January 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_fsv and h263
by Greg Millam 30 Jan '12

30 Jan '12

Hi folks - I'm using mod_fsv to record incoming video to an .fsv file, then using ffmpeg to convert it. It works perfectly fine when the video codec is h264 or h263+. But when the video codec is h263, it creates corrupted video. ffmpeg creates a gibberish movie, and reports a large number of errors. (Pasted below). However, when play_fsv is used to play that same .fsv file back via freeswitch to my videophone, the video is clear and fine. The issue appears to be in mod_fsv's recording, fsvdec, or similar. ffmpeg is compiled with fsvdec.c Output when using h264 or h263+ is typical ffmpeg success output, and the generated .mp4 is nice and clear. Warning output from h263 spans a large number of pages, but are variations on: [h263 @ 0x3143200] illegal ac vlc code at 1x1 [h263 @ 0x3143200] Error at MB: 24 [h263 @ 0x3143200] illegal ac vlc code at 14x1 [h263 @ 0x3143200] Error at MB: 37 [h263 @ 0x3143200] concealing 396 DC, 396 AC, 396 MV errors [h263 @ 0x3143200] illegal ac vlc code at 21x15 [h263 @ 0x3143200] Error at MB: 366 [h263 @ 0x3143200] concealing 118 DC, 118 AC, 118 MV errors [h263 @ 0x3143200] illegal dc 0 at 9 16 Last message repeated 1 times ... mod_fsv documentation on the wiki states that mod_fsv is endian dependent, yet all processing is happening on the same machine. (And the fact it works just fine for h264 and h263+ is just baffling). Any thoughts? Thank you! - Greg Millam -- If we do not endeavor to take a step every day, the journey can become so long as to be infinite.

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UPDATE: Donating To Assist Brian West
by Michael Collins 30 Jan '12

30 Jan '12

Good news, We've contacted PayPal and made sure that they are aware of the fact that Brian may be receive a number of donations. Sometimes this can be construed as "suspicious activity" so it's good to be prepared. PayPal has requested that we use payment type "Personal" and that the money is being sent as a "gift." Here's a quick screenshot: Let's all give Brian some PayPal love! Thanks, Michael P.S. - If you have any questions please email me off list.

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sofia require: timer missing from 200 ok
by Chris Martineau 30 Jan '12

30 Jan '12

Hi, There seems to be something broken between git versions (f2cf68b 2011-11-20 18-40-41 -0500) and (58c3c3a 2011-11-22 18-22-57 -0600) in that the first works fine and issues require headers where necessary, but the second does not. They have duplicate configs so can only assume something is broken between the 2 versions. Remember reading something about this but cannot find it again. Is this a known issue? Is it still possible to pull down the older git? Many thanks Chris

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mod_xml_cdr not saving file correctly -- PLease help
by Paul 30 Jan '12

30 Jan '12

I posted recently but my description may have been to complex for others to replicate, so I have boiled the problem down to the simplest of configurations and still need some help Running version is FreeSWITCH Version 1.0.head (git-f25c5aa 2012-01-29 17-37-56 -0600) Calls are answered with a LUA script that has been working well on another server 1) any call coming in on a softphone (with caller id of 1000) that is registered to the server will post to url and save correctly to the log directory 2)Calls with a 10 digit US caller_id coming in on public dialplan to same script receive an error 500 when posting the cdr using mod_xml_cdr and post a garbled file in the errors directory (see below) 3) calls coming in on public dialplan with a UK caller id post to url and save correctly encode is set to true in the mod_xml conf file It appears that there may be 2 issues here 1) mod_xml_cdr (with encode=true) is trying to send file (and save in error log with bad formatting) 2) when it writes the file to the main log directory it is saved correctly I end up with a failed post and 2 cdr's that are saved differently. see examples below THIS IS THE FILE IN THE MAIN LOG DIRECTORY /usr/local/freeswitch/log/xml_cdr/a_f4eb00ab-a364-4c95-a86d-df6eafa3bf98.cdr.xml <?xml version="1.0"?> <cdr> <channel_data> <state>CS_REPORTING</state> <direction>inbound</direction> <state_number>11</state_number> <flags>0=1;1=1;35=1;36=1;38=1;41=1;51=1</flags> <caps>1=1;2=1;3=1;4=1;5=1;6=1</caps> </channel_data> <variables> <direction>inbound</direction> <uuid>f4eb00ab-a364-4c95-a86d-df6eafa3bf98</uuid> AND THIS IS THEY WAY THE FILE IS SAVED IN THE ERROR LOG %3C%3Fxml%20version%3D%221.0%22%3F%3E%0A%3Ccdr%3E%0A%20%20%3Cchannel_data%3E%0A%20%20%20%20%3Cstate%3ECS_REPORTING%3C/state%3E%0A%20%20%20%20%3Cdirection%3Einbound%3C/direction%3E%0A%20%20%20%20%3Cstate_number%3E11%3C/state_number%3E%0A%20%20%20%20%3Cflags%3E0%3D1%3B1%3D1%3B35%3D1%3B36%3D1%3B38%3D1%3B41%3D1%3B51%3D1%3C/flags%3E%0A%20%20%20%20%3Ccaps%3E1%3D1%3B2%3D1%3B3%3D1%3B4%3D1%3B5%3D1%3B6%3D1%3C/caps%3E%0A%20%20%3C/channel_data%3E%0A%20%20%3Cvariables%3E%0A%20%20%20%20%3Cdirection%3Einbound%3C/direction%3E%0A%20%20%20%20%3Cuuid%3Ef4eb00ab-a364-4c95-a86d-df6eafa3bf98%3C/uuid%3E%0A%20%20%20%20%3Csession_id%3E6%3C/session_id%3E%0A%20%20%20%20%3Csip_local_network_addr%3E192.168.100.82%3C/sip_local_network_addr%3E%0A%20%20%20%20%3Csip_network_ip%3E72.249.14.242%3C/sip_network_ip%3E%0A%20%20%20%20%3Csip_network_port%3E5060%3C/sip_network_port%3E%0A%20%20%20%20%3Csip_received_ip%3E72.249.14.242%3C/sip_received_ip%3E%0A%20%20%20%20%3Csip_received_port%3E5060%3C/sip_received_port%3E%0A%20%20%20%20%3Csip_via_protocol%3Eudp%3C/sip_via_protocol%3E%0A%20%20%20%20%3Csip_from_user%3E4088924027%3C/sip_from_user%3E%0A%20%20%20%20%3Csip_from_uri%3E4088924027%254072.249.14.242%3C/sip_from_uri%3E%0A%20%20%20%20%3Csip_from_host%3E72.249.14.242%3C/sip_from_host%3E%0A%20%20%20%20%3Csip_from_user_stripped%3E4088924027%3C/sip_from_user_stripped%3E%0A%20%20%20%20%3Csofia_profile_name%3Eexternal%3C/sofia_profile_name%3E%0A%20%20%20%20%3Csip_req_user%3E8132007296%3C/sip_req_user%3E%0A%20%20%20%20%3Csip_req_port%3E5080%3C/sip_req_port%3E%0A%20%20%20%20%3Csip_req_uri%3E8132007296%2540192.168.100.82%253A5080%3C/sip_req_uri%3E%0A%20%20%20%20%3Csip_req_host%3E192.168.100.82%3C/sip_req_host%3E%0A%20%20%20%20%3Csip_to_user%3E8132007296%3C/sip_to_user%3E%0A%20%20%20%20%3Csip_to_port%3E5080%3C/sip_to_port%3E%0A%20%20%20%20%3Csip_to_uri%3E8132007296%2540192.168.100.82%253A5080%3C/sip_to_uri%3E%0A%20%20%20%20%3Csip_to_host%3E192.168.100.82%3C/sip_to_host%3E%0A%20%20%20%20%3Csip_contact_user%3E4088924027%3C/sip_contact_user%3E%0A%20%20%20%20%3Csip_contact_uri%3E4088924027%254072.249.14.242%3C/sip_contact_uri%3E%0A%20%20%20%20%3Csip_contact_host%3E72.249.14.242%3C/sip_contact_host%3E%0A%20%20%20%20% -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-xml-cdr-not-saving-file-c… Sent from the freeswitch-users mailing list archive at Nabble.com.

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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

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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"

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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

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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

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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

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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?

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