- FreeSWITCH-users - FreeSWITCH Mailing Lists

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

5 8

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

2 1

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

1 0

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

2 7

Event did not return anything after update.
by DJB International 07 Jun '10

07 Jun '10

After updating from 170404a to 9f73ddd, I could not see anything in event message like before. I connect to mod_event_socket by perl script below: require ESL; my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); $con->events("plain", "all"); while($con->connected()) { #my $e = $con->recvEventTimed(100); my $e = $con->recvEvent(); if ($e) { #print $e->serialize(); my $h = $e->firstHeader(); print "\n--------------------RECEIVED--------------------\n"; while ($h) { printf "Header: [%s] = [%s]\n", $h, $e->getHeader($h); $h = $e->nextHeader(); } } } I wonder whether there is any changes that I need to adjust after the update. Thank you, Dorn B.

2 9

Registration Update of Contact List
by ro ra 07 Jun '10

07 Jun '10

Hi, what do i need to change on my configuration files to make this scenario work: user1 --------- REGISTER ------------> Freeswitch user1 <-------- 401 UnAuthorized ------- Freeswitch user1 --------- REGISTER w/ Auth -----> Freeswitch user1 <------- 200 OK ------------------------ Freeswitch ----------------------------5 secs delay---------------------------------- user1 ----------- REGISTER w/ Auth(Update) -----> Freeswitch user1 <-------- 200 OK ----------------------- Freeswitch or is this possible with freeswitch...? thanks

2 3

LUA - ** - Disconnect the call or stop call progress [ ringing or slient] and jump to the next statement
by Aloysius Lloyd 07 Jun '10

07 Jun '10

Hi All, I would like to implement a feature code ** - Disconnect the leg B or stop call progress [ ringing or slient] and jump to the next statement. Here is the Lua code I am using to bridge two calls. How to implement the feature code like ** for this purpose. session:preAnswer(); digits = session:playAndGetDigits(10, 20, 3, 5000, "#", "enter-dest.wav", "invalid-digits.wav", "\\d+|\\*"); obSession = freeswitch.Session("sofia/gateway/voipms/"..digits,session) if obSession:ready() then obSession:execute("sched_hangup","+60 alloted_timeout"); freeswitch.bridge(session, obSession); end Thanks Lloyd

2 1

What happen when leg-a & leg-b are on SIP/TCP and one get a socket disconnection during a call?
by Fabio Pietrosanti (naif) 07 Jun '10

07 Jun '10

Hi, sorry for using the so big subject, i am experiencing some strange behaviour i am going to enter more in depth debugging in the next weeks. I have the following situation: - A call getting established - both peers are using SIP/TLS over TCP - leg-b get disconnected at TCP level (such as WiFi disconnection or just a TCP hard reset by a NAT device reloading it's rules) In such condition i noticed, but still need detailed investigation, that FS does not detect that leg-b TCP socket died, thus notifying immediately leg-a that the call cannot be completed. FS instead, it seems from log/tcpdump observation, wait until the various SIP timer expire in order to provide back to leg-a error that the call cannot be completed. If it's that way, it would be a nice improvement, when a SIP/TCP or SIP/TLS client get a TCP disconnect, to immediately react on the other leg of the call by notifying the proper SIP error, because the layer4 (SIP) connection is lost due to a layer3 (TCP) connection break. Is this something feasible/reasonable? Fabio

2 1

ccxml/vxml apps needed for testing
by Jan Berger 06 Jun '10

06 Jun '10

Hi, If someone has ccxml and/or vxml apps they are willing to contribute for testing please send them to me off-list. Please include info about what platform they are tested on. The app's will only be used for conformance testing - currently only towards a debug-simulator. jan

1 0

Detect events other than DTMF during .streamFile
by Jon Davies 06 Jun '10

06 Jun '10

Hi I was wondering if it was possible to detect events other than DTMF that are fired during .streamFile?. DTMF events are received by my callback without issue, but if I try to use speech detection via mod_pocketsphinx the callback doesn't get fired. If this is not possible, are there any other options available to me in order to detect speech input during the playback of a file? Kind Regards jdee

1 0