- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re- FreeSWITCH version 1.6.5 installation instructions on CentOS 6.7
by Andrew Keil 23 Nov '15

23 Nov '15

To FreeSWITCH Users, Here are my instructions to install FreeSWITCH 1.6.5 on CentOS 6.7. Please note: Currently mod_fsv and mod_vpx (both are relating to FreeSWITCH video features) are not supported inside these instructions, however if you are wanting to move off 1.4.x (since this is going end-of-life in 2016 at some stage) then this could at least be a very good first step, especially if upgrading to Centos 7 is not possible based on your current environment setup. This is based on a fresh installation of CentOS 6.7 (CentOS-6.7-x86_64-bin-DVD1.iso). In my case I ran a default setup selecting Software Development Workstation. 1) Make sure yum update is run and CentOS is up-to-date 2) {optional} Inside Terminal window select Edit Profile Preferences and select "Use custom default terminal size" and make it 160 x 60. Close terminal and re-open for the change to take effect. 3) yum -y install epel-release 4) yum -y install git gcc-c++ autoconf automake libtool wget python ncurses-devel zlib-devel libjpeg-devel openssl-devel sqlite-devel libcurl-devel libxml2-devel libidn-devel 4.1) These should be already installed and up-to-date 5) yum -y install e2fsprogs-devel pcre-devel speex-devel ldns-devel libedit-devel libyuv-devel opus-devel libvpx-devel unbound-devel libuuid-devel lua-devel libsndfile-devel 5.1) These should install successfully 6) Note: No packages for: libvpx2*, libdb4* (so mod_vpx & mod_fsv will not be supported) 7) cd /usr/local/src 8) git clone -b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git 9) cd /usr/local/src/freeswitch 10) ./bootstrap.sh 11) vi modules.conf 11.1) comment out (ie. place '#' infront): mod_fsv mod_vpx Save and close (<Esc> :wq!) 12) ./configure 13) make 14) make install 15) make cd-sounds-install 16) make cd-moh-install 17) cd /usr/local/freeswitch/conf 18) vi vars.xml 18.1) change: <X-PRE-PROCESS cmd="set" data="default_password=1234"/> { !! set it to something different !! } Save and close (<Esc> :wq!) 19) cd /usr/local/freeswitch/conf/autoload_configs 20) vi modules.conf.xml 20.1) comment out (ie. place '' at the end): mod_fsv mod_vpx Save and close (<Esc> :wq!) 21) cd /usr/local/freeswitch/bin 22) ./freeswitch -nonat -nonatmap 23) ... to exit from the FreeSWITCH console Version: FreeSWITCH Version 1.6.5+git~20151120T205750Z~70b8c17763~64bit (git 70b8c17 2015-11-20 20:57:50Z 64bit) Kind Regards, Andrew Keil Visytel Pty Ltd

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codec inheritance in 3pcc mode
by Serge S. Yuriev 23 Nov '15

23 Nov '15

Hello, Our PBX sending INVITEs w/o SDP so we set enable 3pcc=proxy In dialplan I have set inherit_codec=true We receive from peer: m=audio 43512 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 But FS refuses to send this to Caller 2015-11-06 19:00:06.331944 [DEBUG] switch_ivr_originate.c:416 Codec string G729@8000h@20i not supported on sofia/internal/12550(a)10.0.64.63, skipping inheritance But why while there is no SDP from A side? PBX definitely supports G729. -- Serge S. Yuriev Lead VoIP engineer

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Situation with freeswitch dbh + odbc (mysql) in lua
by John Nash 23 Nov '15

23 Nov '15

I am calling a lua script from dialplan which does following steps ... 1- Gets freeswitch dbh handle (odbc mysql) 2- Runs a mysql stored procedure which returns balance remaining amount for the subscriber who made call. 3- Save the amount in session variable and released dbh handle 4- Play sound files as per output received and saved in step 3 Now upto 200 calls and first 2-3 hours of start all work fine. I can see in Database that script uses 32 connections approx at a given time (I think connection pooling works) but suddenly connections start increasing and go upto 300+ and because of that database stops responding and i have to kill freeswitch to make production DB responsive again. I am not sure where and what I should check. Can someone give me pointers on what I can check?..Or I am doing something fundamentally wrong. Can there be such a case where lua scripts hang and never release connection?

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IPv6 binding
by Frits Wilmink 22 Nov '15

22 Nov '15

Hi, Is there a way to set the IPv6 address FS binds to on startup? As far as I can tell FS just picks the first available dynamic IPv6 address and assigns it to the local_ip_v6 variable. As these address change it makes it hard to setup routing and firewalling. I would like it to use the permanent IPv6 address instead. Thanks, Frits Wilmink

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FS 1.6 multi-platform build without video
by Oleg Stolyar 22 Nov '15

22 Nov '15

Hi guys, There are several threads going on about FS 1.4 end-of-life and the fact that 1.6 can currently only be built on the latest Debian. I wonder if a compromise is possible. My understanding from previous discussions (please correct me if this is wrong) is that the reason for the Debian requirement is the new video related features that require some of the latest dependencies which are hard to figure out. Would it be feasible/easy to create a build that would not have video features but would build on all the same platforms as 1.4? If this is doable, then people who want to use the latest video feature would know that they need to either use Debian or do the legwork to port to their platform but everyone who just want the latest non-video features and bug fixes won't live in fear of 1.4 end-of-life. Does the core team think it's possible? I'd be happy to contribute to it with several of my support incidents or may be able to swing a reasonable bounty.

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Recorded session contains additional audio
by Chris Young 22 Nov '15

22 Nov '15

Hi FS experts! When using uuid_record to record a session, if the recording is stopped and then restarted before the call has been hungup, the second recording includes the audio frames which occurred after the 'stop' command was issued (but before the following 'start'). This leads to the second recording being preceded by a period of silence or unwanted audio. Is this expected behaviour? If so, is there any way to flush the FreeSWITCH's internal audio buffers so that the second recording contains frames from the point where the 'start' command was issued only? Kind regards, Chris Chris Young Software Engineer [cid:image7482a0.PNG@dc00f514.4fb88205] t: +44 118 943 9249 e: chris.young(a)enghouse.com<mailto:chris.young@enghouse.com> w: www.enghouseinteractive.co.uk<http://www.enghouseinteractive.co.uk> [cid:image6c1ba6.PNG@b8963a35.449174f5]

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Problems with mod_soundtouch
by Mike Loiterman 21 Nov '15

21 Nov '15

I was playing around with mod_soundtouch last night and tried this as an extension: <extension name="Soundtouch"> <condition field="destination_number" expression="^9200(\d+)$"> <action application="soundtouch" data="hook_dtmf -0.5o"/> <action application="bridge" data="sofia/gateway/cudatel/99$1"/> </condition> </extension> When I call that extension, any sounds made on the A leg are heard on the B leg as noise. B leg to A leg works perfectly. This exact extension has worked perfectly in the past. Any ideas on what might cause this? System specs: Debian 8.2 FreeSWITCH Version 1.6.2-9~64bit (-9 64bit) Freeswitch was built with Debian packages. ------------------------------ Mike Loiterman Cell: 630-302-4944 Email: mike(a)ascendency.net

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The FreeSWITCH 1.6.5 release is here!
by Ken Rice 20 Nov '15

20 Nov '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1T58y2V The FreeSWITCH 1.6.5 release is here! The FreeSWITCH 1.6.5 release is here! This release contains everything since version 1.6.2. This is a pretty big release for the 1.6 branch so upgrading now is a really good idea. This is a routine maintenance and security release and the resources are located here: Release files are located here: Tarball: http://ift.tt/219gl59 Packaging: http://ift.tt/1U8FMQN (Please Note the updated location for 1.6 packages) Security issues: A bug allowing for a remotely exploited DoS attack through custom crafted network traffic via cJSON has been fixed. We classify this issues as High Severity. A patch was added by Anthony Minessale in commit 4bdca81 to resolve this issue. All versions versions from 1.4.4 through the previous release are vulnerable. We highly recommend updating to the current release version as soon as possible.http://ift.tt/1S9TUXf New features that were added: FS-8243 [mod_opus] Improve the way FEC info is detected within frames by adding support for ptimes higher than 20 ms for FEC detection FS-8161 [mod_opus] Keep FEC enabled only if loss > 10 ( otherwise PLC is supposed to be better) FS-8179 [mod_opus] Improvement on new jitter buffer debugging (debug lookahead FEC) FS-8313 [mod_opus] Introduced new configuration setting ‘decoder-stats’ to show decoder stats at end of call (how many times it did PLC or FEC) FS-8254 [verto_communicator] Create a source map file FS-8263 [verto_communicator] Created the reset banner action, floor and presenter badges, and lock icon in floorLocked status FS-8288 [verto_communicator] Added an About screen with version information and links to FS.org and added a link to Confluence with documentation for VC FS-8289 [verto_communicator] Make mute/unmute audio/video clickable FS-8290 [verto_communicator] Automatically mark dedicated encoder if out/in bandwith isn’t set to ‘Server default’ and adding help text on how to enable dedicated remote encoder FS-8030 [verto_communicator] Added ngSanitize as a dependency, vertoFilters module and picturify filter and changed chat image display behavior (break line before rendering). FS-8293 [verto_communicator] Added built in speed test feature which gives feedback of available bandwidth and customizes call settings based on bandwidth available FS-8401 [verto_communicator] Added Speaker selection in settings modal and video page and refactor the sinkid function into verto lib FS-8545 [verto_communicator] Fixed a read lock regression and do not allow video floor on a member with a reservation id set FS-8195 [core] Compatibility with Solaris 11 process privileges FS-8547 [core] Add error log into stats to log when quality impacting events begin and end FS-8321 [core] Add variable media_mix_inbound_outbound_codecs to mix inbound and outbound codecs. BEHAVIOR CHANGE FS-8281 [core] Expose SRTP and SRTCP crypto keys as channel variables to aid with debugging FS-8287 [mod_local_stream] Refactor local_stream API to be more consistent and add auto complete FS-8375 [mod_conference] Add the field conferenceMemberID to the event broadcasted to inform a verto client about joining a conference. FS-8543 [mod_conference] Improve mute handling on conference and WebRTC FS-8545 [mod_conference][verto_communicator] Improve controls for screen share FS-8546 [mod_conference][mod_verto] Make original video demo back-compatible with livearray-json-status FS-8529 [mod_conference] Added video-floor to personal canvas mode FS-8377 [mod_hiredis] Adding expanded support for limit_* functionality and fixed the handling of hiredis limit release when using an interval. The expectation for interval is to NOT decrement the limit. FS-8380 [mod_av] Improve the handling of vw and vh core file parameters to avoid video cropping and crashing FS-8415 [mod_sofia] Added support for early media with 180 using early_use_180=true FS-8416 [mod_xml_radius] Added the ability to format the variable in the param field FS-8534 [rtcp] Added calculated RTT average (RTCP SR) value to help with detecting congested network links FS-8549 [mod_http_cache] Add support for AWS_ACCESS_KEY_ID and AWS_SECRET_ACCESS_KEY environment variables in S3 profiles FS-8559 [mod_shout] Should have “mpga” in it’s list of supported extensions Improvements in build system, cross platform support, and packaging: FS-8236 [build] Fixed building without libyuv on compilers that throw an error on unused static function and fixed ifdefs for building without libyuv FS-8350 [build] Fix Windows build errors. FS-8389 [build] Fixed msvc 2015 build warnings FS-8316 [build][Debian] Fixed new build warning from latest clang and resolved the build warnings in the modules too FS-8255 [Debian] Fixed codename changes since Jessie was released as stable FS-8271 [Debian] Simplify package building for the default case FS-8270 [Debian] Fix for package installation failing if /etc/freeswitch/tls is missing FS-8285 [Debian] Removed heart attack inducing warning message when updating deb packages FS-7817 Removed use of _NONSTD for Windows builds of spandsp, so (hopefully) eliminate compatibility problem FS-8271 [Debian] Adding some logging, and more cautious handling of spaces in parameters. Now the default will build packages with the upstream FS package repos. This is a change in the default behavior of the Debian packaging system with the justification that in 1.6 it is now required to use the FS public repo for dependencies because system dependencies have been removed from the FS codebase which used to be included. And defaulting to automatically download the binary dependencies because without major changes to package building in cowbuilder(which is the primary supported method of building FS packages), you can’t access the network to build the binary packages from the source package. If using system apt repo list, then include the supplementary ones too FS-7928 FS-7618 [Debian] Systemd and package build improvements FS-8362 [Debian] Now if you install with freeswitch-all you will get the default fonts too FS-8426 [Debian] Put freeswitch.pm into /usr/share/perl5 so it can be found on both Wheezy and Jessie FS-8333 [build][Debian] Added mod_hiredis.deb ESL-111 [python] Fixed esl/python/Makefile to create install directory FS-8233 [automation] In order to clean up build dependencies for the automated tests, convert the tests/*/Makefile.am into an include file for the top level Makefile.am. This will greatly simplify dependency tracking, and allow tests to be rerun easily on FS source code changes. FS-7820 [automation] Use a more appropriate function for printing diagnostics FS-8194 FS-7910 FS-7937 Various systemd service improvements FS-8298 [mod_gsmopen] Fixed a build error FS-8398 [Ubuntu] Added event_handlers/mod_amqp to avoided modules for Ubuntu 14.04 Trusty FS-8239 [mod_av] Fixed the default value to avoid failed build on CentOS 7 FS-8427 [build][mod_av] Fixed an incompatible type for %ld in prinrtf compiler error FS-8248 [mod_event_socket] Moved python binaries into site arch path to match standards The following bugs were squashed: FS-8221 [verto_communicator] Fix number in call history FS-8223 [verto_communicator] Fixing members list layout when callerid is too long FS-8225 [verto_communicator] Avoid duplicate members when recovering calls FS-8214 [verto_communicator] Better handling calls in VC, answering them respecting useVideo param FS-8291 [verto_communicator] Fixed contributors url FS-8229 [verto_communicator] Changing moderator actions bullet menu color to #333 FS-8219 [verto_communicator] Fix for camera not deactivating after init or after hangup FS-8245 [verto_communicator] Fix for Video Resolutions available in “Video Quality” drop down not always correct FS-8251 [verto_communicator] Factory reset now clears all local storage FS-8257 [verto_communicator] Fixed configuration provision url because configuration doesn’t work with `grunt serve` and non pathname urls FS-8273 [verto] [verto_communicator] Clear the CF_RECOVERING flag in a spot that was missed FS-8260 [verto_communicator] Prompt for banner text FS-8067 [verto_communicator] When no email is present make sure mm is the default avatar in the circle this way the talk indicator works on PSTN and SIP callers. FS-8247 [verto_communicator] When websocket disconnects go to splash screen to wait for the reconnect FS-8300 [verto_communicator] Fixing reload bug so reloading twice is no longer needed FS-8331 [verto_communicator] Do not show reconnect splash when user has clicked logout 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-8222 [verto_communicator] Updated getScreenId.js in order to detect plugin issues and attached an ‘ended’ event to screenshare stream in order to detect ‘stop sharing’ click FS-8542 [verto_communicator] Fixed the tooltips of video controls FS-8556 [mod_verto] Screen shares are not recoverable so do not try FS-8293 [mod_verto] Fixed some regressions where speed test caused excessive downlink bandwidth FS-8232 [mod_conference] Conference sending too many video refresh requests FS-8241 [mod_conference] Fix for conference stops playing video when local_stream changes source FS-8261 [mod_conference] Fixed the conference segfaulting when trying to reset the banner FS-8297 [mod_conference] A fix for auto STUN switching IPs quickly and WebRTC video not working FS-8130 [mod_conference] Fix for micro cut-offs and unstable voice issues and fixed a regression causing excessive mark bit detection in some cases FS-8317 [mod_conference] Fix for playing multiple files at once to stack them for immediate playback, sometimes breaking and the floor layer becoming unusable for the rest of the conference. FS-8328 [mod_conference] Fixed missing ‘else’ keyword 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-8384 [mod_conference] Fixed some locking contention issues between external commands and the video engine FS-8527 [mod_conference] Do not send the video of last_video_floor_holder to video_floor_holder if the videos are related FS-8053 [mod_conference][mod_sofia] Fix for WebRTC’s SDP containing a=sendonly for video, but the client still receiving the video stream FS-8220 [core] Fix for DTMF not working between telephone-event/48000 A leg and telephone-event/8000 B leg FS-8166 [core] Mute/unmute while shout is playing audio fails because the channel “has a media bug, hard mute not allowed” FS-8252 [core] Fixed a crash in rtp stack on dtls pointer FS-8283 [core] Handle RTP Contributing Source Identifiers (CSRC) FS-8275 [core] Fix for broken DTMF FS-8282 [core] Fix for sleep is not allowing interruption by uuid_transfer FS-8315 [core] Fix for rtp_media_timeout not working FS-8304 [core] Fix for choppy audio during calls FS-8320 [core] Fixed broken ZRTP not responding to HELLO packet FS-8338 [core] Fix for ringback not working correctly on stereo channels. Also 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-8397 [core] Fixed a race condition incrementing the event-sequence number FS-8154 [core] Fixed a segmentation fault occurring while eavesdropping on video call FS-8391 [core] Fixed a SDP parsing error for rtcp-fb FS-8414 [core] Fixed ptime not updating on codec renegotiation causing audio issues between two legs of a call FS-8417 [core] Fixed SIP offering a=sendonly sometimes replying with a=inactive FS-8404 [core] Media engine will default to PCMU/PCMA if you don’t specify any codecs FS-8411 [core] Replace ping_frame with video_ping_frame in a couple places that were missed causing issues like being unable to record just one side of a video call FS-8425 [core] Fix for DTMF sometimes missed on PSTN call FS-8240 [mod_local_stream] Fixed a/v getting out of sync when running in the background and added video profile parameter for recording 264 and made it default FS-8287 [mod_local_stream] Fixed a segfault from refactor FS-8216 [mod_av] Fixed a regression in hup_local_stream from last commit FS-8274 [mod_av] Fixed a memory leak caused by images not being freed in video_thread_run FS-8318 [mod_av] Fix for recording being out of sync when video from chrome has packet loss FS-8392 [mod_av] Fixed rtpmap to allow both H263 and H263+ codecs to be offered FS-8373 [mod_av] Fix for bad recording quality when using fast encoding FS-8256 [mod_opus] More FMTP cleanup FS-8284 [mod_opus] Use use-dtx setting from config in request to callee. FS-8234 [mod_opus] Send correct (configured) fmtp ptime,minptime,maxptime when originating call FS-8243 [mod_opus] Adding back the missing part removed in 8b088c2 so FEC works in most surroundings FS-8295 [mod_opus] FMTP fixes to continue the cleanup of FEC FS-8302 [mod_opus] Fix some printing/logging because switch_opus_show_audio_bandwidth() was not returning TRUE/FALSE as expected FS-8130 FS-8305 [mod_opus] Fix some warnings and errors caused by dtx and/or jittery webrtc, refactor of last patch, and add suppression of scary harmless message about opus FEC FS-8296 [mod_opus] Improve the way Opus is initialized when a call comes in FS-8179 [mod_opus] Fixed a regression setting fec_decode breaking output on stereo calls FS-8287 [mod_opus] Fixed a segfault from refactor FS-8319 [mod_opus] Fixed and cleaned up switch_opus_has_fec() and switch_opus_info() to avoid FALSE positives for packets with FEC at high frame sizes. FS-8344 [mod_opus] Toggle FEC ON only on the last frame which is to be packed FS-7929 [mod_sofia] Fixed an issue when processing SIP messages while using camp-on FS-6833 [mod_sofia] Add content-type header to ack with sdp FS-6834 [mod_sofia] Found and fixed a few missing content-types in requests/responses with SDP that were outside the norm FS-7834 [mod_sofia] Fixed MOH not working with inbound-bypass-media and resume-media-on-hold FS-8115 [mod_sofia] Disabled unnecessary session timer re-invites for webrtc FS-8536 [mod_sofia] Update to send Keyframe when getting SIP INFO with picture_fast_update FS-7989 [fixbug.pl] Escape double quotes from summary and added more debugging data FS-8246 [mod_json_cdr] Use seconds as default value for delay parameter FS-8308 [mod_format_cdr] Fix to double encode if urlencoding json that is already encoded FS-8311 [mod_voicemail] Fix for leave-message event not containing verbose data for a forwarded voicemail FS-8306 [mod_amqp] If the exchange doesn’t exist, then create it, else fail. This resolves several error cases. And now command queues can specify the queue to subscribe to. This enables very interesting use cases that would involve single job queue, and multiple consumers. FS-8335 [mod_easyroute] Fixed a small error check that results in error message not being displayed 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-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-8413 [mod_lua] Fixed a segfault calling session:getVariable(nil) in lua script. FS-8537 [mod_lua] Fixed a segfault caused by passing nil to various lua functions OPENZAP-240 [mod_freetdm] Fixed a failure to parse caused by using incorrect length when parsing AT responses OPENZAP-238 [mod_freetdm] Several core and gsm improvements including fixing signaling status reporting, a small memory leak, fixing caller id and dnis on inbound calls, span stop functionality, and compilation errors in gcc FS-8553 [configuration] Include verto_contact into the dial-string in the samples FS-8363 [configuration] Don’t register gateways from directory because it registers over what appears to be ipv6 but doesn’t work correctly

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The FreeSWITCH 1.4.26 release is here!
by Ken Rice 20 Nov '15

20 Nov '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1T58wIr The FreeSWITCH 1.4.26 release is here! The FreeSWITCH 1.4.26 release is here! This release contains everything since version 1.4.23. And this is a pretty big release and one of the final routine maintenance releases for the 1.4 branch so upgrading now is a really good idea. The FreeSWITCH 1.4 branch is reaching end of life and the FreeSWITCH Team highly recommends beginning your migration to the 1.6 branch. This is a routine maintenance and security release and the resources are located here: Tarballs: http://ift.tt/1S9TSPa Packaging: http://ift.tt/1U8FMQN Security issues: A bug allowing for a remotely exploited DoS attack through custom crafted network traffic via cJSON has been fixed. We classify this issues as High Severity. A patch was added by Anthony Minessale in commit 4bdca81 to resolve this issue. All versions versions from 1.4.4 through the previous release are vulnerable. We highly recommend updating to the current release version as soon as possible.http://ift.tt/1S9TUXf Improvements in build system, cross platform support, and packaging: FS-8269 [mod_sms] Fixed a build issue FS-8244 [mod_dptools] Fixed a compilation issue The following bugs were squashed: FS-8246 [mod_json_cdr] Use seconds as default value for delay parameter FS-8282 [core] Fix for sleep is not allowing interruption by uuid_transfer FS-8166 [core] Mute/unmute while shout is playing audio fails because the channel “has a media bug, hard mute not allowed” FS-8338 [core] Fix for ringback not working correctly on stereo channels and 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-8215 Fixed the accuracy of MacOSX nanosleep FS-7673 [mod_v8] ODBC NULL value incorrectly evaluated FS-8190 [mod_event_socket] When using nixevent, freeswitch stops sending us certain custom event that were NOT part of the nixevent command stereo the ringback tone is still rendered as mono causing the resulting ringback to be higher pitched and incorrect. FS-8354 [mod_conference] Reverted a back ported patch for rate change detection because it introduced a regression that caused an audio issue 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 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-8308 [mod_format_cdr] Fix to double encode if urlencoding json that is already encoded FS-8413 [mod_lua] Fixed a segfault calling session:getVariable(nil) in lua script.

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uuid_media off with WebRTC
by Dan Edwards 20 Nov '15

20 Nov '15

I'm running FS 1.6.2, trying to get FreeSWITCH to drop out of the media path for 2 WebRTC calls, but I'm getting 'Cannot bypass due to secure connection'. In FS-7166, I found the instructions to set uuid_media_secure_media=true, but that didn’t seem to help. When I tracked down the error message in the source, I find this snippet in switch_ivr.c: if (switch_channel_test_flag(channel, CF_SECURE)) { switch_core_session_rwunlock(session); switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "Cannot bypass %s due to secure connection.\n", switch_channel_get_name(channel)); return SWITCH_STATUS_FALSE; } Wouldn't this code cause for any SRTP call to decline to drop media? Can FS bypass media with 2 WebRTC calls? Thank you, Dan

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