October 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Minimal configuration for FreeSWITCH 1.6 (audio only, works out of the box)
by Igor Bannicov 27 Oct '15

27 Oct '15

Hi all, Lost some time to create for my projects a working minimal configuration of FreeSWITCH 1.6 (no video, no conferences, no IVRs, no logs). May be used as-is or as a base for more sophisticated configuration. Details in repo's README Would be glad, if it will help anybody. https://github.com/alphazo-id/fs1.6-min-audio Best regards! Igor Bannicov.

3 2

mod_cdr_pg_csv not compatible with freeswitch-all?
by Michael Nielsen 27 Oct '15

27 Oct '15

Hello everyone. I’m running FreeSWITCH 1.6.2 on Debian 8 - from the Debian package installation freeswitch-all. I would like to use mod_cdr_pg_csv for saving CDR in PostgreSQL. However, the module mod_cdr_pg_csv is not coming installed with freeswitch-all. When trying to run apt-get install freeswitch-mod-cdr-pg-csv I cannot without removing freeswitch-all and install a bunch of other packages… Isn’t it possible to use the module with freeswitch-all?

4 8

Eavesdrop on early media ?
by Peter Baines 27 Oct '15

27 Oct '15

Hi, Is it possible to get early media on a eavesdrop call ? Regards, Peter

1 0

LUA script not found even though script directory is correct
by Michael Nielsen 27 Oct '15

27 Oct '15

I get the error: cannot open /usr/share/freeswitch/scripts/users.lua My script is located at /etc/freeswitch/scripts/ and I've added the following to my lua.conf.xml <param name="script-directory" value="/etc/freeswitch/scripts/?.lua"/> <param name="xml-handler-script" value="users.lua"/> <param name="xml-handler-bindings" value="directory"/> Why is it still looking in /usr/share/freeswitch...?

6 13

Re: [Freeswitch-users] The FreeSWITCH 1.6.0 release is here!
by Luke Milbourne 27 Oct '15

27 Oct '15

Hi all, Please can someone provide more information on FS-7760 and how the new parameter is used? Happy to fully document once I've got a better understanding of it. Thanks, Luke On 8 September 2015 at 15:56, <freeswitch-users-request(a)lists.freeswitch.org > wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Compiling under SmartOS (Support) > 2. Re: Compiling under SmartOS (Stanislav Sinyagin) > 3. The FreeSWITCH 1.6.0 release is here! (krice(a)freeswitch.org) > > > ---------- Forwarded message ---------- > From: Support <support(a)directvoip.co.uk> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 08 Sep 2015 14:42:42 +0100 > Subject: Re: [Freeswitch-users] Compiling under SmartOS > Stanislav, > > I cannot help with this, my skills don't reach that far but I can point > you in the right direction. > > I had the same problem and it seems to be related to something called > timerfd. Using the ppriv command you can see freeswitch is wanting access > to the high res clock, something not available to smartos zones with the > default privileges. > > I did manage to mess with the smartos privileges and get it to run at one > time but it was only when running as root. > > If you actually used an older version of smartos, I think the one I used > is dated around january this year, then it will compile fine and use some > sort of other timing method. > > This timerfd thing, thing relates to freeswitch finding a file called > timerfd.h that didn't appear in smartos zones until about march this year, > something to do with lx brand I think. > > I have just looked and the smartos version that will compile > is 20150108T111855Z, obviously it would be better on the newer. > > Compiling on the above smartos version and then running it on newer is no > problem which is what I ended up doing. > > Hope this helps > > Regards > Darren > > ------------------------------ > *From:* Stanislav Sinyagin [mailto:ssinyagin@gmail.com] > *To:* FreeSWITCH Users Help [mailto:freeswitch-users@lists.freeswitch.org] > *Sent:* Tue, 08 Sep 2015 07:55:37 +0000 > *Subject:* Re: [Freeswitch-users] Compiling under SmartOS > > See the update at https://freeswitch.org/jira/browse/FS-7967 > > I fixed the compilation problems, and now there's a runtime issue. > > On Mon, Aug 17, 2015 at 10:40 AM, Stanislav Sinyagin > <ssinyagin(a)gmail.com> wrote: > > I see there are some people on the list, working with SmartOS. > > > > The current master fails to compile: > > https://freeswitch.org/jira/browse/FS-7967 > > > > Your input will be appreciated. > > > > I just started looking around and getting the feeling what SmartOS is. > > I worked with Solaris quite a lot, but that was almost 10 years ago. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > 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 > > > > ---------- Forwarded message ---------- > From: Stanislav Sinyagin <ssinyagin(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 8 Sep 2015 16:20:28 +0200 > Subject: Re: [Freeswitch-users] Compiling under SmartOS > Darren, thanks a lot, this will save me a few hours. > > I think I'm able to manage it to use the right timer, it's only a > matter of time. At the moment it's a non-profit project for me, so I > can only dedicate an hour a day or so. I'm also open for commercial > requests :-)) > > > > > > > > On Tue, Sep 8, 2015 at 3:42 PM, Support <support(a)directvoip.co.uk> wrote: > > Stanislav, > > > > I cannot help with this, my skills don't reach that far but I can point > you > > in the right direction. > > > > I had the same problem and it seems to be related to something called > > timerfd. Using the ppriv command you can see freeswitch is wanting > access to > > the high res clock, something not available to smartos zones with the > > default privileges. > > > > I did manage to mess with the smartos privileges and get it to run at one > > time but it was only when running as root. > > > > If you actually used an older version of smartos, I think the one I used > is > > dated around january this year, then it will compile fine and use some > sort > > of other timing method. > > > > This timerfd thing, thing relates to freeswitch finding a file called > > timerfd.h that didn't appear in smartos zones until about march this > year, > > something to do with lx brand I think. > > > > I have just looked and the smartos version that will compile is > > 20150108T111855Z, obviously it would be better on the newer. > > > > Compiling on the above smartos version and then running it on newer is no > > problem which is what I ended up doing. > > > > Hope this helps > > > > Regards > > Darren > > > > ________________________________ > > From: Stanislav Sinyagin [mailto:ssinyagin@gmail.com] > > To: FreeSWITCH Users Help [mailto:freeswitch-users@lists.freeswitch.org] > > Sent: Tue, 08 Sep 2015 07:55:37 +0000 > > Subject: Re: [Freeswitch-users] Compiling under SmartOS > > > > > > See the update at https://freeswitch.org/jira/browse/FS-7967 > > > > I fixed the compilation problems, and now there's a runtime issue. > > > > On Mon, Aug 17, 2015 at 10:40 AM, Stanislav Sinyagin > > <ssinyagin(a)gmail.com> wrote: > >> I see there are some people on the list, working with SmartOS. > >> > >> The current master fails to compile: > >> https://freeswitch.org/jira/browse/FS-7967 > >> > >> Your input will be appreciated. > >> > >> I just started looking around and getting the feeling what SmartOS is. > >> I worked with Solaris quite a lot, but that was almost 10 years ago. > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > 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 > > > > > ---------- Forwarded message ---------- > From: krice(a)freeswitch.org > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Tue, 08 Sep 2015 14:56:10 +0000 > Subject: [Freeswitch-users] The FreeSWITCH 1.6.0 release is here! > New Post on freeswitch.org from Kathleen King > check it out at http://ift.tt/1NfRNk0 > The FreeSWITCH 1.6.0 release is here! > > The FreeSWITCH 1.6.0 release is here! This is a routine maintenance > release and the resources are located here: > > - Tarball: http://ift.tt/1NfRO7L > - Packaging: http://ift.tt/1U8FMQN (Please Note the updated location > for 1.6 packages) > > And we’re dropping support in packaging for anything older than Debian 8.0 > and anything older than Centos 7.0 due to a number of dependency issues on > older platforms. > > New features that were added: > > - FS-7337 <http://ift.tt/1B2enJO> [mod_sofia] Add support for > Remote-Party-ID header in UPDATE request. > - FS-7561 <http://ift.tt/1MDcYLa> [mod_sofia] Add Perfect Forward > Secrecy (DHE PFS) > - FS-7560 <http://ift.tt/1B2ell8> [mod_nibblebill] Added new options > to nibble bill for minimum charges and rounding > - FS-7587 FS-7602 FS-7499 [mod_verto] Add ipv6 support to Verto / > Websockets and additional support ice/dtls ipv6 functionality > - FS-6801 [mod_sofia] Add sip_watched_headers variable to launch > events when a SIP message contains a given SIP header > - FS-7564 [mod_rayo] Added new algorithms for offering calls to clients > - FS-7436 FS-7601 [mod_opus] Added FEC support > - FS-7603 [mod_event_socket] Failover for socket application in > dialplan > - FS-7585 [mod_rtmp] Increased AMF buffer for larger video and add > bandwidth settings to flash video > - FS-7311 [mod_sofia] Updating display name is disabled when caller_id > equal “_undef_” > - FS-7513 [mod_conference] Add video-auto-floor-msec param to control > how long a member must have the audio floor before also taking the video > floor and make sure user does not have auto avatar when not visible > - FS-7620 <http://ift.tt/1BiZfrx> [ftmod_libpri] Correctly set calling > number presentation and screening fields > - FS-7138 <http://ift.tt/1GoRakl> [mod_callcenter] Added a new > reserve-agents param > - FS-7436 <http://ift.tt/1B2elle> FS-7601 <http://ift.tt/1GoRcIR> > [mod_opus] FEC support > - FS-7623 <http://ift.tt/1BiZeE6> [mod_amqp] Allow for custom exchange > name and type for producers and fixed param name ordering bug caused by > exposing these params > - FS-7638 <http://ift.tt/1GoRcIV> Allow ipv4 mapped ipv6 address to > pass ipv4 ACLs properly > - FS-7643 <http://ift.tt/1BiZfrz> [mod_opus] Added interpretation of > maxplaybackrate and sprop-maxcapturerate > - FS-7641 <http://ift.tt/1GoRcIZ> Added video support to eavesdrop > - FS-7656 <http://ift.tt/1ByUASy> [mod_localstream] Added > mod_local_stream video support, and make mod_conference move the video in > and out of a layer when the stream has video or not, scan for relative file > in art/eg.wav.png and display it as video when playing audio files, put > video banner up if artist or title is set, and fixed a/v sync on first > connection > - FS-7629 <http://ift.tt/1ByUBWC> [mod_conference] Added member status > in json format to the conference live array, add livearray-json-status to > conference-flags to enable > - FS-7517 <http://ift.tt/1d2d2Ye> FS-7519 <http://ift.tt/1d2d4zl> > [mod_av] [mod_openh264] Added H264 STAP-A packeting support so it would > work with FireFox > - FS-7664 <http://ift.tt/1RtuVwt> [mod_verto] Set ICE candidate > timeout to wait for only 1 second to fix media delays > - FS-7660 <http://ift.tt/1RtuUJ2> [mod_opus] Enabled with new API > command “opus_debug” to show information about Opus payload for debugging. > - FS-7519 <http://ift.tt/1d2d4zl> [mod_av] Fixed bitrate and added > some presets > - FS-7693 <http://ift.tt/1RtuVwv> [mod_conference] Lower the default > energy level in sample configs to improve voice quality > - FS-7720 <http://ift.tt/1LEZrFh> Improve play_and_detect_speech to > set current_application_response channel variable as follows: “USAGE > ERROR”: bad application arguments’, “GRAMMAR ERROR”: speech recognizer > failed to load grammar, “ASR INIT ERROR”: speech recognizer failed to > allocate a session, and “ERROR”: any other errors > - FS-7732 <http://ift.tt/1S4zFZP> Continue recording with uuid_transfer > - FS-7752 <http://ift.tt/1KIMKck> [mod_rayo] Increase maximum number > of elements from 30 to 1024 to allow adhearsion to create large grammars to > navigate IVR menus. > - FS-7750 <http://ift.tt/1S4zFZT> [mod_commands] Allow for uuid_setvar > to handle arrays > - FS-7758 <http://ift.tt/1KIMKco> [mod_loopback] Emit an event if a > loopback bowout occurs > - FS-7759 <http://ift.tt/1S4zFZW> [mod_sofia] Added the channel > variable ignore_completed_elsewhere to suppress setting the completed > elsewhere cause > - FS-7771 <http://ift.tt/1KIMLwJ> Set a channel variable if the > recording is terminated due to silence hits > - FS-7760 <http://ift.tt/1S4zIoz> Added xml fetch for channels to > externally support nightmare transfer depends on > channel-xml-fetch-on-nightmare-transfer profile param (default is disabled) > - FS-7730 <http://ift.tt/1KIMKsN> [mod_smpp] Added mod_smpp as an > event handler module > and fixed the default configs to provided sample load option for > mod_sms and mod_smpp > - FS-7774 <http://ift.tt/1S4zIoB> Add mod_kazoo > - FS-7780 <http://ift.tt/1O2b0nH> Add new channel variable > max_session_transfers. If set, this variable is used to count the number of > session transfers allowed instead of the max_forwards variable. If not set, > the existing behavior is preserved. > - FS-7783 <http://ift.tt/1K2VF8i> Add channel variable for capturing > DTMF input when using play_and_get_digits when the response does not match > - FS-7772 <http://ift.tt/1K2VBWk> [mod_opus] Add functionality to keep > FEC enabled on the encoder by modifying the bitrate if packet loss changes > (Opus codec specific behaviour). > - FS-7799 <http://ift.tt/1O2b0nN> [mod_png] Add API command > uuid_write_png > - FS-7801 <http://ift.tt/1K2VF8m> [mod_opus] Added support to set CBR > mode > - FS-7685 <http://ift.tt/1O2b0E1> [mod_say_nl] Fix Dutch numbers > pronunciation > - FS-7198 <http://ift.tt/1K2VF8o> Add coma separated values and > reverse ranges for time-of-day and day-of-week matches > - FS-7809 <http://ift.tt/1O2aZ34> [mod_opus] Added 60 ms ptime for > Opus at 8 khz ( opus@8000h@60i ) > - FS-7405 <http://ift.tt/1O2b0E7> [mod_dialplan_xml] Fix condition > regex=”all” to work with time conditions > - FS-7819 <http://ift.tt/1K2VF8s> [mod_opus] Restore bitrate (if > there’s no more packet loss) and added step for 60 ms > - FS-7773 <http://ift.tt/1K2VF8u> [mod_sofia] Adding additional > transfer events when the fire-transfer-events=true profile parameter > is set > - FS-7820 <http://ift.tt/1O2aZ3a> FreeSWITCH automated unit test and > micro benchmark framework > - FS-7769 <http://ift.tt/1TMWpQR> [mod_conference] Add new > multi-canvas and telepresence features > - FS-7847 <http://ift.tt/1JcZmHb> [mod_conference] Add layers that do > not match the aspect ration of conference by using the new hscale layer > param for horizontal scale, and add zoom=true param to crop layer instead > of letterbox, add grid-zoom layout group that demonstrates these layouts, > and fix logo ratios and add borders too. > - FS-7813 <http://ift.tt/1TMWsfi> [mod_conference] Add vmute member > flag. > - FS-7846 <http://ift.tt/1JcZnLc> [mod_dptools] Add > eavesdrop_whisper_aleg=true and eavesdrop_whisper_bleg=true channel > variables to allow you to start eavesdrop in whisper mode of specific call > leg > - FS-7760 <http://ift.tt/1S4zIoz> [mod_sofia] Revise channel fetch on > nightmare transfer and add dial-prefix and absolute-dial-string to the > nightmare xml > - FS-7829 <http://ift.tt/1JcZmHe> [mod_opus] Add sprop-stereo fmtp > param to specify if a sender is likely to send stereo or not so the > receiver can safely downmix to mono to avoid wasting receiver resources > - FS-7830 <http://ift.tt/1TMWpQW> [mod_opus] Added use-dtx param in > config file (enables DTX on the encoder, announces in fmtp) > - FS-7824 <http://ift.tt/1JcZnLe> [mod_png] Add functionality for > capturing screenshots from both legs to uuid_write_png > - FS-7549 <http://ift.tt/1TMWq7a> [mod_ladspa] Added an API for > removing an active ladspa effect on a channel. For conformance reasons, the > uuid_ladspa command now accepts ‘stop’ and ‘start’, while the previous > functionality (without any verb) which will simply add ladspa remains > intact. > - FS-7848 <http://ift.tt/1fDUoYd> [mod_opus] Add support for 80 ms, > 100 ms, 120 ms packetization > - FS-7519 <http://ift.tt/1d2d4zl> FS-7677 <http://ift.tt/1fDUoYf> > [mod_av] Add H.263 codec support > - FS-7885 <http://ift.tt/1Jstpe1> Add getcputime to retrieve > FreeSWITCH process CPU usage > - FS-7889 <http://ift.tt/1fDUpet> [mod_conference] Move conference > chat to use an event channel so messages only go to the right ‘room’ for > the conference and move conference chat functionality to use event_channel. > - FS-7900 <http://ift.tt/1hIBbFY> [mod_png] Allow snapshot of single > legged calls > - FS-7912 <http://ift.tt/1Pi1Hkb> [mod_lua] Added session UUID to lua > error logs, if known and added session UUID to embedded language (lua, > javascript, etc) logs when session sanity check fails > - FS-7760 <http://ift.tt/1S4zIoz> [mod_sofia] Improved the xml fetch > lookup for channels on nightmare transfer > - FS-7922 <http://ift.tt/1Pi1JbL> [mod_commands] Added uuid_redirect > API command. This provides the equivalent functionality of the dptools > “redirect” application as an api command. > - FS-7806 <http://ift.tt/1WzTqxA> FS-7803 <http://ift.tt/1WzTqxA> > [mod_amqp] Added new properties to amqp configuration, fixed the usage for > enable_fallback_format_fields, and added amqp_util_encode to fix a routing > key issue > - FS-7972 <http://ift.tt/1IZhuk5> [verto communicator] Creating Verto > Communicator > - FS-7988 <http://ift.tt/1hDxBgy> Add a perl script to help file bugs > from the command line and add fixbug.pl to tree > - FS-8009 <http://ift.tt/1PvMyf9> [verto communicator] Create a grunt > project with livereload support. Documentation can be found here > <http://ift.tt/1hDxBgA>. > - FS-8010 <http://ift.tt/1PvMxb1> [verto communicator] Add options for > googAutoGainControl, googNoiseSuppression, and googHighpassFilter > - FS-7855 <http://ift.tt/1hDxBgC> [verto communicator] Pass > userVariables back to the live array to allow for displaying the Gravatar > associated with a member’s email address > - FS-8075 <http://ift.tt/1EArcgC> [mod_hiredis] Add mod_hiredis > including support for redis limits and added support for raw redis > commands. Added deprecation notices to mod_redis > - FS-8049 <http://ift.tt/1hw82xq> [mod_commands] Add getenv FSAPI > - FS-8036 <http://ift.tt/1EArcwS> [verto.js] Add chatCallback to > $.verto.conf > > Improvements in build system, cross platform support, and packaging: > > - FS-7610 Fixed a gcc5 compilation issue > - FS-7499 Fixed a build error on 32bit platforms > - FS-7570 Fixed a compilation issue w/ zrtp enabled > - FS-7426 Only disable mod_amqp on Debian Squeeze and Wheezy > - FS-7635 <http://ift.tt/1BiZfrB> Removed msvc 2005, 2008, and 2010 > non working build systems > - FS-7373 <http://ift.tt/1N6YVMs> Expose the custom repo and key path > to the build-all command too > - FS-7648 <http://ift.tt/1BiZeEa> Foundation for QA testing config , > adding leave/check videomail test cases, adding videomail voicemail > profile, adding video record/ playback test cases, adding set video on > hold, force pre-answer prefix, and adding an eavesdrop test case. > - FS-7338 <http://ift.tt/1PQJI97> Removed mod_shout dep libs to system > libs to continue cleaning up the libs for the 1.6 build process and added > Debian packaging for several new modules, as well as handle system lib > change for a handful of modules > - FS-7653 <http://ift.tt/1BiZfrF> Sample build system for a stand > alone(out of tree) FreeSWITCH module > - FS-7601 <http://ift.tt/1GoRcIR> [mod_opus] [mod_silk] Removed a > bounds check that can never be true in opus fec code and modify > jitterbuffer usage to match the api change > - FS-7648 <http://ift.tt/1BiZeEa> More work toward setting up a QA > testing configuration, add condition testing for regex all and xor cases, > adding profile-variable for testing cases , add lipsync tests for playback > and local stream, add stereo, and configuration for mcu test > - FS-7338 <http://ift.tt/1PQJI97> Fixed bug in Debian packaging when > trying to build against custom repo > - FS-7609 <http://ift.tt/1BiZhzA> Enable building of mod_sangoma_codec > for Debian Wheezy/Jessie > - FS-7667 <http://ift.tt/1ByUCcW> [mod_java] Fixed include directory > detection when using Debian java packages and use detected directory > - FS-7655 <http://ift.tt/1RtuUJ7> Make libvpx and libyuv optional > (none of the video features will work without them) The following modules > require these libraries to be installed still: mod_av mod_cv mod_fsv > mod_mp4v2 mod_openh264 mod_vpx mod_imagick mod_vpx mod_yuv mod_png mod_vlc, > fix build issue w/ strict prototypes, and fix a few functions that need to > be disabled without YUV > - FS-7605 <http://ift.tt/1ByUB8U> Fixed default configuration > directory in Debian packages and fixed Debian packaging dependencies on > libyuv and libvpx > - FS-7669 <http://ift.tt/1ByUCd0> When installing from Debian > packaging if you don’t have the /etc/freeswitch directory, we will install > the default packages for you. If you already have this directory, we’ll let > you deal with your own configs. > - FS-7297 <http://ift.tt/1DLZjA7> [mod_com_g729] Updated the make > target installer > - FS-7644 <http://ift.tt/1ByUCd2> Added a working windows build > without video support for msvc 2013 > - FS-7666 <http://ift.tt/1RtuVwx> [mod_managed] Fixed error building > mod_managed on non windows platforms > - FS-7707 <http://ift.tt/1LEZtgc> Fix build error on CentOS7 > - FS-7655 <http://ift.tt/1RtuUJ7> Fixed a build error when we have PNG > but not YUV > - FS-7723 <http://ift.tt/1JmfR1S> Change RPMs to use -ncwait instead > of -nc. This will cause the initscript to pause and wait for FS to be ready > before continuing. > - FS-7648 <http://ift.tt/1BiZeEa> Added a test cases for FS-7724 and > FS-7687 > - FS-7726 <http://ift.tt/1JmfR1W> Additional configurations for a QA > test case > - FS-7715 <http://ift.tt/1LEZtge> Updates to configure and spec files > for next development branch and added images to spec file and fixed > build/freeswitch.init.redhat since redhat likes to override settings in the > script with TAGs in comments > - OPENZAP-238 <http://ift.tt/1KIMLwR> [freetdm] Fix some GSM > compilation errors and do a bit of code cleanup > - OPENZAP-237 <http://ift.tt/1S4zIoD> [freetdm] Use __func__ instead > of __FUNCTION__ to comply with c99 in gcc 5.1 > - FS-7628 <http://ift.tt/1K2VHwV> [mod_erlang_event] Removed unused > variables causing a compilation error > - FS-7776 <http://ift.tt/1O2b0Ec> Add mod_kazoo to packaging > - FS-7845 <http://ift.tt/1JcZnLg> [mod_conference] Break up > mod_conference into multiple source files to improve build performance > - FS-7769 <http://ift.tt/1TMWpQR> [mod_conference] Fixed a build issue > - FS-7820 <http://ift.tt/1O2aZ3a> Fix build system typo. Don’t assign > the same variable twice. > - FS-7043 <http://ift.tt/1JcZnLi> Fixed apr1 unresolved symbols in > libfreeswitch.so.1.0.0 > - FS-7130 <http://ift.tt/1TMWq7c> Make /run/freeswitch persistent in > the Debian packages, so it will start under systemd > - FS-7860 <http://ift.tt/1Jstqik> Prevent a switch_rtp header conflict > - FS-7130 <http://ift.tt/1TMWq7c> Make /run/freeswitch persistent, so > it will start under systemd > - FS-7728 <http://ift.tt/1WzTsFC> Fixed Windows build issues minus > video features > - FS-7965 <http://ift.tt/1PvMxb5> [mod_conference] Fixed an error > thrown when compiling with GCC > - FS-7985 <http://ift.tt/1hDxyS9> [mod_voicemail] Fixed a compilation > error on 32-bit PCC platform > - FS-8015 <http://ift.tt/1PvMyfb> [mod_conference] Add project dir to > include for mod_conference so it picks up mod_conference.h for Windows > - FS-8061 <http://ift.tt/1hw82NY> [verto_communicator] Adding license > to package.json > - FS-8047 <http://ift.tt/1WX6BsB> [build] Fixed build error in > mod_basic, mod_rtmp, mod_oreka, and mod_sangoma_codec due to using > __FUNCTION__ on newer compilers > - FS-8054 <http://ift.tt/1hw80pm> [mod_rayo] Fixed a warning when > building on Debian > - FS-8055 <http://ift.tt/1EArfc0> [build] Add confdir variable to > freeswitch.pc > - FS-7966 <http://ift.tt/1EArfc2> [windows] Working msvc 2015 build > support. Does not yet include video features. > - FS-8019 <http://ift.tt/1EArcwU> [debian] Excluded few modules that > fail to compile from debian/bootstrap.sh, fixed the handling of -T and -t > and added debian/apt_sources.list with necessary dependencies to build > master, and updated debian/README.source > - FS-8058 <http://ift.tt/1hw80FI> [mod_vpx] Build correctly against > libvpx that is not installed into default system library locations > > The following bugs were squashed: > > - FS-7579 [mod_conference] Fixed a bug not allowing suppression of > play-file-done > - FS-7462 [mod_opus] Fix FMTP in the INVITE to use values from > opus.conf.xml > - FS-7593 [mod_skinny] Fixed a bug where skinny phones would stomp on > each other in database when thundering herd occurs > - FS-7597 [mod_codec2] Fixed encoded_data_len for MODE 2400, it should > be 6 bytes. Also replaced 2550 bps bitrate (obsoleted operation mode) by > 2400 > - FS-7604 [fs_cli] Fixed fs_cli tab completion concurrency issues on > newer libedit > - FS-7258 FS-7571 [mod_xml_cdr] Properly encode xml cdr for post to > web server > - FS-7584 More work on rtcp-mux interop issue with chrome canary > causing video transport failure > - FS-7586 [mod_conference] Change the default > min-required-recording-participants option for mod_conference from 2 to 1 > and silence the warning when the value is set to 1 in the configs > - FS-7607 Update URLs to reflect https protocol on freeswitch.org > websites and update additional URLs to avoid 301 redirects. > - FS-7479 Fixed a crash caused by large RTP/PCMA packets and resampling > - FS-7524 [mod_callcenter] Fixing tiers, level and position should > default to 1 instead of 0 > - FS-7613 Fixed a crash in core text rendering > - FS-7612 <http://ift.tt/1GoRcJ1> Fixed invalid json format for > callflow key > - FS-7609 <http://ift.tt/1BiZhzA> [mod_sangoma_codec] Now that > libsngtc-dev and libsngtc are in the FS debian repo, enable > mod_sangoma_codec > - FS-7621 <http://ift.tt/1GoRaAB> [mod_shout] Fixed a slow interrupt > - FS-7432 <http://ift.tt/1BiZfrH> Fixed missing a=setup parameter from > answering SDP > - FS-7622 <http://ift.tt/1GoRaAD> [mod_amqp] Make sure to close the > connections on destroy. Currently the connection is malloc’d from the > module pool, so there is nothing to destroy. > - FS-7586 <http://ift.tt/1B2eo0c> [mod_vlc] A fix for failing to > encode audio during the recording of video calls > - FS-7573 <http://ift.tt/1GoRcZh> Fixed 80bit tag support for zrtp > - FS-7636 <http://ift.tt/1BiZhzE> Fixed an issue with > transfer_after_bridge and park_after_bridge pre-empting transfers > - FS-7654 <http://ift.tt/1GoRaAF> Fixed an issue with eavesdrop audio > not working correctly with a mixture of mono and stereo > - FS-7641 <http://ift.tt/1GoRcIZ> Fixed a segfault in eavesdrop video > support > - FS-7649 <http://ift.tt/1RtuVwz> [mod_verto] Fixed issue with h264 > codec not being configured in verto.conf.xml > - FS-7657 <http://ift.tt/1ByUB8W> [mod_verto] Fixed a bug with TURN > not being used. Note, you can pass an array of stun servers, including > TURN, to the verto when you start it up. (see verto.js where iceServers is > passed) > - FS-7665 <http://ift.tt/1RtuXEx> [mod_conference] Fixed a bug with > the video floor settings not giving the video floor to the speaker > - FS-7650 <http://ift.tt/1ByUCd4> [mod_verto] Fixed crash when making > a call from a verto user with profile-variables in their user profile > - FS-7710 <http://ift.tt/1RtuXEz> [mod_conference] Added the ability > to set bandwidth to “auto” for conference config > - FS-7432 <http://ift.tt/1BiZfrH> Fixed dtls/srtp, use correct a=setup > parameter on recovering channels > - FS-7678 <http://ift.tt/1RtuVwD> Fixed for fail_on_single_reject not > working with | bridge > - FS-7709 <http://ift.tt/1ByUB90> [mod_verto] Verto compatibility > fixes for Firefox > - FS-7689 <http://ift.tt/1RtuXEB> [mod_lua] Fixed a bug with lua not > loading directory configurations > - FS-7694 <http://ift.tt/1ByUCd8> [mod_av] Fixed for leaking file > handles when the file is closed. > - FS-7467 <http://ift.tt/1JmfO69> [mod_callcenter] Fixing stuck > channels using uuid-standby agents > - FS-7699 <http://ift.tt/1LEZrFj> [mod_verto] Fixed for browser > compatibility > - FS-7722 <http://ift.tt/1LEZrFl> Fixed an issue with record_session > including params when creating path > - FS-7489 <http://ift.tt/1JmfRig> [mod_unimrcp] Fixed a TTS Audio > Queue Overflow > - FS-7724 <http://ift.tt/1LEZrVz> [mod_conference] Fixed a segfault > when missing fonts when trying to render banner > - FS-7519 <http://ift.tt/1d2d4zl> [mod_av] Fixed a regression in the > visual appearance of decode app output > - FS-7703 <http://ift.tt/1LEZtgh> Fixed a bug caused by answer_delay > being set in the default configurations > - FS-7679 <http://ift.tt/1LEZtgj> [mod_verto] Fixed a bug causing one > way audio on Chrome when video is enabled and when using a sip without video > - FS-7729 <http://ift.tt/1JmfRio> [mod_verto] Fixed the formatting for > IPv6 addresses > - FS-7734 <http://ift.tt/1KIMLwT> [mod_nibblebill] Fixed a deadlock > - FS-7726 <http://ift.tt/1JmfR1W> Fixed a bug with recording a video > session on DTMF command > - FS-7721 <http://ift.tt/1KIMKsT> Fixed a segfault caused when using > session:recordFile() and session:unsetInputCallback in a lua script > - FS-7429 <http://ift.tt/1U8FNE8> [mod_curl] Fixed to output valid json > - FS-7746 <http://ift.tt/1KIMLwZ> [mod_verto] Fixed a device > permission error in verto client > - FS-7753 <http://ift.tt/1S4zFZZ> [mod_local_stream] Fixed some > glitching and freezing video when using hold/unhold > - FS-7761 <http://ift.tt/1KIMKJb> [core] Fix shutdown races running > api commands during shutdown > - FS-7767 <http://ift.tt/1KIMKJd> [mod_sofia] Fixed a segfault caused > by invalid arguments to sip_dig > - FS-7744 <http://ift.tt/1S4zIEX> [mod_conference] Fixed a bug causing > the first user’s video stream to stop when another verto user calls the > conference > - FS-7486 <http://ift.tt/1KIMLNf> [mod_sofia] Fixed the handling of > queued requests > - FS-7775 <http://ift.tt/1S4zIEZ> [mod_conference] Fix threading issue > causing stuck worker threads > - FS-7777 <http://ift.tt/1KIMLNh> [mod_imagick] Fixed a regression > causing a segfault when playing png & pdf in conference > - FS-7778 <http://ift.tt/1O2aZ3c> [mod_sofia] Fixed a bug causing a > SQL statement to fail because of a double quote instead of a single quote > - FS-7754 <http://ift.tt/1K2VHwX> [freetdm] Fixed a bug relating to > single digit dial-regex with analog devices > - FS-7785 <http://ift.tt/1K2VHwZ> [mod_opus] Fix for invalid ptime 30 > ms for opus@8000h . Replaced 30 ms with 40 ms. > - FS-7762 <http://ift.tt/1O2b1YA> [mod_av] Handle buffer allocation > failures of large buffers > - FS-7849 <http://ift.tt/1JcZmHh> [verto] Remove extra div breaking > full screen in html > - FS-7832 <http://ift.tt/1TMWsfk> [mod_opus] Fixes when comparing > local and remote fmtp params > - FS-7731 <http://ift.tt/1JcZmHj> [mod_xml_cdr] Fixed a curl default > connection timeout > - FS-7844 <http://ift.tt/1TMWq7g> Fix packet loss fraction when > calculating loss average > - FS-7789 <http://ift.tt/1Jstpe3> [mod_av] Fixed issue with audio > dropping out partway through recordings > - FS-7854 <http://ift.tt/1fDUqPH> Add task_runtime to tasks table in > core database > - FS-7856 <http://ift.tt/1Jstpe9> [mod_av] Fix some segfaults and > leaks. > - FS-7866 <http://ift.tt/1fDUqPN> Fixed a crash when running incorrect > var api expansion syntax “eval ${${external_sip_ip}:4}” > - FS-7861 <http://ift.tt/1Jstpeb> FS-7862 [mod_conference] Fixed a > crash and other issues caused by multi canvas feature > - FS-7681 <http://ift.tt/1fDUpey> [mod_conference] Factor out > conference->canvas and allow per canvas record and play > - FS-7869 <http://ift.tt/1JstqyA> [mod_conference] Fixed a deadlock on > shutdown after playing video file that will not display video > - FS-7654 <http://ift.tt/1GoRaAF> Fixed regressions on eavesdropping > on channels playing a file and on channels with unlike rates > - FS-7872 <http://ift.tt/1JstqyC> [mod_verto] Gracefully fail > attempting to transfer 1 legged call > - FS-7874 <http://ift.tt/1fDUqPR> [mod_conference] Fixed incorrect > layout group count > - FS-7870 <http://ift.tt/1Jstped> [mod_conference] Allow jsonapi > commands to pass the string id field to pass special ID’s like “last” > - FS-7882 <http://ift.tt/1JstqyE> [mod_conference] Allow JSON API > commands to send third arg for muting > - FS-7888 <http://ift.tt/1fDUr65> [mod_verto] Fixed namespacing > problems in javascript library masked by global verto object > - FS-7811 <http://ift.tt/1JstqyG> Use more common format CIF for blank > image > - FS-7902 <http://ift.tt/1fDUpeK> [mod_local_stream] Fix for queue > filling up when you have a mix of video and non video files > - FS-7891 <http://ift.tt/1JstqyI> [mod_spandsp] Allow spandsp dtmf > detector to work on rates other than 8k > - FS-7839 <http://ift.tt/1fDUr69> Correct firefox > 38 DTLS behavior > to match new EC requirements > - FS-7769 <http://ift.tt/1TMWpQR> [mod_conference] Fixed vmute on > personal canvas and fixed changing layouts on personal canvas > - FS-7893 <http://ift.tt/1hIBd0C> [mod_conference] Fixed a bug causing > muxing write thread to occasionally not close on shutdown > - FS-7904 <http://ift.tt/1hIBd0E> Fixed alpha image patching > - FS-7906 <http://ift.tt/1Pi1JbP> [mod_av] Correct crash from > multi-threaded opening or closing of multiple files at the same time > - FS-7913 <http://ift.tt/1hIBd0I> [mod_conference] Fixed miscast > variable > - FS-7918 <http://ift.tt/1hIBbG4> [mod_kazoo] Small fixes in mod_kazoo > - FS-7917 <http://ift.tt/1Pi1JbR> [mod_sofia] Fixed default config, we > really shouldn’t be setting ext-*-ip settings for ipv6 profiles > - FS-7908 <http://ift.tt/1hIBdgY> FS-7092 <http://ift.tt/1xMpbcg> > Fixed the generated sdp including telephone-event for the rates of video > codecs (90000) when it should only be audio codec rates > - FS-7927 <http://ift.tt/1hIBdh0> Fixed a typo in variable name: > eavesdrop_annnounce_macro > - FS-7940 <http://ift.tt/1IZhuk8> [mod_conference] Fixed an issue > where the video image does not appear on the new canvas when switching > - FS-7930 <http://ift.tt/1WzTtcv> [mod_conference] Correct termination > of conference when the last member with endconf left. > - FS-7953 <http://ift.tt/1IZhsbU> [verto communicator] Fixed dialing > when typing extension using the keyboard. > - FS-7958 <http://ift.tt/1WzTr4l> [mod_conference] Fixed a race > condition causing crash in conference video MCU > - FS-7951 <http://ift.tt/1IZhsbW> [mod_rayo] Completely clean up > mod_rayo if it fails to load > - FS-7955 <http://ift.tt/1WzTtcx> [mod_sofia] Fixed a crash caused by > invalid contact when using event to send a notify message > - FS-7970 <http://ift.tt/1hDxBgG> Fixed crash in video_bug_thread > caused by double free > - FS-7971 <http://ift.tt/1PvMyfd> [mod_opus] Fixed a rate mismatch and > correctly advertise telephone-event and CN rates based on the advertised > rates of offered codecs > - FS-7960 <http://ift.tt/1hDxBgI> Fixed check_ice routine in > switch_core_media.c to not use dropped candidates > - FS-7975 <http://ift.tt/1PvMxb7> [mod_voicemail] Fix record-greeting > event missing VM-Greeting-Path > - FS-7969 <http://ift.tt/1hDxySb> Fixed a segfault due to > pthread_setschedparam() on a thread that has exited > - FS-7962 <http://ift.tt/1PvMxb9> Fixed sporadic invite/replaces > failure > - FS-8004 <http://ift.tt/1hDxz8p> Send keyframe on receiving nack with > multiple consecutive packets > - FS-8005 <http://ift.tt/1PvMyff> [mod_opus] Fix for rare decoder > error when doing PLC, OPUS_GET_LAST_PACKET_DURATION might return 0 > - FS-8006 <http://ift.tt/1hDxz8r> Changed the typedef of > switch_core_video_thread_callback_func_t for consistency > - FS-7932 <http://ift.tt/1PvMxbb> [mod_verto] Removed the param from > the getMute function in verto class, not needed on underlying method > - FS-8008 <http://ift.tt/1hDxz8t> [mod_verto] Separate verto default > config to have sep v4 and v6 listeners > - FS-8016 <http://ift.tt/1PvMyvu> [mod_conference] Reduce buffering of > video in conference mux > - FS-7977 <http://ift.tt/1hDxBgK> [verto communicator] Fixing default > resolution and cleaning code > - FS-7992 <http://ift.tt/1PvMxbd> [verto communicator] Fixed device > list at settings > - FS-8017 <http://ift.tt/1hDxz8v> [verto communicator] Fixed uses of > serialized verto in local storage > - FS-7986 <http://ift.tt/1PvMyvw> [verto communicator] Fix for devices > not refreshing if system config changes > - FS-7998 <http://ift.tt/1hDxBgO> [verto communicator] Don’t prompt > when recovering call, just do it. > - FS-8003 <http://ift.tt/1PvMxrs> [verto communicator] Use > audioInDevices instead of audioDevices to match verto plugin > - FS-8027 <http://ift.tt/1EArfc6> [verto communicator] Added watchTask > flag to browserSync and add proper regex for replacements > - FS-8026 <http://ift.tt/1EArcx0> [verto_communicator] Added an > auto-focus directive to both dial-pad and login so that enter will just > work. On dial-pad useful to provide keyboard only input without the need to > using the mouse > - FS-7995 <http://ift.tt/1hw80FK> [verto_communicator] Upon call > recovery, emit an event on $rootScope so that controllers are able to > properly clear states. > - FS-7945 <http://ift.tt/1hw80FM> [verto communicator] Use > angular-prompt to ask the user to enter a text for the banner. If > cancelled, nothing is done. > - FS-8045 <http://ift.tt/1EArcx4> [verto communicator] Make the folder > structure compliant with AngularJS best practices and adjust build system. > - FS-7957 <http://ift.tt/1EArfcc> [verto_communicator] Make console > less chatty by commenting liveArray updates and get initial state of the > conference on liveArray boot event. > - FS-7979 <http://ift.tt/1EArfce> [verto_communicator] Prompt for > extension before transferring a conference member > - FS-8001 <http://ift.tt/1hw834i> [verto_communicator] For this to > work, passing in the parameter was missing > - FS-7979 <http://ift.tt/1EArfce> [verto_communicator] Removed extra > console.log and commented line > - FS-8025 <http://ift.tt/1EArfcg> [verto_communicator] Restored the > blue background on the video controls and making icons white again, looking > better. > - FS-8062 <http://ift.tt/1EArfsw> [verto_communicator] Fixed video > controls tool-tips, now they are visible > - FS-8048 <http://ift.tt/1EArfsA> [verto_communicator] Fixed infinite > reconnect after changing hostname and websocket url > - FS-8066 <http://ift.tt/1EAri7K> [verto communicator] Added encoded > avatar url to userVariables so that mod_conference can use it when no > video, or video mute > - FS-8018 <http://ift.tt/1hw80W8> [verto_communicator] Separation of > concerns. Get storage service to manage all settings instead of vertoService > - FS-8043 <http://ift.tt/1EAri7Q> [verto_communicator] Removed > unnecessary calls to localStorage > - FS-8040 <http://ift.tt/1hw834p> [verto_communicator] Check if we > have a valid resolution reported before calling camera routines and hide > controls if none are found > - FS-8092 <http://ift.tt/1EArfsK> [verto_communicator] If there is no > data in localStorage, select best resolution for selected camera > - FS-7840 <http://ift.tt/1EAri7Y> [verto_communicator] Use chatChannel > to send and receive messages from conferences > - FS-8088 <http://ift.tt/1hw834t> [verto_communicator] Call conference > destroy upon hangup and on event destroy to properly unsubscribe from events > - FS-8046 <http://ift.tt/1hw80Wm> [verto] Fixed for library not > passing device preferences to dialog properly > - FS-8053 <http://ift.tt/1hw834v> [verto] Don’t receive video on > screen share > - FS-8059 <http://ift.tt/1EArfJ6> [verto] Fixed typo when transferring > party from conference > - FS-8060 <http://ift.tt/1hw83kN> [verto] Conditionally set video tag > src to null for FF and empty string for others > - FS-8087 <http://ift.tt/1hw83kP> [verto] Fixed issue in camera > selection on recovery, refactor to use localStorage, change res detection, > reload on network change, pass resCheck callback into deviceParams, always > make one basic call to getusermedia to ensure perms are ok, pass valid res > to callback, make $.FSRTC.validRes available globally, sync minified, fix > depth issue in cached json, test for valid cache before setting vars > - FS-8028 <http://ift.tt/1JBF6ur> [mod_shout] Fixed random sockets > being closed regression from FS-7621 > - FS-8029 <http://ift.tt/1hw83kR> [jitterbuffer] Fixed robotic sound > when using jitterbuffer when buffer timestamps get behind that of the > packet timestamps, such as when the source clock is out of sync with our > clock > - FS-8056 <http://ift.tt/1EAriok> [mod_voicemail] Fixed a segfault on > vm_inject, regression from FS-7968 > - FS-7968 <http://ift.tt/1hw83kV> [mod_voicemail] Fixed verbose events > - FS-7942 <http://ift.tt/1EAriom> [udptl] Fixed rare segfault on t.38 > fax FS-8014 is a duplicate of this issue > - FS-8031 <http://ift.tt/1hw81cK> [dtls] Fixed delayed DTLS media due > to changing ICE candidates > - FS-7903 <http://ift.tt/1EArios> [proxy_media] Fix Codec PROXY Exists > but not at the desired implementation. 0hz 0ms 1ch error when using proxy > media. > - FS-7989 <http://ift.tt/1EArfJd> [fixbug.pl] Add –author option > - FS-8037 <http://ift.tt/1JBF3P9> [mod_sofia] Fixed so zrtp-passthru > doesn’t activate unless the zrtp-hash is in the SDP > - FS-7135 <http://ift.tt/1EArfJf> [mod_sofia] Fixed response to > re-invite with duplicate sdp (such as we get from session refresh) when soa > is disabled to include an sdp. Fixed t.38 fax failure on session refresh > - FS-8050 <http://ift.tt/1EArfJh> [mod_av] Fixed a crash when > streaming rtmp of desktop share > - FS-7640 <http://ift.tt/1hw83l1> [rtp] Fixed some comparisons in > switch_rtp.c to be wraparound proof > - FS-8057 <http://ift.tt/1EArfJj> [core] Fixed a segfault when doing > video call when built against libyuv but not libvpx > - FS-8069 <http://ift.tt/1hw81cU> [stun] Fixed ipv6 support missing in > stun code > - FS-8071 <http://ift.tt/1EArfZz> [rtp] remove unnecessary auto adjust > remote address rules when in ICE mode > - FS-8077 <http://ift.tt/1hw83Bl> [mod_conference] Fix memory leak in > record > - FS-8091 <http://ift.tt/1EArfZF> [core] Added some missing message > names to session message name list (would have caused missing information > in some log messages) > - FS-8093 <http://ift.tt/1hw81tc> [mod_silk] Remove giant stack > allocation in switch_silk_decode > > > > _______________________________________________ > 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 > > -- Luke Milbourne <luke.milbourne(a)gmail.com> Tel: 07857154817 Google Talk/Email: luke.milbourne(a)gmail.com

2 1

Push notification from freeswitch
by Murugan Pandian 27 Oct '15

27 Oct '15

Hi , It's possible freeswitch send push notification to mobile sip client, It's possible in mod_verto ? .Thanks for your good time.

4 4

Demo lab for connectivity tests
by Stanislav Sinyagin 26 Oct '15

26 Oct '15

I've set up a small demo lab that you can use for checking the DTMF delivery and two-way speech. Also it demonstrates the software for passive voice quality analysis by Sevana (http://sevana.biz/) http://voxserv.ch/demolab.html Feel free to use it in your tests -- it is online most of the time. If someone wants to attach a new DID to it, feel free to contact me directly. The lab is running some recent version of FreeSWITCH master branch, and is eventually upgraded. I will probably add some video features in the future. cheers, stanislav

1 0

FreeSWITCH Week in Review (Master Branch) October 17th-October 24th
by Ken Rice 26 Oct '15

26 Oct '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1GtSBla FreeSWITCH Week in Review (Master Branch) October 17th-October 24th The FreeSWITCH team added some neat new features this week including: expanded support for limit_* functionality in mod_hiredis and the addition of the field conferenceMemberID to the event broadcasted to inform a verto client about joining a conference. Join us Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-8375 [mod_conference] Add the field conferenceMemberID to the event broadcasted to inform a verto client about joining a conference. FS-8377 [mod_hiredis] Adding expanded support for limit_* functionality Improvements in build system, cross platform support, and packaging: FS-8298 [mod_gsmopen] Fixed a build error FS-8362 [Debian] Now if you install with freeswitch-all you will get the default fonts too The following bugs were squashed: FS-8363 [configuration] Don’t register gateways from directory because it registers over what appears to be ipv6 but doesn’t work correctly FS-8365 [verto_communicator] Fixed a bug with the chat notifications not going away unless you exited and came back to it FS-8336 [verto_communicator] Updating mic and video overlay controls upon receiving member update from live array and use conferenceMemberID when checking if the updated member is the local user FS-8307 [mod_conference] Fixed an issue with the order of codecs causing loss of RTP stream FS-8280 [mod_conference] Fixed an issue with FS sending redundant stop-recording event notifications FS-8130 [mod_conference] Fixed a regression causing excessive mark bit detection in some cases FS-8384 [mod_conference] Fixed some locking contention issues between external commands and the video engine FS-8338 [core] Fixed an issue when setting the ringback variable with an outbound call via the bridge app, if the inbound leg is stereo the ringback tone is still rendered as mono causing the resulting ringback to be higher pitched and incorrect. FS-8366 [core] Fixed a segfault in rxfax FS-8275 [core] Fixed an issue with broken RFC2833 DTMF FS-8368 [core] Reduce logging for audio/video sync because some call lines were repeating too often for callers in a conference FS-8372 [core] Fixed a no media bug caused by sofia sending the wrong response to RTP/SAVPF without DTLS FS-8381 [core] Reset jitter buffer if period loss is too high FS-8382 [core] Fixed a segfault with inbound-proxy-media enabled FS-8378 [mod_esf] [core] Fixed a crash when using esf_page over loopback when transcoding and added tests for esf over loopback. Also refactor a bit to clarify code and get better debug in gdb FS-8115 [mod_sofia] Disabled unnecessary session timer re-invites for webrtc FS-8370 [mod_rayo] Fixed another place in where a message was freed after being queued for delivery. This resulted in a freed object being serialized, crashing FS FS-8287 [mod_opus] Fixed a segfault from refactor The FreeSWITCH 1.4 branch had a few bug fixes added this week. The following bugs were squashed: FS-8335 [mod_easyroute] Fixed a small error check that results in error message not being displayed FS-8370 [mod_rayo] Fixed another place in where a message was freed after being queued for delivery. This resulted in a freed object being serialized, crashing FS

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early media - no ringback
by Dada P 26 Oct '15

26 Oct '15

Hello, I have encountered strange problem - there is no audible ringback on DID routing. Dialplan is simple, on DID call gets transferred to local extension. Local extension is pretty much vanilla. Log shows that both 183/180 is being fired and ringback activated but no sound coming. Ringing is going ok though, as well as connection afterwords. Is that incoming provider's problem or freeswitch? Regards, Paul Kazakov Log: Dialplan: sofia/external/mobile@provider Action transfer(20 XML default) 2015-10-26 01:39:27.463856 [DEBUG] switch_core_state_machine.c:216 (sofia/external/mobile@provider) State Change CS_ROUTING -> CS_EXECUTE 2015-10-26 01:39:27.463856 [DEBUG] switch_core_session.c:1397 Send signal sofia/external/mobile@provider [BREAK] 2015-10-26 01:39:27.463856 [DEBUG] switch_core_state_machine.c:528 (sofia/external/mobile@provider) State ROUTING going to sleep 2015-10-26 01:39:27.463856 [DEBUG] switch_core_state_machine.c:472 (sofia/external/mobile@provider) Running State Change CS_EXECUTE 2015-10-26 01:39:27.463856 [DEBUG] switch_core_state_machine.c:535 (sofia/external/mobile@provider) State EXECUTE 2015-10-26 01:39:27.463856 [DEBUG] mod_sofia.c:178 sofia/external/mobile@provider SOFIA EXECUTE 2015-10-26 01:39:27.463856 [DEBUG] switch_core_state_machine.c:258 sofia/external/mobile@provider Standard EXECUTE EXECUTE sofia/external/mobile@provider set(outside_call=true) 2015-10-26 01:39:27.463856 [DEBUG] mod_dptools.c:1477 sofia/external/mobile@provider SET [outside_call]=[true] EXECUTE sofia/external/mobile@provider export(RFC2822_DATE=Mon, 26 Oct 2015 01:39:27 +0100) 2015-10-26 01:39:27.483860 [DEBUG] switch_channel.c:1267 EXPORT (export_vars) [RFC2822_DATE]=[Mon, 26 Oct 2015 01:39:27 +0100] EXECUTE sofia/external/mobile@provider set(domain_name=192.168.1.5) 2015-10-26 01:39:27.483860 [DEBUG] mod_dptools.c:1477 sofia/external/mobile@provider SET [domain_name]=[192.168.1.5] EXECUTE sofia/external/mobile@provider transfer(20 XML default) 2015-10-26 01:39:27.483860 [DEBUG] switch_ivr.c:1856 (sofia/external/mobile@provider) State Change CS_EXECUTE -> CS_ROUTING 2015-10-26 01:39:27.483860 [DEBUG] switch_core_session.c:1397 Send signal sofia/external/mobile@provider [BREAK] 2015-10-26 01:39:27.483860 [DEBUG] switch_core_session.c:913 Send signal sofia/external/mobile@provider [BREAK] 2015-10-26 01:39:27.483860 [NOTICE] switch_ivr.c:1863 Transfer sofia/external/mobile@provider to XML[20@default] 2015-10-26 01:39:27.483860 [DEBUG] switch_core_state_machine.c:535 (sofia/external/mobile@provider) State EXECUTE going to sleep 2015-10-26 01:39:27.483860 [DEBUG] switch_core_state_machine.c:472 (sofia/external/mobile@provider) Running State Change CS_ROUTING 2015-10-26 01:39:27.483860 [DEBUG] switch_core_state_machine.c:528 (sofia/external/mobile@provider) State ROUTING 2015-10-26 01:39:27.483860 [DEBUG] mod_sofia.c:123 sofia/external/mobile@provider SOFIA ROUTING 2015-10-26 01:39:27.483860 [DEBUG] switch_core_state_machine.c:166 sofia/external/mobile@provider Standard ROUTING 2015-10-26 01:39:27.483860 [INFO] mod_dialplan_xml.c:635 Processing mobile <mobile>->20 in context default Dialplan: sofia/external/mobile@provider parsing [default->unloop] continue=false Dialplan: sofia/external/mobile@provider Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/mobile@provider Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false Dialplan: sofia/external/mobile@provider Absolute Condition [global] Dialplan: sofia/external/mobile@provider Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/external/mobile@provider Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/external/mobile@provider Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/external/mobile@provider Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/mobile@provider parsing [default->Local_Extension] continue=false Dialplan: sofia/external/mobile@provider Regex (PASS) [Local_Extension] destination_number(20) =~ /^([2-5][0-9])$/ break=on-false Dialplan: sofia/external/mobile@provider Action export(dialed_extension=20) Dialplan: sofia/external/mobile@provider Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/external/mobile@provider Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/external/mobile@provider Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/external/mobile@provider Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/external/mobile@provider Action set(ringback=%(1000,4000,425)) Dialplan: sofia/external/mobile@provider Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/external/mobile@provider Action set(call_timeout=30) Dialplan: sofia/external/mobile@provider Action set(hangup_after_bridge=true) Dialplan: sofia/external/mobile@provider Action set(continue_on_fail=true) Dialplan: sofia/external/mobile@provider Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: sofia/external/mobile@provider Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/external/mobile@provider Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/external/mobile@provider Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) Dialplan: sofia/external/mobile@provider Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) Dialplan: sofia/external/mobile@provider Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/external/mobile@provider Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: sofia/external/mobile@provider Action answer() Dialplan: sofia/external/mobile@provider Action sleep(1000) Dialplan: sofia/external/mobile@provider Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 2015-10-26 01:39:27.483860 [DEBUG] switch_core_state_machine.c:216 (sofia/external/mobile@provider) State Change CS_ROUTING -> CS_EXECUTE 2015-10-26 01:39:27.483860 [DEBUG] switch_core_session.c:1397 Send signal sofia/external/mobile@provider [BREAK] 2015-10-26 01:39:27.483860 [DEBUG] switch_core_state_machine.c:528 (sofia/external/mobile@provider) State ROUTING going to sleep 2015-10-26 01:39:27.483860 [DEBUG] switch_core_state_machine.c:472 (sofia/external/mobile@provider) Running State Change CS_EXECUTE 2015-10-26 01:39:27.483860 [DEBUG] switch_core_state_machine.c:535 (sofia/external/mobile@provider) State EXECUTE 2015-10-26 01:39:27.483860 [DEBUG] mod_sofia.c:178 sofia/external/mobile@provider SOFIA EXECUTE 2015-10-26 01:39:27.483860 [DEBUG] switch_core_state_machine.c:258 sofia/external/mobile@provider Standard EXECUTE EXECUTE sofia/external/mobile@provider hash(insert/192.168.1.5-spymap/mobile/0335befe-7b7a-11e5-a603-cb96e1b49eeb) EXECUTE sofia/external/mobile@provider hash(insert/192.168.1.5-last_dial/mobile/20) EXECUTE sofia/external/mobile@provider hash(insert/192.168.1.5-last_dial/global/0335befe-7b7a-11e5-a603-cb96e1b49eeb) EXECUTE sofia/external/mobile@provider export(RFC2822_DATE=Mon, 26 Oct 2015 01:39:27 +0100) 2015-10-26 01:39:27.483860 [DEBUG] switch_channel.c:1267 EXPORT (export_vars) [RFC2822_DATE]=[Mon, 26 Oct 2015 01:39:27 +0100] EXECUTE sofia/external/mobile@provider export(dialed_extension=20) 2015-10-26 01:39:27.503879 [DEBUG] switch_channel.c:1267 EXPORT (export_vars) [dialed_extension]=[20] EXECUTE sofia/external/mobile@provider bind_meta_app(1 b s execute_extension::dx XML features) 2015-10-26 01:39:27.503879 [INFO] switch_ivr_async.c:3932 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE sofia/external/mobile@provider bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/mobile.2015-10-26-01-39-27.wav) 2015-10-26 01:39:27.503879 [INFO] switch_ivr_async.c:3932 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/mobile.2015-10-26-01-39-27.wav EXECUTE sofia/external/mobile@provider bind_meta_app(3 b s execute_extension::cf XML features) 2015-10-26 01:39:27.503879 [INFO] switch_ivr_async.c:3932 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE sofia/external/mobile@provider bind_meta_app(4 b s execute_extension::att_xfer XML features) 2015-10-26 01:39:27.503879 [INFO] switch_ivr_async.c:3932 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE sofia/external/mobile@provider set(ringback=%(1000,4000,425)) 2015-10-26 01:39:27.503879 [DEBUG] mod_dptools.c:1477 sofia/external/mobile@provider SET [ringback]=[%(1000,4000,425)] EXECUTE sofia/external/mobile@provider set(instant_ringback=true) 2015-10-26 01:39:27.503879 [DEBUG] mod_dptools.c:1477 sofia/external/mobile@provider SET [instant_ringback]=[true] EXECUTE sofia/external/mobile@provider set(transfer_ringback=local_stream://moh) 2015-10-26 01:39:27.503879 [DEBUG] mod_dptools.c:1477 sofia/external/mobile@provider SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/external/mobile@provider set(call_timeout=30) 2015-10-26 01:39:27.503879 [DEBUG] mod_dptools.c:1477 sofia/external/mobile@provider SET [call_timeout]=[30] EXECUTE sofia/external/mobile@provider set(hangup_after_bridge=true) 2015-10-26 01:39:27.503879 [DEBUG] mod_dptools.c:1477 sofia/external/mobile@provider SET [hangup_after_bridge]=[true] EXECUTE sofia/external/mobile@provider set(continue_on_fail=true) 2015-10-26 01:39:27.523862 [DEBUG] mod_dptools.c:1477 sofia/external/mobile@provider SET [continue_on_fail]=[true] EXECUTE sofia/external/mobile@provider hash(insert/192.168.1.5-call_return/20/mobile) EXECUTE sofia/external/mobile@provider hash(insert/192.168.1.5-last_dial_ext/20/0335befe-7b7a-11e5-a603-cb96e1b49eeb) EXECUTE sofia/external/mobile@provider set(called_party_callgroup=bakery) 2015-10-26 01:39:27.523862 [DEBUG] mod_dptools.c:1477 sofia/external/mobile@provider SET [called_party_callgroup]=[bakery] EXECUTE sofia/external/mobile@provider hash(insert/192.168.1.5-last_dial_ext/bakery/0335befe-7b7a-11e5-a603-cb96e1b49eeb) EXECUTE sofia/external/mobile@provider hash(insert/192.168.1.5-last_dial_ext/global/0335befe-7b7a-11e5-a603-cb96e1b49eeb) EXECUTE sofia/external/mobile@provider hash(insert/192.168.1.5-last_dial/bakery/0335befe-7b7a-11e5-a603-cb96e1b49eeb) EXECUTE sofia/external/mobile@provider bridge(user/20(a)192.168.1.5) 2015-10-26 01:39:27.523862 [DEBUG] switch_channel.c:1221 sofia/external/mobile@provider EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 26 Oct 2015 01:39:27 +0100] to event 2015-10-26 01:39:27.523862 [DEBUG] switch_channel.c:1221 sofia/external/mobile@provider EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 26 Oct 2015 01:39:27 +0100] to event 2015-10-26 01:39:27.523862 [DEBUG] switch_channel.c:1221 sofia/external/mobile@provider EXPORTING[export_vars] [dialed_extension]=[20] to event 2015-10-26 01:39:27.523862 [DEBUG] switch_ivr_originate.c:2101 Parsing global variables 2015-10-26 01:39:27.543880 [DEBUG] switch_channel.c:1221 sofia/external/mobile@provider EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 26 Oct 2015 01:39:27 +0100] to event 2015-10-26 01:39:27.543880 [DEBUG] switch_channel.c:1221 sofia/external/mobile@provider EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 26 Oct 2015 01:39:27 +0100] to event 2015-10-26 01:39:27.543880 [DEBUG] switch_channel.c:1221 sofia/external/mobile@provider EXPORTING[export_vars] [dialed_extension]=[20] to event 2015-10-26 01:39:27.543880 [DEBUG] switch_ivr_originate.c:2101 Parsing global variables 2015-10-26 01:39:27.543880 [DEBUG] switch_event.c:1698 Parsing variable [sip_invite_domain]=[192.168.1.5] 2015-10-26 01:39:27.543880 [DEBUG] switch_event.c:1698 Parsing variable [presence_id]=[20(a)192.168.1.5] 2015-10-26 01:39:27.543880 [NOTICE] switch_channel.c:1075 New Channel sofia/internal/20@192.168.1.13:5060 [034101c4-7b7a-11e5-a624-cb96e1b49eeb] 2015-10-26 01:39:27.543880 [DEBUG] mod_sofia.c:4700 (sofia/internal/20@192.168.1.13:5060) State Change CS_NEW -> CS_INIT 2015-10-26 01:39:27.543880 [DEBUG] switch_core_session.c:1397 Send signal sofia/internal/20@192.168.1.13:5060 [BREAK] 2015-10-26 01:39:27.543880 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/20@192.168.1.13:5060) Running State Change CS_INIT 2015-10-26 01:39:27.543880 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/20@192.168.1.13:5060) State INIT 2015-10-26 01:39:27.543880 [DEBUG] mod_sofia.c:87 sofia/internal/20@192.168.1.13:5060 SOFIA INIT 2015-10-26 01:39:27.543880 [DEBUG] sofia_glue.c:1241 sofia/internal/20@192.168.1.13:5060 sending invite version: 1.4.23 git 1665168 2015-10-01 18:31:59Z 32bit Local SDP: v=0 o=FreeSWITCH 1445799433 1445799434 IN IP4 192.168.1.5 s=FreeSWITCH c=IN IP4 192.168.1.5 t=0 0 m=audio 20534 RTP/AVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2015-10-26 01:39:27.543880 [DEBUG] switch_core_state_machine.c:40 sofia/internal/20@192.168.1.13:5060 Standard INIT 2015-10-26 01:39:27.543880 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/20@192.168.1.13:5060) State Change CS_INIT -> CS_ROUTING 2015-10-26 01:39:27.543880 [DEBUG] switch_core_session.c:1397 Send signal sofia/internal/20@192.168.1.13:5060 [BREAK] 2015-10-26 01:39:27.543880 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/20@192.168.1.13:5060) State INIT going to sleep 2015-10-26 01:39:27.543880 [INFO] switch_ivr_originate.c:1193 Sending early media 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:3679 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1] 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:3679 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:3679 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:3734 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:3679 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:3679 Audio Codec Compare [G729:18:8000:20:8000:1]/[opus:116:48000:20:0:1] 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:3679 Audio Codec Compare [G729:18:8000:20:8000:1]/[G722:9:8000:20:64000:1] 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:3679 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1] 2015-10-26 01:39:27.543880 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/20@192.168.1.13:5060) Running State Change CS_ROUTING 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:3679 Audio Codec Compare [G729:18:8000:20:8000:1]/[GSM:3:8000:20:13200:1] 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:3679 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1] 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:3679 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:3679 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:3679 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:3595 Set telephone-event payload to 101 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:2506 Set Codec sofia/external/mobile@provider PCMA/8000 20 ms 160 samples 64000 bits 1 channels 2015-10-26 01:39:27.543880 [DEBUG] switch_core_codec.c:111 sofia/external/mobile@provider Original read codec set to PCMA:8 send 1183 bytes to udp/[192.168.1.13]:5060 at 01:39:27.556739: ------------------------------------------------------------------------ INVITE sip:20@192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5;rport;branch=z9hG4bK01mvc388g3F4e Max-Forwards: 68 From: "mobile" <sip:mobile@192.168.1.5>;tag=mX9UByt7H37DQ To: <sip:20@192.168.1.13:5060> Call-ID: daa0593a-f61c-1233-8791-b827eb32337b CSeq: 82573919 INVITE Contact: <sip:mod_sofia@192.168.1.5:5060> User-Agent: FreeSWITCH-mod_sofia/1.4.23+git~20151001T183159Z~1665168d37~32bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 221 X-Ecan: On X-FS-Support: update_display,send_info Remote-Party-ID: "mobile" <sip:mobile@192.168.1.5>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1445799433 1445799434 IN IP4 192.168.1.5 s=FreeSWITCH c=IN IP4 192.168.1.5 t=0 0 m=audio 20534 RTP/AVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 2015-10-26 01:39:27.543880 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/20@192.168.1.13:5060) State ROUTING 2015-10-26 01:39:27.543880 [DEBUG] mod_sofia.c:123 sofia/internal/20@192.168.1.13:5060 SOFIA ROUTING 2015-10-26 01:39:27.543880 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/20@192.168.1.13:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2015-10-26 01:39:27.543880 [DEBUG] switch_core_session.c:1397 Send signal sofia/internal/20@192.168.1.13:5060 [BREAK] 2015-10-26 01:39:27.543880 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/20@192.168.1.13:5060) State ROUTING going to sleep 2015-10-26 01:39:27.543880 [DEBUG] switch_core_session.c:1062 Send signal sofia/internal/20@192.168.1.13:5060 [BREAK] 2015-10-26 01:39:27.543880 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/20@192.168.1.13:5060) Running State Change CS_CONSUME_MEDIA 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:3942 Set 2833 dtmf send/recv payload to 101 2015-10-26 01:39:27.543880 [INFO] switch_nat.c:588 NAT port mapping disabled 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:5178 AUDIO RTP [sofia/external/mobile@provider] 192.168.1.5 port 29360 -> xx.xx.xx.xx port 59648 codec: 8 ms: 20 2015-10-26 01:39:27.543880 [DEBUG] sofia.c:6658 Channel sofia/internal/20@192.168.1.13:5060 entering state [calling][0] 2015-10-26 01:39:27.543880 [DEBUG] switch_core_state_machine.c:547 (sofia/internal/20@192.168.1.13:5060) State CONSUME_MEDIA 2015-10-26 01:39:27.543880 [DEBUG] switch_core_state_machine.c:547 (sofia/internal/20@192.168.1.13:5060) State CONSUME_MEDIA going to sleep 2015-10-26 01:39:27.543880 [DEBUG] switch_rtp.c:3579 Starting timer [soft] 160 bytes per 20ms 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:5476 Set 2833 dtmf send payload to 101 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:5482 Set 2833 dtmf receive payload to 101 2015-10-26 01:39:27.543880 [DEBUG] switch_core_media.c:5504 sofia/external/mobile@provider Set rtp dtmf delay to 40 2015-10-26 01:39:27.543880 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/mobile@provider! 2015-10-26 01:39:27.563871 [DEBUG] switch_channel.c:3419 (sofia/external/mobile@provider) Callstate Change RINGING -> EARLY 2015-10-26 01:39:27.563871 [DEBUG] mod_sofia.c:2264 Ring SDP: v=0 o=FreeSWITCH 1445790607 1445790608 IN IP4 xx.xx.xx.xx s=FreeSWITCH c=IN IP4 ext_ip_addr t=0 0 m=audio 29360 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2015-10-26 01:39:27.563871 [DEBUG] switch_core_session.c:913 Send signal sofia/external/mobile@provider [BREAK] 2015-10-26 01:39:27.563871 [DEBUG] switch_ivr_originate.c:1250 Raw Codec Activation Success L16@8000hz 1 channel 20ms 2015-10-26 01:39:27.563871 [DEBUG] switch_core_codec.c:221 sofia/external/mobile@provider Push codec L16:100 2015-10-26 01:39:27.563871 [DEBUG] switch_ivr_originate.c:1318 Play Ringback Tone [%(1000,4000,425)] send 1067 bytes to udp/[xx.xx.xx.xx]:5060 at 01:39:27.575578: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 62.80.209.30:5060;branch=z9hG4bK-524287-1---f0231a59e0656a3a;rport=5060 From: "mobile"<sip:mobile@provider;user=phone>;tag=b2a8ce1b To: "pstn_did" <sip:pstn_did@siptrunk;user=phone>;tag=gmUjH321ep8jF Call-ID: pSnT7Y8CK7u_vpaaKgC0bw.. CSeq: 1 INVITE Contact: <sip:pstn_did@ext_ip_addr:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.23+git~20151001T183159Z~1665168d37~32bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 224 Remote-Party-ID: "pstn_did" <sip:pstn_did@siptrunk>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1445790607 1445790608 IN IP4 ext_ip_addr s=FreeSWITCH c=IN IP4 ext_ip_addr t=0 0 m=audio 29360 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 2015-10-26 01:39:27.563871 [DEBUG] switch_core_session.c:1062 Send signal sofia/external/mobile@provider [BREAK] 2015-10-26 01:39:27.583855 [DEBUG] sofia.c:6658 Channel sofia/external/mobile@provider entering state [early][183] recv 370 bytes from udp/[192.168.1.13]:5060 at 01:39:27.628443: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.5;rport=5060;branch=z9hG4bK01mvc388g3F4e From: "mobile" <sip:mobile@192.168.1.5>;tag=mX9UByt7H37DQ To: <sip:20@192.168.1.13:5060>;tag=2174874080 Call-ID: daa0593a-f61c-1233-8791-b827eb32337b CSeq: 82573919 INVITE Contact: <sip:20@192.168.1.13:5060> User-Agent: C530 IP/42.207.00.000.000 Content-Length: 0 ------------------------------------------------------------------------ recv 421 bytes from udp/[192.168.1.13]:5060 at 01:39:28.125276: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.5;rport=5060;branch=z9hG4bK01mvc388g3F4e From: "mobile" <sip:mobile@192.168.1.5>;tag=mX9UByt7H37DQ To: <sip:20@192.168.1.13:5060>;tag=2174874080 Call-ID: daa0593a-f61c-1233-8791-b827eb32337b CSeq: 82573919 INVITE Contact: <sip:20@192.168.1.13:5060> Allow-Events: message-summary, refer, ua-profile User-Agent: C530 IP/42.207.00.000.000 Content-Length: 0 ------------------------------------------------------------------------ 2015-10-26 01:39:28.123847 [DEBUG] switch_core_session.c:1062 Send signal sofia/internal/20@192.168.1.13:5060 [BREAK] 2015-10-26 01:39:28.123847 [DEBUG] switch_core_session.c:1062 Send signal sofia/internal/20@192.168.1.13:5060 [BREAK] 2015-10-26 01:39:28.123847 [DEBUG] sofia.c:6658 Channel sofia/internal/20@192.168.1.13:5060 entering state [proceeding][180] 2015-10-26 01:39:28.123847 [NOTICE] sofia.c:6760 Ring-Ready sofia/internal/20@192.168.1.13:5060! 2015-10-26 01:39:28.123847 [DEBUG] switch_channel.c:3297 (sofia/internal/20@192.168.1.13:5060) Callstate Change DOWN -> RINGING

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originate_timeout/call_timeout not being respected in s sequence like dialstring
by Nuno Reis 26 Oct '15

26 Oct '15

Hi guys. I've noticed today that the originate_timeout on a sequence like dialstring is not respected. originate {originate_timeout=12,domain_name=domain.example.com }[leg_delay_start=0,leg_timeout=10]user/999(a)domain.example.com |[leg_delay_start=0,leg_timeout=10]user/997(a)domain.example.com |[leg_delay_start=0,leg_timeout=10]user/998(a)domain.example.com &park() In a case like this all the endpoints are called and the call ends only after 30 sec with the disposition ALLOTED_TIMEOUT from the last target. Although if I set originate_timeout in the same dialstring let's say to 7 seconds the call ends after 7 seconds with a NO ANSWER disposition (in this case the originate_timeout is indeed respected). originate {originate_timeout=7,domain_name=domain.example.com }[leg_delay_start=0,leg_timeout=10]user/999(a)domain.example.com |[leg_delay_start=0,leg_timeout=10]user/997(a)domain.example.com |[leg_delay_start=0,leg_timeout=10]user/998(a)domain.example.com &park() Should I open a jira for this? Cheers, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* <http://www.wavecom.pt/> <http://www.wavecom.pt/pt/>

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