April 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

compile errors
by srinivasula reddy 09 Apr '10

09 Apr '10

Hi, i am compiling the freeswitch1.0.2 its giving compilation errors in libsofia_sip_ua_static project. my enviroment is windows xp. any idea -- Srinivasula Reddy K

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loop within conference
by excelsio＠gmx.net 09 Apr '10

09 Apr '10

Hi, I setup a conference following the hints on: http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR So far, I can dial in: - I´m told to enter the conference room number => ok - I´m told to enter the moderator pin for that conference room => ok But then I´m again told to enter the conference number and again the key for it and again and again and again. Let´s see output: 2010-04-09 14:30:15.768367 [NOTICE] mod_spidermonkey.c:2066 Channel [sofia/external/1333(a)188.168.2.1] has been answered 2010-04-09 14:30:15.768367 [INFO] conf-ivr.js:14 Prompting for Conference Room... 2010-04-09 14:30:19.621485 [INFO] conf-ivr.js:14 Conference Room Digits Collected = [666#] 2010-04-09 14:30:19.621485 [INFO] conf-ivr.js:14 Room [666] is valid, it is called [888] 2010-04-09 14:30:20.441945 [INFO] conf-ivr.js:14 Prompting for Moderator Password... 2010-04-09 14:30:21.682133 [INFO] conf-ivr.js:14 Password Digits Collected = [777#] 2010-04-09 14:30:21.682133 [INFO] conf-ivr.js:14 This is the first person in the conference 2010-04-09 14:30:24.721511 [INFO] conf-ivr.js:14 Entering conference room [666] as a [MODERATOR] 2010-04-09 14:30:27.142215 [NOTICE] switch_ivr.c:1447 Transfer sofia/external/1333(a)188.168.2.1 to XML[conf-room--666@normal-conference+flags{moderator}@public] 2010-04-09 14:30:27.146007 [INFO] mod_dialplan_xml.c:418 Processing 1333 Test->conf-room--666@normal-conference+flags{moderator} in context public 2010-04-09 14:30:27.162559 [INFO] conf-ivr.js:14 Prompting for Conference Room... 2010-04-09 14:30:32.382171 [INFO] conf-ivr.js:14 Conference Room Digits Collected = [666#] 2010-04-09 14:30:32.382171 [INFO] conf-ivr.js:14 Room [666] is valid, it is called [888] 2010-04-09 14:30:33.202121 [INFO] conf-ivr.js:14 Prompting for Moderator Password... 2010-04-09 14:30:36.101702 [INFO] conf-ivr.js:14 Password Digits Collected = [777#] 2010-04-09 14:30:36.101702 [INFO] conf-ivr.js:14 This is the first person in the conference 2010-04-09 14:30:39.141451 [INFO] conf-ivr.js:14 Entering conference room [666] as a [MODERATOR] 2010-04-09 14:30:41.562049 [NOTICE] switch_ivr.c:1447 Transfer sofia/external/1333(a)188.168.2.1 to XML[conf-room--666@normal-conference+flags{moderator}@public] => So, there´s obviously a kind of loop. But where does this occur? Also, I have this behavior, when I enter no password. In that case I´m a normal user: 2010-04-09 14:44:56.355377 [INFO] mod_dialplan_xml.c:418 Processing 1333 Test->3975 in context public 2010-04-09 14:44:56.377115 [NOTICE] mod_spidermonkey.c:2066 Channel [sofia/external/1333(a)188.168.2.1] has been answered 2010-04-09 14:44:56.377115 [INFO] conf-ivr.js:14 Prompting for Conference Room... 2010-04-09 14:45:00.322060 [INFO] conf-ivr.js:14 Conference Room Digits Collected = [666#] 2010-04-09 14:45:00.322060 [INFO] conf-ivr.js:14 Room [666] is valid, it is called [888] 2010-04-09 14:45:01.141552 [INFO] conf-ivr.js:14 Prompting for Moderator Password... 2010-04-09 14:45:03.542187 [INFO] conf-ivr.js:14 Password Digits Collected = [#] 2010-04-09 14:45:03.542187 [INFO] conf-ivr.js:14 This is the first person in the conference 2010-04-09 14:45:06.581903 [INFO] conf-ivr.js:14 Entering conference room [666] as a [USER] 2010-04-09 14:45:06.581903 [INFO] conf-ivr.js:14 Moderator is NOT present in the room 2010-04-09 14:45:09.222209 [NOTICE] switch_ivr.c:1447 Transfer sofia/external/1333(a)188.168.2.1 to XML[conf-room--666@normal-conference@public] Error in my_thread_global_end(): 1 threads didn't exit 2010-04-09 14:45:14.225453 [INFO] mod_dialplan_xml.c:418 Processing 1333 Test->conf-room--666@normal-conference in context public 2010-04-09 14:45:14.242444 [INFO] conf-ivr.js:14 Prompting for Conference Room... 2010-04-09 14:45:18.341791 [INFO] conf-ivr.js:14 Conference Room Digits Collected = [666#] 2010-04-09 14:45:18.341791 [INFO] conf-ivr.js:14 Room [666] is valid, it is called [888] 2010-04-09 14:45:19.162446 [INFO] conf-ivr.js:14 Prompting for Moderator Password... 2010-04-09 14:45:21.722027 [INFO] conf-ivr.js:14 Password Digits Collected = [#] 2010-04-09 14:45:21.722027 [INFO] conf-ivr.js:14 This is the first person in the conference 2010-04-09 14:45:24.762133 [INFO] conf-ivr.js:14 Entering conference room [666] as a [USER] 2010-04-09 14:45:24.762133 [INFO] conf-ivr.js:14 Moderator is NOT present in the room 2010-04-09 14:45:27.401371 [NOTICE] switch_ivr.c:1447 Transfer sofia/external/1333(a)188.168.2.1 to XML[conf-room--666@normal-conference@public] Error in my_thread_global_end(): 1 threads didn't exit 2010-04-09 14:45:32.405295 [INFO] mod_dialplan_xml.c:418 Processing 1333 Test->conf-room--666@normal-conference in context public 2010-04-09 14:45:32.423449 [INFO] conf-ivr.js:14 Prompting for Conference Room... => Again that loop So, where´s the mistake I made? Michael -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

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xml_curl loading/unloading user specific gateways.
by Ghulam Mustafa 09 Apr '10

09 Apr '10

Hi, is there a way to load a user specific gateway on sofia::register event, and killgw/unregister gateway on sofia::unregister event, i know this can be accomplished in many ways. but i am looking for a cleaner and proper way, anyone? thanks and best regards, -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity(a)ekiga.net mail: mustafa.pk(a)gmail.com web: cyrenity.wordpress.com

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Setting channel variables for FIFO-originated calls
by Ken Treis 08 Apr '10

08 Apr '10

I'm still pretty new to FreeSWITCH, but learning my way around. I've been trying to figure out how to use a different filename for each call in an on-hook agent fifo setup. I've been setting fifo_record_template in the originate string of the member definition, e.g. > {member_wait=nowait,fifo_record_template=$${base_dir}/recordings/fifo.wav}sofia/gateway/sip.jnctn.net/18005551212 This works, but as soon as I put a (non-preprocessor) variable in there, I get a complaint that my variable contains a variable: "Invalid data (${fifo_record_template} contains a variable)". But the code in mod_fifo.c seems to expect to find variables in there: > if (record_template) { > expanded = switch_channel_expand_variables(other_channel, record_template); > switch_ivr_record_session(session, expanded, 0, NULL); > } So my only conclusions at this point are that either: 1. The validation rule that gives me the "contains a variable" is being overly paranoid, or 2. There must be some other way to set this variable. I've chased #2 for a little bit, but apparently since FIFO-originated calls are created without any linked session (first arg to switch_ivr_originate is NULL), I can't export the variable from the original channel either. Is there some other way to set channel variables on the calls the FIFO originates? -- Ken Treis Miriam Technologies, Inc.

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mod_conference with H.323 Video
by Jeremy Shaffner 08 Apr '10

08 Apr '10

Hello, Is this known to work at all? -Jeremy

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Re: [Freeswitch-users] Problem sending DTMF using an FXS channel
by devel＠thom.fr.eu.org 08 Apr '10

08 Apr '10

Log is in http://pastebin.freeswitch.org/12665 Dialplan is following (included in default freeswitch) : <extension name="Outgoing FXO Free"> <condition field="destination_number" expression="^(0[1234589].*)$">  <action application="set" data="effective_caller_id_name=Francois Legal"/> <action application="set" data="effective_caller_id_number=0950640430"/> <action application="set" data="ringback=${hold_music}"/> <action application="set" data="transfer_ringback=${hold_music}"/> <action application="bridge" data="openzap/2/2/$1|sofia/gateway/freephonie.net/$1|openzap/2/1/$1"/> </condition> </extension> I forgot to mention that Im using sangoma A400 with HW DTMF detection. François De : freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Michael Collins Envoyé : jeudi 8 avril 2010 19:59 À : freeswitch-users(a)lists.freeswitch.org Objet : Re: [Freeswitch-users] Problem sending DTMF using an FXS channel Pastebin your dialplan config that handles this call as well as a debug trace from the console. -MC On Thu, Apr 8, 2010 at 10:03 AM, <devel(a)thom.fr.eu.org> wrote: Hello, Im having trouble with calls to remote IVR using DTMF, when the A leg is an FXS port. What happens is when the key is pressed on the phone, the DTMF is sent inband to the callee party as voice, but also detected by freeswitch and so resent by freeswitch to the callee party. This results in unusability of called IVR. Is there any setting that could be used to prevent freeswitch from detecting DTMF and/or prevent freeswitch from resending the DTMF. Thanks François _______________________________________________ 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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G.729 Info On Wiki
by Michael Collins 08 Apr '10

08 Apr '10

Hi folks, Just a quick FYI, there is g.729 license info now on the wiki: http://wiki.freeswitch.org/wiki/Mod_com_g729 Thanks to Steve Ayre for starting this page. If you have g.729 knowledge to add to the page please do so. Also, the INSTALL.txt file has been updated with a bit more detail: http://files.freeswitch.org/g729/INSTALL.txt Happy transcoding! -Michael

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Another IRC day in #freeswitch!
by Michael Collins 08 Apr '10

08 Apr '10

Hey everyone, come join us in #freeswitch on irc.freenode.net! We are going for 250 users. (No sockpuppets, please.) If you don't have an IRC client then visit http://www.freeswitch.org and click the Join Us On IRC link to use the Web IRC applet. See you in channel! -MC (IRC: mercutioviz)

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Re: [Freeswitch-users] FS cannot produce any log file
by Michael Collins 08 Apr '10

08 Apr '10

Make absolutely certain that there isn't a permissions issue with FreeSWITCH writing to the /var/log/ directory. -MC On Thu, Apr 8, 2010 at 7:02 AM, Bi,Xianchun <xianchun.bi(a)quanshi.com> wrote: > Hi, > > > > I have installed FreeSWITCH in my test server, the OS is Ubuntu 8.04 Server > > > > In module configuration file conf/autoload_configs/modules.conf.xml, > log_file module is set active. > > > > <load module="mod_logfile"/> > > > > And in the log_file configuration file > conf/autoload_configs/logfile.conf.xml, the settings are as following: > > > > <configuration name="logfile.conf" description="File Logging"> > > <settings> > >  > > <param name="rotate-on-hup" value="true"/> > > </settings> > > <profiles> > > <profile name="default"> > > <settings> > >  > > <param name="logfile" value="/var/log/freeswitch.log"/> > >  > > <param name="rollover" value="10485760"/> > >  > >  > > </settings> > > <mappings> > >  > > <map name="all" value="debug,info,notice,warning,err,crit,alert"/> > > </mappings> > > </profile> > > </profiles> > > </configuration> > > > > > > But the logfile never appers. I also try using touch to create a dummy > file, but the file never grow. > > Restarting FreeSWITCH has no effect. > > What wrong with my settings? > > Anyone could give any suggestion? > > Thanks in advance. > > > > _______________________________________________ > 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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Unable to call out on FXO interface -- can receive calls
by Chang Zhao 08 Apr '10

08 Apr '10

Hi, I have a TDM400 card with 4 fxo interfaces on it, I can receive calls just fine, but when trying to make an outbound call, the following shows up: 2010-04-08 17:23:46.843723 [DEBUG] ozmod_analog.c:655 Detected tone DIAL on 1:1 2010-04-08 17:23:46.843723 [DEBUG] mod_openzap.c:1530 got FXO sig 1:1 [TONE_DETECTED] 2010-04-08 17:23:46.843723 [WARNING] mod_openzap.c:1577 Unhandled msg type 9 for channel 1:1 2010-04-08 17:23:46.843723 [ERR] ozmod_analog.c:671 No Digits to send! 2010-04-08 17:23:46.843723 [DEBUG] ozmod_analog.c:672 Changing state on 1:1 from DIALING to BUSY So it appears that it's detecting a dialtone, but then what's msg type 9? Does anyone have any insights? Using latest GIT head version and CentOS 5.4 64bit. Thanks and regards, CZ

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