuid> call-command: execute execute-app-name: send_dtmf execute-app-arg: 3 And none of these methods leads to the "barged in" call. Anyone knows how to press "3" programmatically on behalf of the given channel with mod_socket?! Artem

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