June 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_conference Wiki - lists
by David Swardstrom 07 Jun '10

07 Jun '10

I have been looking at the mod_conference Wiki page and have checked some of it's information against the current code in mod_conference.c. In particular, I have been interested in the “list” capability which is supported by conference_list(). The following comments will be used to change the “Mod Conference” Wiki page: This applies to the API reference, list entry: <current entry> Output: First line Conference <conference name> (<member_count> member ) Following lines are csv separated list for each conference leg with the following items: id of participiant (starts at 1 after FS restart) Register string of participiant Uuid of participiants call leg Cid number Cid name Status (hear|speak|floor) Volume In Volume Out Energy Level </current entry> <corrected entry> Output: First line Conference <conference name> (<member_count> member[s][ locked]) Following lines are csv separated list for each conference leg with the following items: id of participant (starts at 1 after FS restart) Register string of participant Uuid of participants call leg Caller id number Caller id name Status (hear|speak|talking|video|floor) Volume In Volume Out Energy Level Notes: First line: “locked” reflects the lock/unlock status of the conference. Status: “hear” reflects the mute/unmute status of the member. Status: “speak” reflects the “deaf /undeaf” status of the member. Status: “talking” indicates that the input channel is providing some amount of sound energy. Status: “video” ??? Status: “floor” indicates that this member currently owns the floor </corrected entry> Does anyone have any comments I should add for floor? Is there anything significant to owning the floor? How about video? It seems that video and video conferencing is a capability that is desired and some support is provided. But is it really working? Does anyone use it? Is there an assumption in conferencing that video is currently sourced from only one place? Regards, David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom

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RTP Proxy
by Gustavo Espeche 07 Jun '10

07 Jun '10

Hi some one know an rtp proxy that work with freeswitch. Thanks. Gustavo Espeche www.easyipcall.com

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FS "Holds On" to ITSP's 183 for 2 Sec.Before Relaying to Phone?
by Alan Frisch 07 Jun '10

07 Jun '10

Hi, Just tinkering around with FreeSWITCH pretty much using the default configuration, with some customization to connect my Polycom 430 and my ITSP. The problem I am having is that the first few seconds of calls (or early media if available) have no audio. When I NGREP'd the SIP dialog between my Phone <> FS and FS <> ITSP with timestamps, I can see that FreeSwitch is holding onto the 183 and subsequent 200 OK for about 2 seconds before relaying it onto the Polycom. FWIW, server is an Intel Atom 330. Any idea why this is... I am sure it must be a configuration issue somewhere? I did not have this issue with Asterisk, so I am hoping to troubleshoot it and move on with FreeSWITCH Thanks. AF.

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get your job aguinst freeswitch
by Meftah Tayeb 07 Jun '10

07 Jun '10

hello list, i am meftah tayeb, a blind person from algeria that was using freeswitch sunse 1.0.1 release i started firstly with asterisk, that was the very strangett voip application in my life and thank to miconda that redirected me to freeswitch, from asterisk in #openser in 2008 i started learning voip basic in dec 2008 thank to the #freeswitch folk that teached me all this, including firstly anthm, Michael S Collin, brian (BKW), sekil the nice GUI and all other in ogust 2009, algeria telecom and the algeria gouvernmant started blocking sip traffic i was not using it for voip business, but, honestly, just to connect to the public freeswitch conference and the weekely voip users conference i start a complain in my local city, no reply from AT i decided to go to the general office, because i can't use my PC without SIP i got the general directore secrutary ok, he receyved me and heare me saying why you are blocking sip? so he asked me why you need sip? do you do voip business without autorisation? i explaned to him my actual situation and he say: ok, no problem i will open the sip for you, but conditionaly the next mondey you will be here i say ok no problem so, i returned to my home and i see something new: 1. a static ip address linked to my ADSL account 2. sip completly open ok, next monday i was in the general office and i meet the gebneral directotore surprisingly he asked me: do you have a job? i say no but he say yes, you have, tel me i say no aguin and he say you work for algeria telecom ;) he surprised me with this now, he gave me a: good job free house free care with driver so please say thank to the freeswitch project especialy the owner and try to donate to him a much a pocible thank you -- Meftah Tayeb algérie télécom SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz

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Re: [Freeswitch-users] Freeswicth nms
by suvirkumar＠gmail.com 07 Jun '10

07 Jun '10

Budi, Try Nagios for network monitoring Regards Suvir ------Original Message------ From: Budi wibowo Sender: freeswitch-users-bounces(a)lists.freeswitch.org To: freeswitch-users(a)lists.freeswitch.org ReplyTo: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] Freeswicth nms Sent: Jun 3, 2010 06:26 Hi all I'm looking for nms that can do minimum alarm,fault, performance and statistic. Alarm could be sent via sms or email. I need the such solution due I have project to involve more than 10 fs box. Both free and commercial solution are ok Regards Budi _______________________________________________ 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 Sent via BlackBerry® from AIS

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Skypopen problem
by bakko 07 Jun '10

07 Jun '10

Hi, I'm trying tu install skypopen on my centOS server witout success. This is my configuration: virtual xen server on linode.com centos 5.5 32 bit freswitch latest trunk xen-devel given from linode If i try to compile the alsa-driver-1.0.20 modifing the files and lines indicates in the wiki when i do make this is the result: /usr/src/alsa-driver-1.0.20/acore/oss/pcm_oss.c: In function 'snd_pcm_oss_open_file' /usr/src/alsa-driver-1.0.20/acore/oss/pcm_oss.c:2304: error: 'fmode_t' undeclared (first use in this function) /usr/src/alsa-driver-1.0.20/acore/oss/pcm_oss.c:2304: error: (Each undeclared identifier is reported only once /usr/src/alsa-driver-1.0.20/acore/oss/pcm_oss.c:2304: error: for each function it appears in.) If i try to compile the alsa-driver-1.0.20 without any modification and then compile dummy.c, when i try to load snd-dummy module this is the result modprobe snd-dummy WARNING: Error inserting snd_page_alloc (/lib/modules/2.6.18.8-linode22/misc/acore/snd-page-alloc.ko): Invalid module format WARNING: Error inserting snd_timer (/lib/modules/2.6.18.8-linode22/misc/acore/snd-timer.ko): Invalid module format WARNING: Error inserting snd_pcm (/lib/modules/2.6.18.8-linode22/misc/acore/snd-pcm.ko): Invalid module format FATAL: Error inserting snd_dummy (/lib/modules/2.6.18.8-linode22/misc/drivers/snd-dummy.ko): Invalid module format modprobe -f snd-dummy WARNING: Error inserting snd_page_alloc (/lib/modules/2.6.18.8-linode22/misc/acore/snd-page-alloc.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.18.8-linode22/misc/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.18.8-linode22/misc/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_dummy (/lib/modules/2.6.18.8-linode22/misc/drivers/snd-dummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) Any idea? BR - Andrea Sannucci -

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outbound socket problem / fs_ivrd general protection, segfault
by Christian Löschenkohl 07 Jun '10

07 Jun '10

hello list recently i have problems with outbound socket scripts (written in php) dialplan looks like this <condition field="destination_number" expression="^(\d+)$" break="on-true"> <action application="set" data="ivr_path=/opt/freeswitch/scripts/script1.php"/> <action application="socket" data="127.0.0.1:8084 full"/> ... do something ... <action application="set" data="ivr_path=/opt/freeswitch/scripts/script2.php"/> <action application="socket" data="127.0.0.1:8084 full"/> <action application="hangup"/> </condition> script1.php is executed script2.php fails (see syslog msg) syslog says fs_ivrd[11971] general protection ip:7f2f7b94bbf5 sp:7fffffffa148 error:0 in libpthread-2.7.so[7f2f7b942000+16000] or fs_ivrd[12037]: segfault at a006c6c7576 ip 7f2f7b94bbf5 sp 7fffffffa148 error 4 in libpthread-2.7.so[7f2f7b942000+16000] uname -a Linux vts02.vie.xpirio.net 2.6.26-2-amd64 #1 SMP Tue Mar 9 22:29:32 UTC 2010 x86_64 GNU/Linux version FreeSWITCH Version 1.0.6 (svn-exported) does anybody have similar problems? br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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Freeswitch kills my home voip phone
by Oliver Schenk 07 Jun '10

07 Jun '10

Hi All, I'm a total freeswitch noob at the moment. I was playing at home with FreeSwitch on Windows compiled from source. I have a router at home that registers with my ISP's voip SIP server. My hard phone is connected to that router to make calls. Whenever I have FreeSwitch running on my PC (which is on the same LAN as the router) I lose my VOIP connection to the ISP and I can't re-register until I reboot my internet connection. This happens straight out of the box. Why is this happening? Also while freeswitch is running I can't seem to make any calls ... is Freeswitch hijacking my connection? Thanks, Oliver

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Fwd: [Kde-accessibility] OpenTTS 0.1 released.
by Meftah Tayeb 07 Jun '10

07 Jun '10

-------- Message original -------- Sujet: [Kde-accessibility] OpenTTS 0.1 released. Date : Sun, 6 Jun 2010 13:13:40 +1000 De : Luke Yelavich <themuso(a)themuso.com> Pour : kde-accessibility(a)kde.org I am proud to announce the very first release of OpenTTS, version 0.1. Thanks to everybody who made this release possible, you know who you are. The first release series is focused on delivering a text to speech framework for the GNOME orca screen reader, that is the same, and hopefully better, than the deprecated gnome-speech solution that has been in use ever since the GNOME accessibility project's inception. The OpenTTS development team is also planning on making further usability improvements, which will aid both system administrators, and end users alike. PLEASE NOTE: This release is NOT considered stable. It is recommended that only those who are willing to test, and report bugs use this first release of OpenTTS. Since we are aiming to provide a usable solution for orca, our aim is to deliver a stable release at the time GNOME 3 officially ships. Between now and September therefore, we hope to make regular monthly releases for the community to track our work, test, and give feedback. These releases will not necessarily be considered stable, but will give users a snapshot they can use, which is free of major show stopper bugs. Release notes: * OpenTTS can be used as a system wide service, however at the moment, this is slightly extra work. The OpenTTS development team hopes to make OpenTTS as a system service much easier to use in the near future. Where can I get it? OpenTTS 0.1 is available from the following locations: Tarball: http://files.opentts.org/opentts/opentts-0.1.tar.gz git: git://git.opentts.org/opentts.git Getting involved: We are always willing to accept any help, no matter where it comes from, or what skills you bring to the table. If you would like to contact us and get involved, please do so in the following ways: * OpenTTS project mailing lists, at http://lists.opentts.org. Please sign up to the opentts-dev mailing list for development discussions. Moderate traffic, low signal to noise ratio. There is also opentts-users, for user questinos, and support. Extremely low traffic, currently low signal to noise ratio. * Reporting/triaging bugs on the OpenTTS issue tracker, available at http://www.opentts.org. Bugs can also be reported on the mailing lists, if you prefer. We welcome any and all contributions. Changelog: For those interested, below you will find a shortlog of work that has been done since the project was started just over 2 months ago: Andrei Kholodnyi (46): moved spd_audio_endian to AudioID moved log_level from spd_audio.h to spd_audio.c and rename it moved pulse.c from spd_audio.c to Makefile.am moved alsa.c from spd_audio.c to Makefile.am moved oss.c from spd_audio.c to Makefile.am moved libao.c from spd_audio.c to Makefile.am moved nas.c from spd_audio.c to Makefile.am moved AudioID alloc into plugins move pulse specific data from AudioID to pulse.c move nas specific data from AudioID to nas.c move oss specific data from AudioID to oss.c move alsa specific data from AudioID to alsa.c remove AudioOutputType split spd library and spd plugin headers added plugin name to audio plugin structure move audio static plugins initialization into static_plugins.c moved module_main.c from the dummy.c to Makefile.am moved module_main.c from cicero.c to Makefile.am moved module_main.c from espeak.c to Makefile.am moved module_main.c from festival.c to Makefile.am moved module.main.c from flite.c to Makefile.am moved module_main.c from generic.c to Makefile.am moved module_main.c from ibmtts.c to Makefile.am moved module_main.c from ivona.c to Makefile.am moved fdsetconv.c from module_utils.c to Makefile.am added pkgconfig support moved debug defines to libspeechd.c update pkg-config support added dynamic load of audio plugins fix module_dbgfile memory leak fix rep_line memory leak replace EPunctMode with SPDPunctuation replace ECapLetRecogn with SPDCapitalLetters replace ESpellMode with SPDSpelling replace EVoiceType with SPDVoiceType replace VoiceDescription with SPDVoice replace ENotification with SPDNotification replace int ssml_mode with SPDDataMode replace-int-priority-with-SPDPriority moved EMessageType to SPDMessageType in opentts_types.h moved SPDMsgSettings from fdset.h to module_utils.h fix refs to fdset.h, general text review fix p_client_socket memory leak fix o_buf memory leak fix o_buf memory_leaks in parse.c fix make distcheck Christopher Brannon (113): Prevent automake from failing when we have no ChangeLog. Added spd_audio_get_playcmd function. Removed SPDAudioSettings Add missing include directives. Use const * consistently for function tables. Fix memory leaks in module_audio_init_spd. Fix copyright oversight. Give Andrei credit in the AUTHORS file. Add prototypes for module_utils_addvoice.c. Add a prototype for festival_read_response. Fix calls to ivona_get_msgpart. Declare functions from ivona_client.c in a header file. Possibly include sndfile.h in ivona_client.c. Fix inclusion of config.h for two files. Add ivona_client.h to Makefile.am. Remove use of strdup in do_list_voices. Change LIST VOICES in the internal module protocol. Use glib consistently in command handlers. Don't redefine PACKAGE and VERSION. Reorganized options.c and options.h for the "say" client. Cleanup options.c and options.h for the server. Stop using xfree in the audio subsystem. Fix link order in the modules. Use ltdl to handle static plugins. Use glib allocation in the server. Use glib allocation in the modules. Use glib allocation in the audio subsystem. Use glib allocation in the common library. Surround calls to OL_RET with braces, and add semicolons. Remove some complex macros. Simplify a macro in parse.c. Indent remaining files. Properly indent header files. Replace calls to free with g_free. Fix segfault when compiled against libao 1.0.0. Remove redundant library from modules/Makefile.am. Add new code for generating timestamps. Rename the Python bindings, including the speechd_config suite. Fix a mistake in the python subdirectory. Use automake in the session subdirectory. Convert the python subdirectory to use the autotools. spd_audio_endian should be id->format. Use otts_getline instead of getline. Fix module_add_config_option in module_utils.c. Change the way that lines are read in the server. Do not include the common library in libopentts. Reopen the connection to pulseaudio when the audio format changes. Add an extra newline when writing to .profile. Add the Unix socket communication method, and use it by default. The communication method is now configurable. Support Unix sockets in the C client API. Autospawn is controlled by an argument to spd_open2. Make autospawn optional for clients. The daemon no longer checks SPEECHD_PORT. Don't require the ~/.speech-dispatcher directory to already exist. Autospawn mechanism fine-tuning Redirect stderr when autospawning in the Python library. Check for the existence of the pid file before handling autospawn. Update otts-conf to generate config for Unix sockets. Do not require ~/.speech-dispatcher to exist. Remove some unused options from the configure script. Put Unix sockets in the home directory, and respect SPEECHD_SOCK. Don't try to compile OSS support if OSS is not available. Fix compilation on Mac OS X. Correct the loop boundary in a convenience function. Properly handle the return value of spd_sayf in the otts-say client. Improved portability checks. Don't store pointers to strings that dotconf will free. Move spd_audio.c and spd_audio.h to the modules subdirectory. Fix prefixes of entry functions in the audio modules. Adjust init_settings_tables to account for renaming of constants. Add constants for reporting errors. Install opentts_types.h. More renaming. Remove the "Future Design" section. More replacement. Change pathnames in the documentation. Rename speech-dispatcher.texi to opentts.texi. Let the otts-say client speak messages that begin with dash. Additional renaming in the otts-say client. Change names of files. Fix a bug in libopentts.c. Replace many references to Speech Dispatcher. Fix two memory leaks in the parse_set function. Use proper comparisons while traversing the history list. Prevent another macro from storing pointers to dotconf's strings. Fix: two command-line options had no effect. Fix a nasty off-by-one bug in the Festival module. Bugfix: ALSA now resumes when suspended. Fix an error-handling clause in alsa_play. Change the test-runner program to use Unix sockets. Update the documentation to reflect the code. Replace the SPEECHD_* environment variables with OPENTTSD_*. Add an introduction to the C API section of the manual. Only change endianness for 16-bit audio. Set correct endianness during libao playback. The function flite_set_voice should be a no-op. Discuss our history in the documentation. Fix the help and version messages for the server. Change speechd to openttsd in log messages. Change speechd to openttsd in error messages and comments. Amend authorship and copyright info for the docs. otts-say.texi needs more authorship and copyright info. Cast return value of lt_dlsym to the proper type. Update the AUTHORS file. Document the process of building ibmtts for 64-bit. Unknown voice types should map to SPD_NO_VOICE. Refactor spd_open2 and fix spd_close. Properly handle autospawning in the Python library. Break out of a loop in pulse_play when an error occurs. Wait for the auto-spawned server process to terminate. Use the same parameter to listen() for both Unix and TCP sockets. Break out of a loop in libao_play when an error occurs. Jason White (1): fix otts-conf default port José Vilmar Estácio de Souza (1): memory leak Luke Yelavich (15): Initial OpenTTSd git repository creation based on unofficial 0.6.68 release of speech-dispatcher session - Re-add accidentally lost Makefile.in Re-add lost python build file Remove reference to ChangeLog file in top directory Reset version to 0.0 Re-add yet another lost file from the openttsd repo transition Set correct audio endianness for audio playback using pulseaudio server - change last references to speech-dispatcher config file directories to opentts Change all other ~/.speech-dispatcher references to ~/.opentts Change more speech-dispatcher references in the pulseaudio output code Rename speechd/spd-say/Speech Dispatcher to their OpenTTS equivalents in otts-conf Don't refer to non-existant fdset.h file in src/common Distribute fdset.h in the tarball Link against libcommon.la in the build dir, not the src dir Release 0.1 Rui Batista (1): Renamed Speech Dispatcher naming to Opentts in INSTALL and minor cleanups on that file Steve Holmes (2): Updated documentation to reflect OpenTTS name. Updated opentts.desktop file. Trevor Saunders (24): change calls to signal to sigaction clean up module_utils update docs rename variables and functions in the server rename spdsend to otts-send cleanup documentation for otts-send rename files in src/clients/send/ change references to spdsend to otts-send update comment in libao.c rename spd_audio_log_level to audio_log_level renamed several audio types rename some audio functions change LIBSPEECHD_DEBUG macro to LIBOPENTTS_DEBUG rename SPEECHD_DEBUG to OPENTTSD_DEBUG and move it to openttsd.h rename SpeechdSocket to openttsd_sockets renamed multiple functions update several comments rename options.home_speechd_dir to options.opentts_dir rename SPEECHD_OPTION_xxx macros change the include guard in openttsd.h to use openttsd instead of speechd change TSpeechDQueue to queue_t rename TSpeechDSock to sock_t rename speechd_queue_alloc to queue_alloc rename SPEECHD_DEFAULT_PORT to OPENTTSD_DEFAULT_PORT William Hubbs (83): rename configure.in to configure.ac fixed typo updated extra_dist in main makefile update documentation Makefile rearranged the api and client directories fix issues with installation only build test binaries when make check is run clean up several makefiles distribute the API's for common lisp and guile started cleaning up configure.ac detect sound systems before speech synthesizers define installed headers correctly don't declare libraries in LDFLAGS more makefile cleanup fixed an autotools conflict include config.h in all sources move the code in intl to a convenience library tests should use the convenience library remove PACKAGE and VERSION definitions from server's makefile clean up ignore patterns cleaned up EXTRA_DIST for modules added more ignore patterns convert libsdaudio to a convenience library configuration system update remove rule for html documentation fixed a comment remove old speech dispatcher announcement disable session integration support by default clean up the modules makefile removed extra_dist from src/audio/Makefile.am install spd_audio_plugin header file rename the C API library tests should link against libopentts fix include in the say client fixed an include in the guile bindings updated the header in libopentts.h updated an include in libopentts.c rename directories from speech-dispatcher to opentts audio subsystem updates move the modules sub directory rename ottslibdir to audiodir fixed variable in audio makefile include spd_audio_plugin.h from spd_audio.c instead of spd_audio.h. Revert "include spd_audio_plugin.h from spd_audio.c instead of spd_audio.h." ran code through Lindent python build updates session build fixes do not install Czech documentation fix declaration of festival_connection_crashed fix include in spd_audio.c fix declaration of current_index_mark convert global variable declarations to externs consolidate global variables in modules remove DBG macro from modules remove the prefix from the module names rename the daemon make better use of conditionals in modules makefile rename fixes move public header files to common include directory build system updates remove the session directory add configure option to disable the installation of the python bindings remove prefix from audio plugin names Revert "remove prefix from audio plugin names" removed several change logs remove prefix from names of audio plugins fix includes of libopentts.h and opentts_types.h in the say client and api change main config file to openttsd.conf fix config makefile remove duplicate include directive in openttsd.conf move desktop autostart file to config directory clean up config files fix permissions on the socket permission fixes fix typo remove dotconf from pkg-config file fix Makefile substitutions clarify the contact information in the SSIP document fix copyright assignment request make configure fail when necessary libraries are not on the system build fixes for the audio subsystem remove ability to disable autospawn from server fix implicit declaration of g_unlink jose vilmar estacio de souza (1): possible memory leaks tbsaunde (1): refactor module_main to use disatch_cmd instead of macros Regards Luke Yelavich OpenTTS Project lead. _______________________________________________ Opentts-dev mailing list Opentts-dev(a)lists.opentts.org http://lists.opentts.org/listinfo.cgi/opentts-dev-opentts.org _______________________________________________ kde-accessibility mailing list kde-accessibility(a)kde.org https://mail.kde.org/mailman/listinfo/kde-accessibility

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Hi
by Zuhair Raza 07 Jun '10

07 Jun '10

> Hi everyone > I am working on skype module,according to wiki I have successfuly installed > freeswitch and skype module with alsa driver. I am facing trouble at > creating clone a directory step, I am using centos 5.4 and skype static > version. First i enter command at my freeswitch box under > mod_skypopen/configs > > gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11 > > then I install xauth on server after that i ssh with x forwarding to my > server from an another linux desktop and open skype by typing > /usr/bin/skype, it launched skype client at Linux desktop but it didn't ask > for connecting with skypopen api, although it creates ".Skype" directory on > my server but when i load mod_skypopen it says could not find any skype > instance and when i typed skypopen_auth it also says no skype instance found > on X 0:0. Can anyone tell me where I have mistaken?? > > I also observed it connected sometimes but sometimes it does not. > -- > Regards, > Zuhair Raza > > > -- Regards, Zuhair Raza -- Regards, Zuhair Raza

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