- FreeSWITCH-users - FreeSWITCH Mailing Lists

Not recognizing voice in mod_pocketsphinx
by Thangappan.M 30 Jan '12

30 Jan '12

Dear all, I have installed the mod_pocketsphinx module successfully and enabled with FreeSWITCH. Copied the relevant files for pizza application and placed into appropriate directory. Made a call to "74992" and able to hear the voice. When I said "TAKEOUT' it was not recognizing my words. In the freeswitch CLI it shown the following messages. EXECUTE sofia/internal/1000(a)192.168.1.222 javascript(ps_pizza.js) 2010-09-06 13:18:48.023673 [DEBUG] sofia_glue.c:2594 AUDIO RTP [sofia/internal/1000(a)192.168.1.222] 192.168.1.222 port 32590 -> 192.168.6.60 port 8000 codec: 8 ms: 20 2010-09-06 13:18:48.024705 [DEBUG] switch_rtp.c:1182 Starting timer [soft] 160 bytes per 20ms 2010-09-06 13:18:48.025710 [DEBUG] sofia_glue.c:2774 Set 2833 dtmf send payload to 101 2010-09-06 13:18:48.026799 [DEBUG] sofia_glue.c:2779 Set 2833 dtmf receive payload to 101 2010-09-06 13:18:48.026799 [DEBUG] mod_sofia.c:636 Local SDP sofia/internal/ 1000(a)192.168.1.222: v=0 o=FreeSWITCH 1283726738 1283726739 IN IP4 192.168.1.222 s=FreeSWITCH c=IN IP4 192.168.1.222 t=0 0 m=audio 32590 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2010-09-06 13:18:48.028235 [DEBUG] switch_core_session.c:641 Send signal sofia/internal/1000(a)192.168.1.222 [BREAK] 2010-09-06 13:18:48.028235 [DEBUG] sofia.c:4153 Channel sofia/internal/ 1000(a)192.168.1.222 entering state [completed][200] 2010-09-06 13:18:48.029256 [NOTICE] mod_spidermonkey.c:2066 Channel [sofia/internal/1000(a)192.168.1.222] has been answered 2010-09-06 13:18:48.030269 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16@8000hz 1 channels 20ms 2010-09-06 13:18:48.123555 [DEBUG] sofia.c:4153 Channel sofia/internal/ 1000(a)192.168.1.222 entering state [ready][200] 2010-09-06 13:18:48.171705 [DEBUG] switch_rtp.c:2066 Correct ip/port confirmed. 2010-09-06 13:18:50.212417 [DEBUG] switch_ivr_play_say.c:1444 done playing file EXECUTE sofia/internal/1000(a)192.168.1.222 detect_speech(pocketsphinx pizza_order undefined) 2010-09-06 13:18:51.768030 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/1000(a)192.168.1.222 EXECUTE sofia/internal/1000(a)192.168.1.222 detect_speech(resume) 2010-09-06 13:18:51.774074 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16@8000hz 1 channels 20ms 2010-09-06 13:18:54.492023 [DEBUG] switch_ivr_play_say.c:1444 done playing file EXECUTE sofia/internal/1000(a)192.168.1.222 detect_speech(resume) EXECUTE sofia/internal/1000(a)192.168.1.222 detect_speech(resume) 2010-09-06 13:19:01.534474 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16@8000hz 1 channels 20ms Please help me..... -- Regards, Thangappan.M

3 3

building mod_opal fails on gentoo linux
by Tamer Higazi 29 Jan '12

29 Jan '12

Hi people! I have successfully built the latest freeswitch_snapshot but I am not capable to build "mod_opal". Can somebody tell me, what I did wrong? ptlib and opal are installed on my system. thanks for any support. here is the output: making all mod_opal Compiling /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp... Compiling /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp ... /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp: In constructor 'FSConnection::FSConnection(OpalCall&, FSEndPoint&, void*, unsigned int, OpalConnection::StringOptions*, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)': /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp:564:26: error: no matching function for call to 'OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, void*&, unsigned int&, OpalConnection::StringOptions*&)' /usr/include/opal/opal/localep.h:249:5: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&, void*) /usr/include/opal/opal/localep.h:242:1: note: OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&) /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)': /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp:1295:25: error: 'class OpalMediaPatch' has no member named 'OnStartMediaPatch' /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)': /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp:1417:25: error: 'class OpalMediaPatch' has no member named 'OnStartMediaPatch' make[5]: *** [mod_opal.lo] Fehler 1 make[4]: *** [all] Fehler 1 make[3]: *** [mod_opal-all] Fehler 1 make[2]: *** [all-recursive] Fehler 1 make[1]: *** [all-recursive] Fehler 1 make: *** [all] Fehler 2 and my system: tamer@office /storage/downloads/freeswitch-snapshot $ emerge --info opal ptlib Portage 2.1.10.44 (default/linux/amd64/10.0/desktop/gnome, gcc-4.5.3, glibc-2.13-r4, 3.0.6-gentoo x86_64) ================================================================= System Settings ================================================================= System uname: Linux-3.0.6-gentoo-x86_64-Intel-R-_Core-TM-2_CPU_6600_(a)_2.40GHz-with-gentoo-2.0.3 Timestamp of tree: Thu, 26 Jan 2012 23:30:01 +0000 app-shells/bash: 4.1_p9 dev-java/java-config: 2.1.11-r3 dev-lang/python: 2.7.2-r3, 3.1.4-r3 dev-util/cmake: 2.8.6-r4 dev-util/pkgconfig: 0.26 sys-apps/baselayout: 2.0.3 sys-apps/openrc: 0.9.8.2 sys-apps/sandbox: 2.5 sys-devel/autoconf: 2.13, 2.68 sys-devel/automake: 1.10.3, 1.11.1 sys-devel/binutils: 2.21.1-r1 sys-devel/gcc: 4.4.5, 4.5.3-r1 sys-devel/gcc-config: 1.4.1-r1 sys-devel/libtool: 2.4-r1 sys-devel/make: 3.82-r1 sys-kernel/linux-headers: 3.1 (virtual/os-headers) sys-libs/glibc: 2.13-r4 Repositories: gentoo freeswitch voyageur proaudio calculate x-liquidx ACCEPT_KEYWORDS="amd64" ACCEPT_LICENSE="*" CBUILD="x86_64-pc-linux-gnu" CFLAGS="-march=core2 -O3 -pipe" CHOST="x86_64-pc-linux-gnu" CONFIG_PROTECT="/etc /opt/openfire/resources/security/ /usr/lib64/fax /usr/share/config /usr/share/gnupg/qualified.txt /var/spool/fax/etc" CONFIG_PROTECT_MASK="/etc/ca-certificates.conf /etc/dconf /etc/env.d /etc/env.d/java/ /etc/fonts/fonts.conf /etc/gconf /etc/gentoo-release /etc/revdep-rebuild /etc/sandbox.d /etc/terminfo /etc/texmf/language.dat.d /etc/texmf/language.def.d /etc/texmf/updmap.d /etc/texmf/web2c" CXXFLAGS="-march=core2 -O3 -pipe" DISTDIR="/usr/portage/distfiles" EMERGE_DEFAULT_OPTS="--autounmask=n" FEATURES="assume-digests binpkg-logs distlocks ebuild-locks fixlafiles news parallel-fetch protect-owned sandbox sfperms strict unknown-features-warn unmerge-logs unmerge-orphans userfetch" FFLAGS="" GENTOO_MIRRORS="http://gentoo.mneisen.org/ http://gentoo.mirror.dkm.cz/pub/gentoo/ http://mirrors.linuxant.fr/distfiles.gentoo.org/ http://mirror.netcologne.de/gentoo/" LANG="de_DE.UTF-8" LDFLAGS="-Wl,-O1 -Wl,--as-needed" LINGUAS="de en ar" MAKEOPTS="-j3" PKGDIR="/usr/portage/packages" PORTAGE_CONFIGROOT="/" PORTAGE_RSYNC_OPTS="--recursive --links --safe-links --perms --times --compress --force --whole-file --delete --stats --timeout=180 --exclude=/distfiles --exclude=/local --exclude=/packages" PORTAGE_TMPDIR="/var/tmp" PORTDIR="/usr/portage" PORTDIR_OVERLAY="/var/lib/layman/freeswitch /var/lib/layman/voyageur /var/lib/layman/pro-audio /var/lib/layman/calculate /var/lib/layman/liquidx" SYNC="rsync://rsync.gentoo.org/gentoo-portage" USE="X a52 aac acl acpi alsa amd64 apm avahi berkdb bluetooth branding bzip2 cairo cdda cdr cli colord consolekit cracklib crypt cups custom-cflags cxx dbus disk-partition dri dts dvd dvdr eds emboss encode evo exif fam firefox flac fontconfig fortran gdbm gdu gif gnome gnome-keyring gnome-online-accounts gpm gstreamer gtk gtk3 iconv ipv6 java jpeg kde kerberos lcms ldap libnotify mad mmx mng modules mp3 mp4 mpeg mudflap multilib nautilus ncurses nls nptl nptlonly ogg opengl openmp pam pango pcre pdf png policykit ppds pppd pulseaudio python qt3 qt3support qt4 readline scanner sdl session socialweb spell sqlite sse sse2 ssl startup-notification svg sysfs system-sqlite tcpd tiff truetype udev unicode unlock-notify usb vorbis x264 xcb xinerama xml xorg xulrunner xv xvid zlib" ALSA_CARDS="ali5451 als4000 atiixp atiixp-modem bt87x ca0106 cmipci emu10k1x ens1370 ens1371 es1938 es1968 fm801 hda-intel intel8x0 intel8x0m maestro3 trident usb-audio via82xx via82xx-modem ymfpci" ALSA_PCM_PLUGINS="adpcm alaw asym copy dmix dshare dsnoop empty extplug file hooks iec958 ioplug ladspa lfloat linear meter mmap_emul mulaw multi null plug rate route share shm softvol" APACHE2_MODULES="actions alias auth_basic authn_alias authn_anon authn_dbm authn_default authn_file authz_dbm authz_default authz_groupfile authz_host authz_owner authz_user autoindex cache cgi%* cgid%* dav dav_fs dav_lock deflate dir disk_cache env expires ext_filter file_cache filter headers include info log_config logio mem_cache mime mime_magic negotiation rewrite setenvif speling status unique_id userdir usertrack vhost_alias" CALLIGRA_FEATURES="kexi words flow plan stage tables krita karbon braindump" CAMERAS="ptp2" COLLECTD_PLUGINS="df interface irq load memory rrdtool swap syslog" ELIBC="glibc" GPSD_PROTOCOLS="ashtech aivdm earthmate evermore fv18 garmin garmintxt gpsclock itrax mtk3301 nmea ntrip navcom oceanserver oldstyle oncore rtcm104v2 rtcm104v3 sirf superstar2 timing tsip tripmate tnt ubx" INPUT_DEVICES="evdev synaptics" KERNEL="linux" LCD_DEVICES="bayrad cfontz cfontz633 glk hd44780 lb216 lcdm001 mtxorb ncurses text" LINGUAS="de en ar" PHP_TARGETS="php5-3" RUBY_TARGETS="ruby18" USERLAND="GNU" VIDEO_CARDS="nvidia nv" XTABLES_ADDONS="quota2 psd pknock lscan length2 ipv4options ipset ipp2p iface geoip fuzzy condition tee tarpit sysrq steal rawnat logmark ipmark dhcpmac delude chaos account" Unset: CPPFLAGS, CTARGET, INSTALL_MASK, LC_ALL, PORTAGE_BUNZIP2_COMMAND, PORTAGE_COMPRESS, PORTAGE_COMPRESS_FLAGS, PORTAGE_RSYNC_EXTRA_OPTS ================================================================= Package Settings ================================================================= net-libs/opal-3.6.8-r2 was built with the following: USE="audio celt dtmf fax ffmpeg h224 h281 h323 iax ipv6 java ldap (multilib) plugins sip sipim srtp ssl theora video wav x264 xml -capi -debug -doc -examples -ilbc -ivr -ixj -lid -sbc -static-libs -stats -swig -vpb -vxml -x264-static" CFLAGS="-march=core2 -O3 -pipe -fno-visibility-inlines-hidden -D__STDC_CONSTANT_MACROS" CXXFLAGS="-march=core2 -O3 -pipe -fno-visibility-inlines-hidden -D__STDC_CONSTANT_MACROS" net-libs/ptlib-2.6.7-r1 was built with the following: USE="alsa asn audio dtmf ffmpeg http ipv6 jabber ldap (multilib) sasl sdl soap ssl stun v4l video vxml wav xml xmlrpc -debug -doc (-esd) -examples -ftp -mail -odbc -oss -pch -qos -remote -serial -shmvideo -snmp -socks -static-libs -telnet -tts"

2 1

Event system overloading. Taking a 10 second break
by Stephen Wilde 29 Jan '12

29 Jan '12

When I have many traffic, sometimes it appear in the log: 2012-01-20 17:09:34.028067 [CRIT] switch_event.c:339 Event system overloading. Taking a 10 second break 2012-01-20 17:09:35.036984 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. Any advise on avoid that? The cpu is running at 40%-50% of load. Stephen

8 22

NDLB-connectile-dysfunction does not work
by Anestis Mavro 29 Jan '12

29 Jan '12

Hello, I have tried to add <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> to the user according to the Wiki page about NAT traversal, but it is not working as expected. I see the answer to the registration request going to the correct (Ethernet) IP address, but the port is wrong. It still uses the port from within the SIP packet, instead of the Ethernet port. I have tried also to add at the same time <param name="NDLB-force-rport" value="true"/> without success I am using XML curl for the directory and I confirmed that both settings are in the xml. Is there any other additional setting missing? Thank you

2 1

mod_fifo: agent not ringing
by georg＠riseup.net 28 Jan '12

28 Jan '12

He all, I'm still quite new to FS, so sorry if this is just a stupid question. I'm trying to setup a queue with mod_fifo. Really simple, just with two phones. Idea is that people call a number, hear moh and the two phones are ringing. If picked up, the call is bridged. So I tried this: fifo.con.xml <configuration name="fifo.conf" description="FIFO Configuration"> <fifos> <fifo name="fifo_kunden(a)192.168.1.93" importance="0"> <member timeout="120" simo="1" lag="5">{call_timeout=30,fifo_member_wait=nowait}user/10(a)192.168.1.93</member> <member timeout="120" simo="1" lag="5">{call_timeout=30,fifo_member_wait=nowait}user/15(a)192.168.1.93</member> </fifo> </fifos> </configuration> In my public.xml dialplan I put: <extension name="Local_Extension"> <condition field="destination_number" expression="^(4930XXXXXXXX)$"> <action application="transfer" data="fifo_kunden"/> </condition> </extension> <extension name="fifo_kunden"> <condition field="destination_number" expression="^fifo_kunden$"> <action application="answer"/> <action application="fifo" data="fifo_kunden(a)192.168.1.93 in undef $${base_dir}/sounds/music/8000/ponce-preludio-in-e-major.wav"/> </condition> </extension> A caller now hears the music, but the phones aren't ringing (and there are no error messages about this). I changed the @192.168.1.93 from @$${domain}, cause FS tried to call for example 12@a different ip then my phones are registered. This is, because I've got two ips. One from my provider, receiving and sending calls there, and 192.168.1.93, where my phones are registered. Before I changed the @setting, FS gave me an error that the phone 12@my-provider-ip is absent, which is true. So I thought I just have to change the domain setting. What I'm missing here? Thanks, Georg

1 1

Can FreeSWITCH go into idle mode or something?
by Henrik Aagaard Sørensen 28 Jan '12

28 Jan '12

I have a completely clean installation of FreeSWITCH on a brand new CentOS 6. Everything works as it should. But testing over a longer period, I see some issues when after 2-3 days without any device registrations or contact to the server in any way, the server doesn't accept registrations. anymore I get timeouts on all devices. A wireshark monitor does see the incoming SIP requests. But nothing happens. A quick restart of the server fixes everything. The FS log only contains: 2012-01-26 10:40:32.184037 [WARNING] switch_scheduler.c:114 Task was executed late by 2 seconds 1 heartbeat (core) Can FS somehow enter an idle period, if no activity has been made, or what could cause this sort of problem?

3 3

Line echo cancellation for voice
by georg＠riseup.net 27 Jan '12

27 Jan '12

Hello all, Could someone tell me, how to enable this: http://docs.freeswitch.org/echo_can_page.html ? Thanks, Georg

2 1

sipXecs Conference March 5-6th
by Douglas Hubler 27 Jan '12

27 Jan '12

I wanted to invite everyone interested in learning about the upcoming release of sipXecs v4.6 to come to SIPfoundry's Co-Lab conference on March 5-6th in Fort Collins, CO USA. For those that do not know about sipXecs, it's an open source project that combines SIP based services together with FreeSWITCH, OpenFire and OpenACD into a complete unified communications system. New to the upcoming release 4.6, is sipXecs's tremendous platform capabilities to build and launch your own communication applications. This platform allows you to create services that are secure, redundant, load balancing and easily managed in any cluster. Customize the cluster to run whatever services you want. sipXecs takes a different approach to accomplishing this by letting you chose what technology stack and language to build your apps on. sipXecs just gives you a platform to compile, install and configure your services and integrate them other services down to the finest level of control. Once you're done building your component, you can share it with a greater SIPfoundry community all while maintaining your own source code repository. I've worked on sipXecs since version sipXecs v1.0 and I've never been more excited about a release than sipXecs v4.6. If I've just peeked your interest, please email me at dhubler(a)ezuce.com and I can provide more details. You can also email me on the sipxecs user's mailing list : sipx-users(a)list.sipfoundry.org http://www.sipfoundry.org/sipx-colab Douglas Hubler sipXecs developer

1 0

Re: [Freeswitch-users] Sendmail segfaulting
by Rob Morin 27 Jan '12

27 Jan '12

I saw that, and tried everything I could. Still no luck. My stack limit is 8192, so that shouldn't be the issue. I'm running CentOS x_64. I tried creating a script as the wiki suggests, so that I could make changes without having to restart FS. But that didn't work either (I'm still a noob). My goal was to add a sleep delay, in the case that FS is calling the sendmail command before it's completely written and released the email it's sending. Any idea how to make that work? Thank you, Rob From: Michael Jerris [mailto:mike@jerris.com] Sent: Wednesday, January 25, 2012 1:57 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sendmail segfaulting before second message is sent http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings On Jan 25, 2012, at 1:31 PM, Rob Morin wrote: For some extensions, I've got FreeSWITCH configured to send both the voicemail to an email address and a notification to a cell phone that there is a message. Sendmail is segfaulting (error 6) before the second email message is sent. Sometimes the notification message is going out, sometimes the email message goes. If I am just sending an email, it is *generally* successful. Although sometimes it fails, so I really need to fix this. Does anyone have any suggestions as to how I can troubleshoot this? (Running CentOS 5.7, sendmail 8.13).

3 10

Options ping
by Elliott Vogel 27 Jan '12

27 Jan '12

Hello, how can I make FS send Option pings to users that are registered? I would like to send a ping every 120 seconds because we are having problems with remote users that are behind firewalls using polycom phones, their firewalls are closing ports if there is no activity and calls fail. Also if anyone has better way to keep alive a firewall port I would love to hear it. Elliott

2 4