September 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

dialpaln
by Cliconnect 22 Sep '08

22 Sep '08

Hi, I can call extensions like 1000 with <extension name="extensions"> <condition field="destination_number" expression="^(\d+)$"> <action application="bridge" data="sofia/doublenat/$1%voipclic.com "/> </condition> </extension> and I can call PSTN with <extension name="PSTN"> <condition field="destination_number" expression="^(\d+)$"> <action application="bridge" data="sofia/gateway/inphonex/$0"/> </condition> </extension> Both config being in the same default.xml dialplan but I must use one or the other config. What I have to do in order to have them working at the same time. In other words the same user can call PSTN and the extension. thanks Duan

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Java script test
by imthiyaz＠peopletech.co.in 22 Sep '08

22 Sep '08

Hi, I am new to freeswitch and trying to run a javascript example calltest.js.I have put that code inside the folder /usr/local/freeswitch.I have started the freeswitch and from that console I have tried to run the program like jsrun /usr/local/freeswitch/calltest.js. but i am getting the following error: 2008-09-20 15:49:32 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/1001(a)192.168.1.2 [d1987c09-ebd2-4590-b072-ce851e4b5794] 2008-09-20 15:49:32 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/0000000000(a)192.168.1.2 [c98bc58d-2ef9-4656-8f74-9c024000d18c] 2008-09-20 15:49:32 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing FreeSWITCH->1001@public 2008-09-20 15:49:32 [ERR] sofia.c:2845 sofia_handle_sip_i_refer() Cannot Blind Transfer 1 Legged calls 2008-09-20 15:49:32 [NOTICE] switch_core_state_machine.c:157 switch_core_standard_on_execute() Hangup sofia/internal/0000000000(a)192.168.1.2 [CS_EXECUTE] [NORMAL_CLEARING] 2008-09-20 15:49:32 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/internal/1001(a)192.168.1.2 [CS_ROUTING] [NO_USER_RESPONSE] 2008-09-20 15:49:32 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 23 (sofia/internal/0000000000(a)192.168.1.2) Ended 2008-09-20 15:49:32 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/0000000000(a)192.168.1.2 [CS_HANGUP] 2008-09-20 15:49:32 [WARNING] mod_spidermonkey.c:2933 session_originate() Cannot Create Outgoing Channel! [sofia/default/1001(a)192.168.1.2] 2008-09-20 15:49:32 [ERR] inline:1 mod_spidermonkey() You must call the session.originate method before calling this method! 2008-09-20 15:49:32 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 22 (sofia/internal/1001(a)192.168.1.2) Ended 2008-09-20 15:49:32 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1001(a)192.168.1.2 [CS_HANGUP] the code is: new_session = new Session(); new_session.originate(session, "sofia/default/1001(a)192.168.1.2"); new_session = new Session(session, "sofia/default/1001(a)192.168.1.2"); // wait for it ... answer new_session.waitForAnswer(10000); // IF everybody is ready, then bridge our current session & the new_session if (session.ready() && new_session.ready()) { bridge(session, new_session); } // log to the console console_log("***********Voila!\n"); // hangup when done new_session.hangup(); Is there anything i have to add to the program or any other requirements are needed to run this javascript successfully? please assist me to do this. -------------------------------------------------------------------- mail2web.com - Microsoft® Exchange solutions from a leading provider - http://link.mail2web.com/Business/Exchange

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cant record any session
by xbipin 22 Sep '08

22 Sep '08

i just copied the dialplan given in wiki for record session but cant record or lets say first i cant initiate the call only, sofia says codec not found as soon as i hit dial on the dialer can some1 help in the xml as im trying to record calls made from a specific user whose account is created only and not in general for every1 -- View this message in context: http://www.nabble.com/cant-record-any-session-tp19594481p19594481.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Gbridge
by EdPimentl 21 Sep '08

21 Sep '08

http://www.gbridge.com/ + FS = Awesome -E

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Khushil Dep invites you to their network on Ecademy
by khushil.dep＠gmail.com 21 Sep '08

21 Sep '08

Connect with me on Ecademy Hi I've added you as one of my contacts on Ecademy where successful business people grow their business. Click here to connect with me: http://www.ecademy.com/account.php?op=signup&xref=310185 Thank you, Khushil Dep khushil.dep(a)gmail.com This email was sent to freeswitch-users(a)lists.freeswitch.org To avoid receiving these emails in the future send an email from that address to invite-unsubscribe(a)ecademy.com or go to http://www.ecademy.com/blockinvites.php?e=freeswitch-users@lists.freeswitch… Email:support@ecademy.com Company Registration:3651083 VAT:718 0377 36

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
by Jon Bruel 21 Sep '08

21 Sep '08

The FREESWITCH version is the 1.0.1 release version. I have updated to the snapshot from the 20th of September. But I get problems with the ./configure. The bootstrap.sh works OK after I have updated autoconf to version 2.63 and automake to version 1.10.1. The error from ./configure after half a page of lines is: ./configure: line 4693: syntax error near unexpected token `build_libtool_libs,' ./configure: line 4693: ` _LT_DECL(build_libtool_libs, enable_shared, 0, Operating system is CentOS, uname -r = 2.6.9.-67.0.4.plus.c4. Jon

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Re: [Freeswitch-users] How to upgrade ?
by Christian Jensen 21 Sep '08

21 Sep '08

I am working on a patch that would externalize the install configs, much the same way as the extensions folder (which i propose to be called "default") The new folders will be the same name as the context they are called from, for example the one i am most interested in is "public.d" Any objections? -----Original Message----- From: Michael Jerris <mike(a)jerris.com> Sent: September 20, 2008 3:24 PM To: freeswitch-users(a)lists.freeswitch.org <freeswitch-users(a)lists.freeswitch.org> Cc: freeswitch-users(a)lists.freeswitch.org <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] How to upgrade ? Make current will not work from release tarballs but if you grab the new version it should install without wiping out your config. Mike On Sep 20, 2008, at 6:10 PM, Michael S Collins <msc(a)freeswitch.org> wrote: > Yes "make current" will update you without breaking your existing > configs. Even if you do "make samples" it won't overwrite your > configs bit it will replace any missing conf files with default ones. > > -MC > > Sent from my iPhone > > On Sep 20, 2008, at 2:46 PM, henkoegema <pabx_freeswitch(a)telenet.be> > wrote: > >> >> At the moment I'm using FS version 1.0.0. >> >> I want to upgrade to the latest version. >> Can I just type >> #make current (?) :working: >> without loosing my own (or edited standard) conf files ? >> >> >> -- >> View this message in context: http://www.nabble.com/How-to-upgrade---tp19589317p19589317.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> 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 _______________________________________________ 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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Conference members variable
by Torstein Knutsen 21 Sep '08

21 Sep '08

Hi Everybody I am building a conferencing service, and I was wondering if there is a system variable that I can use, that holds the number of members in a specific conference. Is there a variable I can read (in java, or anywhere) that tells me this ? If there is, is there a list of FreeSWITCH variables anywhere ? I can read the members (and the total) via CLI (conference <conference> list), but I don't know how to "import" this figure to my (java)script. Thank you very much for your time Torstein

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Re: [Freeswitch-users] Error when running javascript
by imthiyaz＠peopletech.co.in 20 Sep '08

20 Sep '08

Hi brain pls check this link http://wiki.freeswitch.org/wiki/Examples_calltest_js thanks Imthiyaz Original Message: ----------------- From: Brian West brian(a)freeswitch.org Date: Sat, 20 Sep 2008 11:32:36 -0500 To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Error when running javascript Can you tell me on the wiki where this example is? /b On Sep 20, 2008, at 5:28 AM, preetha Ayyappan wrote: > Hi, > I am new to freeswitch and trying to run a javascript example > calltest.js.I have put that code inside the folder /usr/local/ > freeswitch.I have started the freeswitch and from that console I > have tried to run the program like jsrun /usr/local/freeswitch/ > calltest.js. > but i am getting the following error: > > 2008-09-20 15:49:32 [NOTICE] switch_channel.c:534 > switch_channel_set_name() New Channel sofia/internal/ > 1001(a)192.168.1.2 [d1987c09-ebd2-4590-b072-ce851e4b5794] > 2008-09-20 15:49:32 [NOTICE] switch_channel.c:534 > switch_channel_set_name() New Channel sofia/internal/0000000000(a)192.168.1.2 > [c98bc58d-2ef9-4656-8f74-9c024000d18c] > 2008-09-20 15:49:32 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() > Processing FreeSWITCH->1001@public > 2008-09-20 15:49:32 [ERR] sofia.c:2845 sofia_handle_sip_i_refer() > Cannot Blind Transfer 1 Legged calls > 2008-09-20 15:49:32 [NOTICE] switch_core_state_machine.c:157 > switch_core_standard_on_execute() Hangup sofia/internal/0000000000(a)192.168.1.2 > [CS_EXECUTE] [NORMAL_CLEARING] > 2008-09-20 15:49:32 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() > Hangup sofia/internal/1001(a)192.168.1.2 [CS_ROUTING] [NO_USER_RESPONSE] > 2008-09-20 15:49:32 [NOTICE] switch_core_session.c:807 > switch_core_session_thread() Session 23 (sofia/internal/0000000000(a)192.168.1.2 > ) Ended > 2008-09-20 15:49:32 [NOTICE] switch_core_session.c:809 > switch_core_session_thread() Close Channel sofia/internal/0000000000(a)192.168.1.2 > [CS_HANGUP] > 2008-09-20 15:49:32 [WARNING] mod_spidermonkey.c:2933 > session_originate() Cannot Create Outgoing Channel! [sofia/default/1001(a)192.168.1.2 > ] > 2008-09-20 15:49:32 [ERR] inline:1 mod_spidermonkey() You must call > the session.originate method before calling this method! > 2008-09-20 15:49:32 [NOTICE] switch_core_session.c:807 > switch_core_session_thread() Session 22 (sofia/internal/1001(a)192.168.1.2 > ) Ended > 2008-09-20 15:49:32 [NOTICE] switch_core_session.c:809 > switch_core_session_thread() Close Channel sofia/internal/1001(a)192.168.1.2 > [CS_HANGUP] > > the code is: > new_session = new Session(); > > new_session.originate(session, "sofia/default/1001(a)192.168.1.2"); > > new_session = new Session(session, "sofia/default/1001(a)192.168.1.2"); > > // wait for it ... answer > new_session.waitForAnswer(10000); > > // IF everybody is ready, then bridge our current session & the > new_session > if (session.ready() && new_session.ready()) { > bridge(session, new_session); > } > > // log to the console > console_log("***********Voila!\n"); > > // hangup when done > new_session.hangup(); > > > Is there anything i have to add to the program or any other > requirements are needed to run this javascript successfully? > please assist me to do this. > -- > Preetha.A > _______________________________________________ > 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 -------------------------------------------------------------------- myhosting.com - Premium Microsoft® Windows® and Linux web and application hosting - http://link.myhosting.com/myhosting

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g729 sounds files - voicemail
by Gabriel Kuri 20 Sep '08

20 Sep '08

I've pretty much standardized on using g729 as the codec of choice for the users connecting to my FS installation, primarily due to the users I have that are on very slow connections and all of my carriers support g729, so no need to transcode. The sounds files that comes with FS are raw PCM files. Since FS is unable to currently transcode to/from g729, is there any way to use sounds files that are in the g729 format, such that it is technically still "pass-through" on the FS side and no transcoding is required? Ideally, I'm looking to install pre-coded g729 files for voicemail and have FS playback those files when someone logs into voicemail. I haven't tried simply replacing the files in the sounds directory, what I would prefer is to have a separate directory structure with g729 sound files and have FS detect the codec in use and automatically playback the files for the appropriate codec? Thanks Much... Gabriel Kuri

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