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not enough buffer space for required resample operation!
by Tim Meade 02 Oct '15

02 Oct '15

Intermediate transcoding server. Debian 8 FreeSWITCH Version 1.7.0+git~20150827T040104Z~eefe21536d~64bit (git eefe215 2015-08-27 04:01:04Z 64bit) We have ONE cell phone number that triggers this. All other numbers in testing have worked fine. I can't seem to find much on this error. Before I open a JIRA I thought I would ask if anyone else had seen this. We are in the process form moving from long term cents to debian. Maybe something was missed. Thanks Tim 2015-10-02 11:52:10.942998 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/2125551212(a)162.213.51.55! 2015-10-02 11:52:10.942998 [NOTICE] switch_ivr_originate.c:3526 Channel [sofia/internal/2125551212(a)162.213.51.55] has been answered 2015-10-02 11:52:11.303018 [NOTICE] switch_core_io.c:1191 Activating write resampler 2015-10-02 11:52:11.342999 [NOTICE] switch_core_io.c:1191 Activating write resampler 2015-10-02 11:52:11.342999 [CRIT] switch_core_io.c:1261 sofia/internal/2125551212(a)162.213.51.55 not enough buffer space for required resample operation! 2015-10-02 11:52:11.342999 [NOTICE] switch_core_io.c:1263 Hangup sofia/internal/2125551212(a)162.213.51.55 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] # ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) 15798 max locked memory (kbytes, -l) 64 max memory size (kbytes, -m) unlimited open files (-n) 65536 pipe size (512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 real-time priority (-r) 0 stack size (kbytes, -s) 8192 cpu time (seconds, -t) unlimited max user processes (-u) 15798 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited

1 0

Non regression tests?
by fayçal noushi 02 Oct '15

02 Oct '15

Hi all, A long time ago, I opened a ticket for an issue that I had. I wanted to extract SDP from multipart messages (this was in a SIP-I scenario case). To do so, I've had to modify a function in nua_session.c. I'm not used to C programming, but I made sure the modifications only impacted the SDP multipart case which isn't supported anyway. The thing is, it was an important step to make a SIP-I call go through FreeSwitch. I've discussed this in verto with you, people. But, as I understood it, this couldn't be merged since it had to undergo regression testing. My question is this: how can I make those tests? Since then, the pull request got declined since I had to redo the same thing with FS 1.6. I didn't do it since I don't know it's really useful. Thanks, -- NOUSHI Fayçal

1 0

How to test a Pull Request! aka how to help us out!
by Brian West 02 Oct '15

02 Oct '15

Checkout freeswitch master first! git clone https://freeswitch.org/stash/scm/fs/freeswitch.git freeswitch.git cd freeswitch.git git config --add remote.origin.fetch '+refs/pull-requests/*/from:refs/remotes/origin/pr/*' git config alias.pr '!f() { git checkout pr/$1; git config branch.pr/$1.merge refs/pull-requests/$1/from;} ; f' git pr <PR_NUMBER> build, test, report back. You can view the list of pull requests here: https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests Questions? Comments? Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> | Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

3 4

kamailio not starting
by Vikas Kumar 02 Oct '15

02 Oct '15

Hi all, Getting error : 0(32420) ERROR: db_mysql [km_my_con.c:114]: db_mysql_new_connection(): driver error: Access denied for user 'kamailio'@'localhost' (using password: YES) 0(32420) ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool 0(32420) ERROR: usrloc [dlist.c:654]: register_udomain(): failed to open database connection 0(32420) ERROR: registrar [reg_mod.c:563]: domain_fixup(): failed to register domain 0(32420) ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:609 0(32420) ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:611 0(32420) ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:615 Any idea to resolve this. -- Regards, Vikas Kumar

2 1

Is there a channel variable available to a dialplan that contains the current number of active sessions? ie. matches the command: "show channels count"
by Andrew Keil 02 Oct '15

02 Oct '15

To FreeSWITCH users, I just wondered if there is a way to return a count of the number of active sessions as a channel variable for use within a diaplan? >From the FreeSWITCH console or FS_CLI the command "show channels count" returns <x> total (where <x> is the current number of active sessions and it is the value of <x> that I would like to use). Please note: I do not require to check against any limits, I just need this to expose the number of concurrent calls (including the session that is running) to the dialplan script. Thanks in advance, Andrew Keil Visytel Pty Ltd

1 0

Detect source IP of a call
by Dominique Jeannerod 01 Oct '15

01 Oct '15

Hello, I think it's a kind of newbie question, but i'm trying to find the simplest way of detecting the source of a call (public trunks), which of course rely on the source IP of the call. I already defined one ACL per trunk, to authorize only allowed IP adresses, and test the different acls in the dialplan to identify the source trunk of a call, and set the associated accountcode. Example : <condition field="${acl(${network_addr} T_CNX_VPN)}" expression="true" break="on-true"> <action inline="true" application="set" data="accountcode=T_CNX"/> </condition> It works but the dialplan needs to be modified for each new trunk. I am wondering if there is a best way to do that, using for example a channel variable, containing the ACL which matched for the call. I tested sip_acl_authed_by, but this seems to be available only when the call is authenticated (?) I'm sure a lot of people had the same question What would be your advice ? D. Jeannerod

2 1

Compile error on 1.4 branch in mod_dptools (Ubuntu 14.04)
by Mark Boots 01 Oct '15

01 Oct '15

Using our existing build scripts, we started seeing this compile error today when building the 1.4 branch on Ubuntu 14.04. Has anyone else come across this? Thanks! Regards, +Mark making all mod_dptools make[4]: Entering directory `/usr/src/freeswitch/src/mod/applications/mod_dptools' CC mod_dptools_la-mod_dptools.lo In file included from ../../../../src/include/switch.h:130:0, from mod_dptools.c:40: mod_dptools.c: In function 'mod_dptools_shutdown': mod_dptools.c:5832:29: error: 'FILE_STRING_CLOSE' undeclared (first use in this function) switch_event_free_subclass(FILE_STRING_CLOSE); ^ ../../../../src/include/switch_event.h:375:97: note: in definition of macro 'switch_event_free_subclass' #define switch_event_free_subclass(subclass_name) switch_event_free_subclass_detailed(__FILE__, subclass_name) ^ mod_dptools.c:5832:29: note: each undeclared identifier is reported only once for each function it appears in switch_event_free_subclass(FILE_STRING_CLOSE); ^ ../../../../src/include/switch_event.h:375:97: note: in definition of macro 'switch_event_free_subclass' #define switch_event_free_subclass(subclass_name) switch_event_free_subclass_detailed(__FILE__, subclass_name) ^ mod_dptools.c:5833:29: error: 'FILE_STRING_FAIL' undeclared (first use in this function) switch_event_free_subclass(FILE_STRING_FAIL); ==================================== Mark Boots, PhD | CEO and CTO, VOTO Mobile PO Box KS17672, Kumasi, Ashanti Region Ghana +233 26 416 4182 40 King St. W, Toronto, Ontario Canada M5H 3Y4 +1 639 471 0295 e: mark(a)votomobile.org w: http://votomobile.org skype: markjboots twitter: @markjboots ====================================

9 18

Clean installation of FS 1.6.2 fails to start
by Michael Nielsen 01 Oct '15

01 Oct '15

I've setup a clean Debian 8 with FS 1.6.2. When trying to start I get the following error: 2015-10-01 06:26:13.544697 [ERR] sofia.c:2945 Error Creating SIP UA for profile: internal (sip:mod_sofia@139.125.289.11:5060;transport=udp,tcp) The likely causes for this are: 1) Another application is already listening on the specified address. 2) The IP the profile is attempting to bind to is not local to this system. No other application is running on port 5060, I've also tried to change this - same error still occurs on other ports. The IP profile cause, I'm not sure what that means. The public port 5080 does get binded correctly. My internal.xml contains $${local_ip_v4} for IP address.

5 11

Freeswitch send UDP to port outside range
by Charles Bujold 01 Oct '15

01 Oct '15

We are encountering an error which we do not know how to fix. If somebody can help, it would be appreciated. Our configuration is we have 2 offices. Both offices are joined together via a VPN. The users in the remote office use Freeswitch via the VPN and connect to Freeswitch SIP port 5060 via the VPN. Their phones register without issue. The issue comes when they try to make a call. They connect to Freeswitch via SIP without error however early in the connection Freeswitch no longer recognizes them as being local and tries to communicate with them via the WAN. Worst case we could open the firewall to permit such communication but the issue with that, for some reason Freeswitch no longer uses the UDP port range set in Freeswitch it uses a port outside of the range causing the call to fail. Our acl.config has both Lan entered into it 192.168.20.0/24 (Main Office) and 192.168.25.0/24 (Remote Office) however the main office lan is set to deny, we presume it is because one of the default list already includes it. Here is a pcap summary of what we see. How can we setup so that the remote office will work every time and still be seen as part of the overall local office? 192.168.25.18 is a remote phone 192.168.20.153 is Freeswitch server in main office. 142.162.8.143 is our WAN IP Port 49790 is outside of the max port which is 32768 83 12.279953 192.168.25.18 192.168.20.153 SIP/SDP 935 Request: INVITE sip:*97@192.168.20.153 84 12.375683 192.168.20.153 192.168.25.18 SIP 375 Status: 100 Trying | 85 12.376097 192.168.20.153 192.168.25.18 SIP 880 Status: 407 Proxy Authentication Required 86 12.393746 192.168.25.18 192.168.20.153 SIP 318 Request: ACK sip:*97@192.168.20.153 87 12.458854 192.168.25.18 192.168.20.153 SIP/SDP 1181 Request: INVITE sip:*97@192.168.20.153 88 12.542911 192.168.20.153 192.168.25.18 SIP 375 Status: 100 Trying 89 12.718778 192.168.20.153 192.168.25.18 SIP/SDP 1153 Status: 200 OK 90 12.752832 192.168.25.18 142.162.8.143 SIP 680 Request: ACK sip:*97@142.162.8.143:49790;transport=udp Our acl.conf file <configuration name="acl.conf" description="Network Lists"> <network-lists>  <list name="lan" default="allow"> <node type="deny" cidr="192.168.20.0/24"/> <node type="allow" cidr="192.168.25.0/24"/> </list>  <list name="domains" default="deny">  <node type="allow" domain="$${domain}"/>    </list> </network-lists> </configuration> If you can tell us how we should configure Freeswitch to work for both offices it would be appreciated. Thanks cjb

3 3

The FreeSWITCH 1.6.0 release is here!
by krice＠freeswitch.org 01 Oct '15

01 Oct '15

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 [mod_sofia] Add support for Remote-Party-ID header in UPDATE request. FS-7561 [mod_sofia] Add Perfect Forward Secrecy (DHE PFS) FS-7560 [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 [ftmod_libpri] Correctly set calling number presentation and screening fields FS-7138 [mod_callcenter] Added a new reserve-agents param FS-7436 FS-7601 [mod_opus] FEC support FS-7623 [mod_amqp] Allow for custom exchange name and type for producers and fixed param name ordering bug caused by exposing these params FS-7638 Allow ipv4 mapped ipv6 address to pass ipv4 ACLs properly FS-7643 [mod_opus] Added interpretation of maxplaybackrate and sprop-maxcapturerate FS-7641 Added video support to eavesdrop FS-7656 [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 [mod_conference] Added member status in json format to the conference live array, add livearray-json-status to conference-flags to enable FS-7517 FS-7519 [mod_av] [mod_openh264] Added H264 STAP-A packeting support so it would work with FireFox FS-7664 [mod_verto] Set ICE candidate timeout to wait for only 1 second to fix media delays FS-7660 [mod_opus] Enabled with new API command “opus_debug” to show information about Opus payload for debugging. FS-7519 [mod_av] Fixed bitrate and added some presets FS-7693 [mod_conference] Lower the default energy level in sample configs to improve voice quality FS-7720 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 Continue recording with uuid_transfer FS-7752 [mod_rayo] Increase maximum number of elements from 30 to 1024 to allow adhearsion to create large grammars to navigate IVR menus. FS-7750 [mod_commands] Allow for uuid_setvar to handle arrays FS-7758 [mod_loopback] Emit an event if a loopback bowout occurs FS-7759 [mod_sofia] Added the channel variable ignore_completed_elsewhere to suppress setting the completed elsewhere cause FS-7771 Set a channel variable if the recording is terminated due to silence hits FS-7760 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 [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 Add mod_kazoo FS-7780 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 Add channel variable for capturing DTMF input when using play_and_get_digits when the response does not match FS-7772 [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 [mod_png] Add API command uuid_write_png FS-7801 [mod_opus] Added support to set CBR mode FS-7685 [mod_say_nl] Fix Dutch numbers pronunciation FS-7198 Add coma separated values and reverse ranges for time-of-day and day-of-week matches FS-7809 [mod_opus] Added 60 ms ptime for Opus at 8 khz ( opus@8000h@60i ) FS-7405 [mod_dialplan_xml] Fix condition regex=”all” to work with time conditions FS-7819 [mod_opus] Restore bitrate (if there’s no more packet loss) and added step for 60 ms FS-7773 [mod_sofia] Adding additional transfer events when the fire-transfer-events=true profile parameter is set FS-7820 FreeSWITCH automated unit test and micro benchmark framework FS-7769 [mod_conference] Add new multi-canvas and telepresence features FS-7847 [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 [mod_conference] Add vmute member flag. FS-7846 [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 [mod_sofia] Revise channel fetch on nightmare transfer and add dial-prefix and absolute-dial-string to the nightmare xml FS-7829 [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 [mod_opus] Added use-dtx param in config file (enables DTX on the encoder, announces in fmtp) FS-7824 [mod_png] Add functionality for capturing screenshots from both legs to uuid_write_png FS-7549 [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 [mod_opus] Add support for 80 ms, 100 ms, 120 ms packetization FS-7519 FS-7677 [mod_av] Add H.263 codec support FS-7885 Add getcputime to retrieve FreeSWITCH process CPU usage FS-7889 [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 [mod_png] Allow snapshot of single legged calls FS-7912 [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 [mod_sofia] Improved the xml fetch lookup for channels on nightmare transfer FS-7922 [mod_commands] Added uuid_redirect API command. This provides the equivalent functionality of the dptools “redirect” application as an api command. FS-7806 FS-7803 [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 [verto communicator] Creating Verto Communicator FS-7988 Add a perl script to help file bugs from the command line and add fixbug.pl to tree FS-8009 [verto communicator] Create a grunt project with livereload support. Documentation can be found here. FS-8010 [verto communicator] Add options for googAutoGainControl, googNoiseSuppression, and googHighpassFilter FS-7855 [verto communicator] Pass userVariables back to the live array to allow for displaying the Gravatar associated with a member’s email address FS-8075 [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 [mod_commands] Add getenv FSAPI FS-8036 [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 Removed msvc 2005, 2008, and 2010 non working build systems FS-7373 Expose the custom repo and key path to the build-all command too FS-7648 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 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 Sample build system for a stand alone(out of tree) FreeSWITCH module FS-7601 [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 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 Fixed bug in Debian packaging when trying to build against custom repo FS-7609 Enable building of mod_sangoma_codec for Debian Wheezy/Jessie FS-7667 [mod_java] Fixed include directory detection when using Debian java packages and use detected directory FS-7655 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 Fixed default configuration directory in Debian packages and fixed Debian packaging dependencies on libyuv and libvpx FS-7669 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 [mod_com_g729] Updated the make target installer FS-7644 Added a working windows build without video support for msvc 2013 FS-7666 [mod_managed] Fixed error building mod_managed on non windows platforms FS-7707 Fix build error on CentOS7 FS-7655 Fixed a build error when we have PNG but not YUV FS-7723 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 Added a test cases for FS-7724 and FS-7687 FS-7726 Additional configurations for a QA test case FS-7715 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 [freetdm] Fix some GSM compilation errors and do a bit of code cleanup OPENZAP-237 [freetdm] Use __func__ instead of __FUNCTION__ to comply with c99 in gcc 5.1 FS-7628 [mod_erlang_event] Removed unused variables causing a compilation error FS-7776 Add mod_kazoo to packaging FS-7845 [mod_conference] Break up mod_conference into multiple source files to improve build performance FS-7769 [mod_conference] Fixed a build issue FS-7820 Fix build system typo. Don’t assign the same variable twice. FS-7043 Fixed apr1 unresolved symbols in libfreeswitch.so.1.0.0 FS-7130 Make /run/freeswitch persistent in the Debian packages, so it will start under systemd FS-7860 Prevent a switch_rtp header conflict FS-7130 Make /run/freeswitch persistent, so it will start under systemd FS-7728 Fixed Windows build issues minus video features FS-7965 [mod_conference] Fixed an error thrown when compiling with GCC FS-7985 [mod_voicemail] Fixed a compilation error on 32-bit PCC platform FS-8015 [mod_conference] Add project dir to include for mod_conference so it picks up mod_conference.h for Windows FS-8061 [verto_communicator] Adding license to package.json FS-8047 [build] Fixed build error in mod_basic, mod_rtmp, mod_oreka, and mod_sangoma_codec due to using __FUNCTION__ on newer compilers FS-8054 [mod_rayo] Fixed a warning when building on Debian FS-8055 [build] Add confdir variable to freeswitch.pc FS-7966 [windows] Working msvc 2015 build support. Does not yet include video features. FS-8019 [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 [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 Fixed invalid json format for callflow key FS-7609 [mod_sangoma_codec] Now that libsngtc-dev and libsngtc are in the FS debian repo, enable mod_sangoma_codec FS-7621 [mod_shout] Fixed a slow interrupt FS-7432 Fixed missing a=setup parameter from answering SDP FS-7622 [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 [mod_vlc] A fix for failing to encode audio during the recording of video calls FS-7573 Fixed 80bit tag support for zrtp FS-7636 Fixed an issue with transfer_after_bridge and park_after_bridge pre-empting transfers FS-7654 Fixed an issue with eavesdrop audio not working correctly with a mixture of mono and stereo FS-7641 Fixed a segfault in eavesdrop video support FS-7649 [mod_verto] Fixed issue with h264 codec not being configured in verto.conf.xml FS-7657 [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 [mod_conference] Fixed a bug with the video floor settings not giving the video floor to the speaker FS-7650 [mod_verto] Fixed crash when making a call from a verto user with profile-variables in their user profile FS-7710 [mod_conference] Added the ability to set bandwidth to “auto” for conference config FS-7432 Fixed dtls/srtp, use correct a=setup parameter on recovering channels FS-7678 Fixed for fail_on_single_reject not working with | bridge FS-7709 [mod_verto] Verto compatibility fixes for Firefox FS-7689 [mod_lua] Fixed a bug with lua not loading directory configurations FS-7694 [mod_av] Fixed for leaking file handles when the file is closed. FS-7467 [mod_callcenter] Fixing stuck channels using uuid-standby agents FS-7699 [mod_verto] Fixed for browser compatibility FS-7722 Fixed an issue with record_session including params when creating path FS-7489 [mod_unimrcp] Fixed a TTS Audio Queue Overflow FS-7724 [mod_conference] Fixed a segfault when missing fonts when trying to render banner FS-7519 [mod_av] Fixed a regression in the visual appearance of decode app output FS-7703 Fixed a bug caused by answer_delay being set in the default configurations FS-7679 [mod_verto] Fixed a bug causing one way audio on Chrome when video is enabled and when using a sip without video FS-7729 [mod_verto] Fixed the formatting for IPv6 addresses FS-7734 [mod_nibblebill] Fixed a deadlock FS-7726 Fixed a bug with recording a video session on DTMF command FS-7721 Fixed a segfault caused when using session:recordFile() and session:unsetInputCallback in a lua script FS-7429 [mod_curl] Fixed to output valid json FS-7746 [mod_verto] Fixed a device permission error in verto client FS-7753 [mod_local_stream] Fixed some glitching and freezing video when using hold/unhold FS-7761 [core] Fix shutdown races running api commands during shutdown FS-7767 [mod_sofia] Fixed a segfault caused by invalid arguments to sip_dig FS-7744 [mod_conference] Fixed a bug causing the first user’s video stream to stop when another verto user calls the conference FS-7486 [mod_sofia] Fixed the handling of queued requests FS-7775 [mod_conference] Fix threading issue causing stuck worker threads FS-7777 [mod_imagick] Fixed a regression causing a segfault when playing png & pdf in conference FS-7778 [mod_sofia] Fixed a bug causing a SQL statement to fail because of a double quote instead of a single quote FS-7754 [freetdm] Fixed a bug relating to single digit dial-regex with analog devices FS-7785 [mod_opus] Fix for invalid ptime 30 ms for opus@8000h . Replaced 30 ms with 40 ms. FS-7762 [mod_av] Handle buffer allocation failures of large buffers FS-7849 [verto] Remove extra div breaking full screen in html FS-7832 [mod_opus] Fixes when comparing local and remote fmtp params FS-7731 [mod_xml_cdr] Fixed a curl default connection timeout FS-7844 Fix packet loss fraction when calculating loss average FS-7789 [mod_av] Fixed issue with audio dropping out partway through recordings FS-7854 Add task_runtime to tasks table in core database FS-7856 [mod_av] Fix some segfaults and leaks. FS-7866 Fixed a crash when running incorrect var api expansion syntax “eval ${${external_sip_ip}:4}” FS-7861 FS-7862 [mod_conference] Fixed a crash and other issues caused by multi canvas feature FS-7681 [mod_conference] Factor out conference->canvas and allow per canvas record and play FS-7869 [mod_conference] Fixed a deadlock on shutdown after playing video file that will not display video FS-7654 Fixed regressions on eavesdropping on channels playing a file and on channels with unlike rates FS-7872 [mod_verto] Gracefully fail attempting to transfer 1 legged call FS-7874 [mod_conference] Fixed incorrect layout group count FS-7870 [mod_conference] Allow jsonapi commands to pass the string id field to pass special ID’s like “last” FS-7882 [mod_conference] Allow JSON API commands to send third arg for muting FS-7888 [mod_verto] Fixed namespacing problems in javascript library masked by global verto object FS-7811 Use more common format CIF for blank image FS-7902 [mod_local_stream] Fix for queue filling up when you have a mix of video and non video files FS-7891 [mod_spandsp] Allow spandsp dtmf detector to work on rates other than 8k FS-7839 Correct firefox > 38 DTLS behavior to match new EC requirements FS-7769 [mod_conference] Fixed vmute on personal canvas and fixed changing layouts on personal canvas FS-7893 [mod_conference] Fixed a bug causing muxing write thread to occasionally not close on shutdown FS-7904 Fixed alpha image patching FS-7906 [mod_av] Correct crash from multi-threaded opening or closing of multiple files at the same time FS-7913 [mod_conference] Fixed miscast variable FS-7918 [mod_kazoo] Small fixes in mod_kazoo FS-7917 [mod_sofia] Fixed default config, we really shouldn’t be setting ext-*-ip settings for ipv6 profiles FS-7908 FS-7092 Fixed the generated sdp including telephone-event for the rates of video codecs (90000) when it should only be audio codec rates FS-7927 Fixed a typo in variable name: eavesdrop_annnounce_macro FS-7940 [mod_conference] Fixed an issue where the video image does not appear on the new canvas when switching FS-7930 [mod_conference] Correct termination of conference when the last member with endconf left. FS-7953 [verto communicator] Fixed dialing when typing extension using the keyboard. FS-7958 [mod_conference] Fixed a race condition causing crash in conference video MCU FS-7951 [mod_rayo] Completely clean up mod_rayo if it fails to load FS-7955 [mod_sofia] Fixed a crash caused by invalid contact when using event to send a notify message FS-7970 Fixed crash in video_bug_thread caused by double free FS-7971 [mod_opus] Fixed a rate mismatch and correctly advertise telephone-event and CN rates based on the advertised rates of offered codecs FS-7960 Fixed check_ice routine in switch_core_media.c to not use dropped candidates FS-7975 [mod_voicemail] Fix record-greeting event missing VM-Greeting-Path FS-7969 Fixed a segfault due to pthread_setschedparam() on a thread that has exited FS-7962 Fixed sporadic invite/replaces failure FS-8004 Send keyframe on receiving nack with multiple consecutive packets FS-8005 [mod_opus] Fix for rare decoder error when doing PLC, OPUS_GET_LAST_PACKET_DURATION might return 0 FS-8006 Changed the typedef of switch_core_video_thread_callback_func_t for consistency FS-7932 [mod_verto] Removed the param from the getMute function in verto class, not needed on underlying method FS-8008 [mod_verto] Separate verto default config to have sep v4 and v6 listeners FS-8016 [mod_conference] Reduce buffering of video in conference mux FS-7977 [verto communicator] Fixing default resolution and cleaning code FS-7992 [verto communicator] Fixed device list at settings FS-8017 [verto communicator] Fixed uses of serialized verto in local storage FS-7986 [verto communicator] Fix for devices not refreshing if system config changes FS-7998 [verto communicator] Don’t prompt when recovering call, just do it. FS-8003 [verto communicator] Use audioInDevices instead of audioDevices to match verto plugin FS-8027 [verto communicator] Added watchTask flag to browserSync and add proper regex for replacements FS-8026 [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 [verto_communicator] Upon call recovery, emit an event on $rootScope so that controllers are able to properly clear states. FS-7945 [verto communicator] Use angular-prompt to ask the user to enter a text for the banner. If cancelled, nothing is done. FS-8045 [verto communicator] Make the folder structure compliant with AngularJS best practices and adjust build system. FS-7957 [verto_communicator] Make console less chatty by commenting liveArray updates and get initial state of the conference on liveArray boot event. FS-7979 [verto_communicator] Prompt for extension before transferring a conference member FS-8001 [verto_communicator] For this to work, passing in the parameter was missing FS-7979 [verto_communicator] Removed extra console.log and commented line FS-8025 [verto_communicator] Restored the blue background on the video controls and making icons white again, looking better. FS-8062 [verto_communicator] Fixed video controls tool-tips, now they are visible FS-8048 [verto_communicator] Fixed infinite reconnect after changing hostname and websocket url FS-8066 [verto communicator] Added encoded avatar url to userVariables so that mod_conference can use it when no video, or video mute FS-8018 [verto_communicator] Separation of concerns. Get storage service to manage all settings instead of vertoService FS-8043 [verto_communicator] Removed unnecessary calls to localStorage FS-8040 [verto_communicator] Check if we have a valid resolution reported before calling camera routines and hide controls if none are found FS-8092 [verto_communicator] If there is no data in localStorage, select best resolution for selected camera FS-7840 [verto_communicator] Use chatChannel to send and receive messages from conferences FS-8088 [verto_communicator] Call conference destroy upon hangup and on event destroy to properly unsubscribe from events FS-8046 [verto] Fixed for library not passing device preferences to dialog properly FS-8053 [verto] Don’t receive video on screen share FS-8059 [verto] Fixed typo when transferring party from conference FS-8060 [verto] Conditionally set video tag src to null for FF and empty string for others FS-8087 [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 [mod_shout] Fixed random sockets being closed regression from FS-7621 FS-8029 [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 [mod_voicemail] Fixed a segfault on vm_inject, regression from FS-7968 FS-7968 [mod_voicemail] Fixed verbose events FS-7942 [udptl] Fixed rare segfault on t.38 fax FS-8014 is a duplicate of this issue FS-8031 [dtls] Fixed delayed DTLS media due to changing ICE candidates FS-7903 [proxy_media] Fix Codec PROXY Exists but not at the desired implementation. 0hz 0ms 1ch error when using proxy media. FS-7989 [fixbug.pl] Add –author option FS-8037 [mod_sofia] Fixed so zrtp-passthru doesn’t activate unless the zrtp-hash is in the SDP FS-7135 [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 [mod_av] Fixed a crash when streaming rtmp of desktop share FS-7640 [rtp] Fixed some comparisons in switch_rtp.c to be wraparound proof FS-8057 [core] Fixed a segfault when doing video call when built against libyuv but not libvpx FS-8069 [stun] Fixed ipv6 support missing in stun code FS-8071 [rtp] remove unnecessary auto adjust remote address rules when in ICE mode FS-8077 [mod_conference] Fix memory leak in record FS-8091 [core] Added some missing message names to session message name list (would have caused missing information in some log messages) FS-8093 [mod_silk] Remove giant stack allocation in switch_silk_decode

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